Real Time Streaming Protocol 2.0 (RTSP)
draft-ietf-mmusic-rfc2326bis-34
The information below is for an old version of the document.
| Document | Type |
This is an older version of an Internet-Draft that was ultimately published as RFC 7826.
|
|
|---|---|---|---|
| Authors | Henning Schulzrinne , Anup Rao , Rob Lanphier , Magnus Westerlund , Martin Stiemerling | ||
| Last updated | 2013-06-10 (Latest revision 2013-04-04) | ||
| RFC stream | Internet Engineering Task Force (IETF) | ||
| Formats | |||
| Reviews |
GENART Last Call review
by Elwyn Davies
Ready w/issues
GENART Last Call review
by Robert Sparks
Ready w/issues
|
||
| Additional resources | Mailing list discussion | ||
| Stream | WG state | Submitted to IESG for Publication | |
| Document shepherd | Flemming Andreasen | ||
| Shepherd write-up | Show Last changed 2013-05-08 | ||
| IESG | IESG state | Became RFC 7826 (Proposed Standard) | |
| Consensus boilerplate | Unknown | ||
| Telechat date | (None) | ||
| Responsible AD | Gonzalo Camarillo | ||
| IESG note | ** No value found for 'doc.notedoc.note' ** | ||
| Send notices to | mmusic-chairs@tools.ietf.org, draft-ietf-mmusic-rfc2326bis@tools.ietf.org | ||
| IANA | IANA review state | IANA - Not OK |
draft-ietf-mmusic-rfc2326bis-34
Internet-Draft Real Time Streaming Protocol 2.0 (RTSP) April 2013
number as the original (i.e., the sequence number is not incremented
for retransmissions of the same request). For each new RTSP request
the CSeq value MUST be incremented by one. The initial sequence
number MAY be any number, however, it is RECOMMENDED to start at 0.
Each sequence number series is unique between each requester and
responder, i.e., the client has one series for its request to a
server and the server has another when sending request to the client.
Each requester and responder is identified with its socket address
(IP address and port number).
Proxies that aggregate several sessions on the same transport will
have to ensure that the requests sent towards a particular server
have a joint sequence number space, i.e., they will regularly need to
renumber the CSeq header field in requests (from proxy to server) and
responses (from server to proxy) to fulfill the rules for the header.
The proxy MUST increase the CSeq by one for each request it
transmits, without regard of different sessions.
Example:
CSeq: 239
18.20. Date
The Date header field represents the date and time at which the
message was originated. The inclusion of the Date header in RTSP
message follows these rules:
o An RTSP message, sent either by the client or the server,
containing a body MUST include a Date header, if the sending host
has a clock;
o Clients and servers are RECOMMENDED to include a Date header in
all other RTSP messages, if the sending host has a clock;
o If the server does not have a clock that can provide a reasonable
approximation of the current time, its responses MUST NOT include
a Date header field. In this case, this rule MUST be followed:
Some origin server implementations might not have a clock
available. An origin server without a clock MUST NOT assign
Expires or Last-Modified values to a response, unless these values
were associated with the resource by a system or user with a
reliable clock. It MAY assign an Expires value that is known, at
or before server configuration time, to be in the past (this
allows "pre-expiration" of responses without storing separate
Expires values for each resource).
A received message that does not have a Date header field MUST be
assigned one by the recipient if the message will be cached by that
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recipient. An RTSP implementation without a clock MUST NOT cache
responses without revalidating them on every use. An RTSP cache,
especially a shared cache, SHOULD use a mechanism, such as NTP, to
synchronize its clock with a reliable external standard.
The RTSP-date sent in a Date header SHOULD NOT represent a date and
time subsequent to the generation of the message. It SHOULD
represent the best available approximation of the date and time of
message generation, unless the implementation has no means of
generating a reasonably accurate date and time. In theory, the date
ought to represent the moment just before the message body is
generated. In practice, the date can be generated at any time during
the message origination without affecting its semantic value.
18.21. Expires
The Expires message-header field gives a date and time after which
the description or media-stream should be considered stale. The
interpretation depends on the method:
DESCRIBE response: The Expires header indicates a date and time
after which the presentation description (body) SHOULD be
considered stale.
SETUP response: The Expires header indicate a date and time after
which the media stream SHOULD be considered stale.
A stale cache entry may not normally be returned by a cache (either a
proxy cache or an user agent cache) unless it is first validated with
the origin server (or with an intermediate cache that has a fresh
copy of the message body). See Section 16 for further discussion of
the expiration model.
The presence of an Expires field does not imply that the original
resource will change or cease to exist at, before, or after that
time.
The format is an absolute date and time as defined by RTSP-date. An
example of its use is
Expires: Thu, 01 Dec 1994 16:00:00 GMT
RTSP/2.0 clients and caches MUST treat other invalid date formats,
especially including the value "0", as having occurred in the past
(i.e., already expired).
To mark a response as "already expired," an origin server should use
an Expires date that is equal to the Date header value. To mark a
response as "never expires," an origin server SHOULD use an Expires
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date approximately one year from the time the response is sent.
RTSP/2.0 servers SHOULD NOT send Expires dates more than one year in
the future.
18.22. From
The From request-header field, if given, SHOULD contain an Internet
e-mail address for the human user who controls the requesting user
agent. The address SHOULD be machine-usable, as defined by "mailbox"
in [RFC1123].
This header field MAY be used for logging purposes and as a means for
identifying the source of invalid or unwanted requests. It SHOULD
NOT be used as an insecure form of access protection. The
interpretation of this field is that the request is being performed
on behalf of the person given, who accepts responsibility for the
method performed. In particular, robot agents SHOULD include this
header so that the person responsible for running the robot can be
contacted if problems occur on the receiving end.
The Internet e-mail address in this field MAY be separate from the
Internet host which issued the request. For example, when a request
is passed through a proxy the original issuer's address SHOULD be
used.
The client SHOULD NOT send the From header field without the user's
approval, as it might conflict with the user's privacy interests or
their site's security policy. It is strongly recommended that the
user be able to disable, enable, and modify the value of this field
at any time prior to a request.
18.23. If-Match
The If-Match request-header field is especially useful for ensuring
the integrity of the presentation description, independent of how the
presentation description was received. The presentation description
can be fetched via means external to RTSP (such as HTTP) or via the
DESCRIBE message. In the case of retrieving the presentation
description via RTSP, the server implementation is guaranteeing the
integrity of the description between the time of the DESCRIBE message
and the SETUP message. By including the MTag given in or with the
session description in an If-Match header part of the SETUP request,
the client ensures that resources set up are matching the
description. A SETUP request with the If-Match header for which the
MTag validation check fails, MUST generate a response using 412
(Precondition Failed).
This validation check is also very useful if a session has been
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redirected from one server to another.
18.24. If-Modified-Since
The If-Modified-Since request-header field is used with the DESCRIBE
and SETUP methods to make them conditional. If the requested variant
has not been modified since the time specified in this field, a
description will not be returned from the server (DESCRIBE) or a
stream will not be set up (SETUP). Instead, a 304 (Not Modified)
response MUST be returned without any message-body.
An example of the field is:
If-Modified-Since: Sat, 29 Oct 1994 19:43:31 GMT
18.25. If-None-Match
This request header can be used with one or several message body tags
to make DESCRIBE requests conditional. A client that has one or more
message bodies previously obtained from the resource, can verify that
none of those entities is current by including a list of their
associated message body tags in the If-None-Match header field. The
purpose of this feature is to allow efficient updates of cached
information with a minimum amount of transaction overhead. As a
special case, the value "*" matches any current entity of the
resource.
If any of the message body tags match the message body tag of the
message body that would have been returned in the response to a
similar DESCRIBE request (without the If-None-Match header) on that
resource, or if "*" is given and any current entity exists for that
resource, then the server MUST NOT perform the requested method,
unless required to do so because the resource's modification date
fails to match that supplied in an If-Modified-Since header field in
the request. Instead, if the request method was DESCRIBE, the server
SHOULD respond with a 304 (Not Modified) response, including the
cache-related header fields (particularly MTag) of one of the message
bodies that matched. For all other request methods, the server MUST
respond with a status of 412 (Precondition Failed).
See Section 16.1.3 for rules on how to determine if two message body
tags match.
If none of the message body tags match, then the server MAY perform
the requested method as if the If-None-Match header field did not
exist, but MUST also ignore any If-Modified-Since header field(s) in
the request. That is, if no message body tags match, then the server
MUST NOT return a 304 (Not Modified) response.
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If the request would, without the If-None-Match header field, result
in anything other than a 2xx or 304 status, then the If-None-Match
header MUST be ignored. (See Section 16.1.4 for a discussion of
server behavior when both If-Modified-Since and If-None-Match appear
in the same request.)
The result of a request having both an If-None-Match header field and
an If-Match header field is unspecified and MUST be considered an
illegal request.
18.26. Last-Modified
The Last-Modified message-header field indicates the date and time at
which the origin server believes the presentation description or
media stream was last modified. For the method DESCRIBE, the header
field indicates the last modification date and time of the
description, for SETUP that of the media stream.
An origin server MUST NOT send a Last-Modified date which is later
than the server's time of message origination. In such cases, where
the resource's last modification would indicate some time in the
future, the server MUST replace that date with the message
origination date.
An origin server SHOULD obtain the Last-Modified value of the message
body as close as possible to the time that it generates the Date
value of its response. This allows a recipient to make an accurate
assessment of the message body's modification time, especially if the
message body changes near the time that the response is generated.
RTSP servers SHOULD send Last-Modified whenever feasible.
18.27. Location
The Location response-header field is used to redirect the recipient
to a location other than the Request-URI for completion of the
request or identification of a new resource. For 3xx responses, the
location SHOULD indicate the server's preferred URI for automatic
redirection to the resource. The field value consists of a single
absolute URI.
Note: The Content-Location header field (Section 18.17) differs from
Location in that the Content-Location identifies the original
location of the message body enclosed in the request. It is
therefore possible for a response to contain header fields for both
Location and Content-Location. Also, see Section 16.2 for cache
requirements of some methods.
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18.28. Media-Properties
This general header is used in SETUP response or PLAY_NOTIFY requests
to indicate the media's properties that currently are applicable to
the RTSP session. PLAY_NOTIFY MAY be used to modify these properties
at any point. However, the client SHOULD have received the update
prior to any action related to the new media properties take effect.
For aggregated sessions, the Media-Properties header will be returned
in each SETUP response. The header received in the latest response
is the one that applies on the whole session from this point until
any future update. The header MAY be included without value in
GET_PARAMETER requests to the server with a Session header included
to query the current Media-Properties for the session. The responder
MUST include the current session's media properties.
The media properties expressed by this header is the one applicable
to all media in the RTSP session. For aggregated sessions, the
header expressed the combined media-properties. As a result,
aggregation of media MAY result in a change of the media properties,
and thus the content of the Media-Properties header contained in
subsequent SETUP responses.
The header contains a list of property values that are applicable to
the currently setup media or aggregate of media as indicated by the
RTSP URI in the request. No ordering is enforced within the header.
Property values should be grouped into a single group that handles a
particular orthogonal property. Values or groups that express
multiple properties SHOULD NOT be used. The list of properties that
can be expressed MAY be extended at any time. Unknown property
values MUST be ignored.
This specification defines the following 4 groups and their property
values:
Random Access:
Random-Access: Indicates that random access is possible. May
optionally include a floating point value in seconds indicating
the longest duration between any two random access points in
the media.
Begining-Only: Seeking is limited to the beginning only.
No-Seeking: No seeking is possible.
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Content Modifications:
Immutable: The content will not be changed during the life-time
of the RTSP session.
Dynamic: The content may be changed based on external methods or
triggers
Time-Progressing The media accessible progresses as wallclock
time progresses.
Retention:
Unlimited: Content will be retained for the duration of the life-
time of the RTSP session.
Time-Limited: Content will be retained at least until the
specified wallclock time. The time must be provided in the
absolute time format specified in Section 4.6.
Time-Duration Each individual media unit is retained for at least
the specified time duration. This definition allows for
retaining data with a time based sliding window. The time
duration is expressed as floating point number in seconds. 0.0
is a valid value as this indicates that no data is retained in
a time-progressing session.
Supported Scale:
Scales: A quoted comma separated list of one or more decimal
values or ranges of scale values supported by the content in
arbitrary order. A range has a start and stop value separated
by a colon. A range indicates that the content supports fine
grained selection of scale values. Fine grained allows for
steps at least as small as one tenth of a scale value. A
content is considered to support fine grained selection when
the server in response to a given scale value can produce
content with an actual scale that is less than 1 tenth of scale
unit, i.e., 0.1, from the requested value. Negative values are
supported. The value 0 has no meaning and MUST NOT be used.
Examples of this header for on-demand content and a live stream
without recording are:
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On-demand:
Media-Properties: Random-Access=2.5s, Unlimited, Immutable,
Scales="-20, -10, -4, 0.5:1.5, 4, 8, 10, 15, 20"
Live stream without recording/timeshifting:
Media-Properties: No-Seeking, Time-Progressing, Time-Duration=0.0
18.29. Media-Range
The Media-Range general header is used to give the range of the media
at the time of sending the RTSP message. This header MUST be
included in SETUP response, and PLAY and PAUSE response for media
that are Time-Progressing, and PLAY and PAUSE response after any
change for media that are Dynamic, and in PLAY_NOTIFY request that
are sent due to Media-Property-Update. Media-Range header without
any range specifications MAY be included in GET_PARAMETER requests to
the server to request the current range. The server MUST in this
case include the current range at the time of sending the response.
The header MUST include range specifications for all time formats
supported for the media, as indicated in Accept-Ranges header
(Section 18.5) when setting up the media. The server MAY include
more than one range specification of any given time format to
indicate media that has non-continuous range.
For media that has the Time-Progressing property, the Media-Range
values will only be valid for the particular point in time when it
was issued. As wallclock progresses so will also the media range.
However, it shall be assumed that media time progresses in direct
relationship to wallclock time (with the exception of clock skew) so
that a reasonably accurate estimation of the media range can be
calculated.
18.30. MTag
The MTag response header MAY be included in DESCRIBE, GET_PARAMETER
or SETUP responses. The message body tags (Section 4.8) returned in
a DESCRIBE response, and the one in SETUP refers to the presentation,
i.e. both the returned session description and the media stream.
This allows for verification that one has the right session
description to a media resource at the time of the SETUP request.
However, it has the disadvantage that a change in any of the parts
results in invalidation of all the parts.
If the MTag is provided both inside the message body, e.g. within the
"a=mtag" attribute in SDP, and in the response message, then both
tags MUST be identical. It is RECOMMENDED that the MTag is primarily
given in the RTSP response message, to ensure that caches can use the
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MTag without requiring content inspection. However, for session
descriptions that are distributed outside of RTSP, for example using
HTTP, etc. it will be necessary to include the message body tag in
the session description as specified in Appendix D.1.9.
SETUP and DESCRIBE requests can be made conditional upon the MTag
using the headers If-Match (Section 18.23) and If-None-Match (
Section 18.25).
18.31. Notify-Reason
The Notify Reason header is solely used in the PLAY_NOTIFY method.
It indicates the reason why the server has sent the asynchronous
PLAY_NOTIFY request (see Section 13.5).
18.32. Pipelined-Requests
The Pipelined-Requests general header is used to indicate that a
request is to be executed in the context created by a previous
request(s). The primary usage of this header is to allow pipelining
of SETUP requests so that any additional SETUP request after the
first one does not need to wait for the session ID to be sent back to
the requesting agent. The header contains a unique identifier that
is scoped by the persistent connection used to send the requests.
Upon receiving a request with the Pipelined-Requests the responding
agent MUST look up if there exists a binding between this Pipelined-
Requests identifier for the current persistent connection and an RTSP
session ID. If that exists then the received request is processed
the same way as if it contained the Session header with the found
session ID. If there does not exist a mapping and no Session header
is included in the request, the responding agent MUST create a
binding upon the successful completion of a session creating request,
i.e. SETUP. A binding MUST NOT be created, if the request failed to
create an RTSP session. In case the request contains both a Session
header and the Pipelined-Requests header the Pipelined-Requests MUST
be ignored.
Note: Based on the above definition at least the first request
containing a new unique Pipelined-Requests will be required to be a
SETUP request (unless the protocol is extended with new methods of
creating a session). After that first one, additional SETUP requests
or request of any type using the RTSP session context may include the
Pipelined-Requests header.
When responding to any request that contained the Pipelined-Requests
header the server MUST also include the Session header when a binding
to a session context exist. An RTSP agent that knows the session ID
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SHOULD NOT use the Pipelined-Requests header in any request and only
use the Session header. This as the Session identifier is persistent
across transport contexts, like TCP connections, which the Pipelined-
Requests identifier is not.
The RTSP agent sending the request with a Pipelined-Requests header
has the responsibility for using a unique and previously unused
identifier within the transport context. Currently only a TCP
connection is defined as such transport context. A server MUST
delete the Pipelined-Requests identifier and its binding to a session
upon the termination of that session. Despite the previous mandate,
RTSP agents are RECOMMENDED to not reuse identifiers to allow for
better error handling and logging.
RTSP Proxies may need to translate Pipelined-Requests identifier
values from incoming requests to outgoing to allow for aggregation of
requests onto a persistent connection.
18.33. Proxy-Authenticate
The Proxy-Authenticate response-header field MUST be included as part
of a 407 (Proxy Authentication Required) response. The field value
consists of a challenge that indicates the authentication scheme and
parameters applicable to the proxy for this Request-URI.
The HTTP access authentication process is described in [RFC2617].
Unlike WWW-Authenticate, the Proxy-Authenticate header field applies
only to the current connection and SHOULD NOT be passed on to
downstream agents. However, an intermediate proxy might need to
obtain its own credentials by requesting them from the downstream
agent, which in some circumstances will appear as if the proxy is
forwarding the Proxy-Authenticate header field.
18.34. Proxy-Authorization
The Proxy-Authorization request-header field allows the client to
identify itself (or its user) to a proxy which requires
authentication. The Proxy-Authorization field value consists of
credentials containing the authentication information of the user
agent for the proxy and/or realm of the resource being requested.
The HTTP access authentication process is described in [RFC2617].
Unlike Authorization, the Proxy-Authorization header field applies
only to the next outbound proxy that demanded authentication using
the Proxy-Authenticate field. When multiple proxies are used in a
chain, the Proxy-Authorization header field is consumed by the first
outbound proxy that was expecting to receive credentials. A proxy
MAY relay the credentials from the client request to the next proxy
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if that is the mechanism by which the proxies cooperatively
authenticate a given request.
18.35. Proxy-Require
The Proxy-Require request-header field is used to indicate proxy-
sensitive features that MUST be supported by the proxy. Any Proxy-
Require header features that are not supported by the proxy MUST be
negatively acknowledged by the proxy to the client using the
Unsupported header. The proxy MUST use the 551 (Option Not
Supported) status code in the response. Any feature-tag included in
the Proxy-Require does not apply to the end-point (server or client).
To ensure that a feature is supported by both proxies and servers the
tag needs to be included in also a Require header.
See Section 18.41 for more details on the mechanics of this message
and a usage example. See discussion in the proxies section
(Section 15.1) about when to consider that a feature requires proxy
support.
Example of use:
Proxy-Require: play.basic
18.36. Proxy-Supported
The Proxy-Supported header field enumerates all the extensions
supported by the proxy using feature-tags. The header carries the
intersection of extensions supported by the forwarding proxies. The
Proxy-Supported header MAY be included in any request by a proxy. It
MUST be added by any proxy if the Supported header is present in a
request. When present in a request, the receiver MUST in the
response copy the received Proxy-Supported header.
The Proxy-Supported header field contains a list of feature-tags
applicable to proxies, as described in Section 4.7. The list is the
intersection of all feature-tags understood by the proxies. To
achieve an intersection, the proxy adding the Proxy-Supported header
includes all proxy feature-tags it understands. Any proxy receiving
a request with the header, MUST check the list and removes any
feature-tag(s) it does not support. A Proxy-Supported header present
in the response MUST NOT be touched by the proxies.
Example:
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C->P1: OPTIONS rtsp://example.com/ RTSP/2.0
Supported: foo, bar, blech
User-Agent: PhonyClient/1.2
P1->P2: OPTIONS rtsp://example.com/ RTSP/2.0
Supported: foo, bar, blech
Proxy-Supported: proxy-foo, proxy-bar, proxy-blech
Via: 2.0 pro.example.com
P2->S: OPTIONS rtsp://example.com/ RTSP/2.0
Supported: foo, bar, blech
Proxy-Supported: proxy-foo, proxy-blech
Via: 2.0 pro.example.com, 2.0 prox2.example.com
S->C: RTSP/2.0 200 OK
Supported: foo, bar, baz
Proxy-Supported: proxy-foo, proxy-blech
Public: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN
Via: 2.0 pro.example.com, 2.0 prox2.example.com
18.37. Public
The Public response header field lists the set of methods supported
by the response sender. This header applies to the general
capabilities of the sender and its only purpose is to indicate the
sender's capabilities to the recipient. The methods listed may or
may not be applicable to the Request-URI; the Allow header field
(Section 18.6) MAY be used to indicate methods allowed for a
particular URI.
Example of use:
Public: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN
In the event that there are proxies between the sender and the
recipient of a response, each intervening proxy MUST modify the
Public header field to remove any methods that are not supported via
that proxy. The resulting Public header field will contain an
intersection of the sender's methods and the methods allowed through
by the intervening proxies.
In general, proxies should allow all methods to transparently pass
through from the sending RTSP agent to the receiving RTSP agent,
but there may be cases where this is not desirable for a given
proxy. Modification of the Public response header field by the
intervening proxies ensures that the request sender gets an
accurate response indicating the methods that can be used on the
target agent via the proxy chain.
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18.38. Range
The Range header specifies a time range in PLAY (Section 13.4), PAUSE
(Section 13.6), SETUP (Section 13.3), REDIRECT (Section 13.10), and
PLAY_NOTIFY (Section 13.5) requests and responses. It MAY be
included in GET_PARAMETER requests from the client to the server with
only a Range format and no value to request the current media
position, whether the session is in Play or Ready state in the
included format. The server SHALL, if supporting the range format,
respond with the current playing point or pause point as the start of
the range. If an explicit stop point was used in the previous PLAY
request, then that value shall be included as stop point. Note that
if the server is currently under any type of media playback
manipulation affecting the interpretation of Range, like Scale, that
is also required to be included in any GET_PARAMETER response to
provide complete information.
The range can be specified in a number of units. This specification
defines smpte (Section 4.4), npt (Section 4.5), and clock
(Section 4.6) range units. While byte ranges [H14.35.1] and other
extended units MAY be used, their behavior is unspecified since they
are not normally meaningful in RTSP. Servers supporting the Range
header MUST understand the NPT range format and SHOULD understand the
SMPTE range format. If the Range header is sent in a time format
that is not understood, the recipient SHOULD return 456 (Header Field
Not Valid for Resource) and include an Accept-Ranges header
indicating the supported time formats for the given resource.
Example:
Range: clock=19960213T143205Z-
The Range header contains a range of one single range format. A
range is a half-open interval with a start and an end point,
including the start point, but excluding the end point. A range may
either be fully specified with explicit values for start point and
end point, or have either start or end point be implicit. An
implicit start point indicates the session's pause point, and if no
pause point is set the start of the content. An implicit end point
indicates the end of the content. The usage of both implicit start
and end point is not allowed in the same range header, however, the
exclusion of the range header has that meaning, i.e. from pause point
(or start) until end of content.
Regarding the half-open intervals; a range of A-B starts exactly
at time A, but ends just before B. Only the start time of a media
unit such as a video or audio frame is relevant. For example,
assume that video frames are generated every 40 ms. A range of
10.0-10.1 would include a video frame starting at 10.0 or later
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time and would include a video frame starting at 10.08, even
though it lasted beyond the interval. A range of 10.0-10.08, on
the other hand, would exclude the frame at 10.08.
Please note the difference between NPT time scales' "now" and an
implicit start value. Implicit value reference the current pause-
point. While "now" is the currently ongoing time. In a time-
progressing session with recording (retention for some or full
time) the pause point may be 2 min into the session while now
could be 1 hour into the session.
By default, range intervals increase, where the second point is
larger than the first point.
Example:
Range: npt=10-15
However, range intervals can also decrease if the Scale header (see
Section 18.44) indicates a negative scale value. For example, this
would be the case when a playback in reverse is desired.
Example:
Scale: -1
Range: npt=15-10
Decreasing ranges are still half open intervals as described above.
Thus, for range A-B, A is closed and B is open. In the above
example, 15 is closed and 10 is open. An exception to this rule is
the case when B=0 in a decreasing range. In this case, the range is
closed on both ends, as otherwise there would be no way to reach 0 on
a reverse playback for formats that have such a notion, like NPT and
SMPTE.
Example:
Scale: -1
Range: npt=15-0
In this range both 15 and 0 are closed.
A decreasing range interval without a corresponding negative Scale
header is not valid.
18.39. Referrer
The Referrer request-header field allows the client to specify, for
the server's benefit, the address (URI) of the resource from which
the Request-URI was obtained. The URI refers to that of the
presentation description, typically retrieved via HTTP. The Referrer
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request-header allows a server to generate lists of back-links to
resources for interest, logging, optimized caching, etc. It also
allows obsolete or mistyped links to be traced for maintenance. The
Referrer field MUST NOT be sent if the Request-URI was obtained from
a source that does not have its own URI, such as input from the user
keyboard.
If the field value is a relative URI, it SHOULD be interpreted
relative to the Request-URI. The URI MUST NOT include a fragment.
Because the source of a link might be private information or might
reveal an otherwise private information source, it is strongly
recommended that the user be able to select whether or not the
Referrer field is sent. For example, a streaming client could have a
toggle switch for openly/anonymously, which would respectively
enable/disable the sending of Referrer and From information.
Clients SHOULD NOT include a Referrer header field in a (non-secure)
RTSP request if the referring page was transferred with a secure
protocol.
18.40. Request-Status
This request header is used to indicate the end result for requests
that takes time to complete, such a PLAY (Section 13.4). It is sent
in PLAY_NOTIFY (Section 13.5) with the end-of-stream reason to report
how the PLAY request concluded, either in success or in failure. The
header carries a reference to the request it reports on using the
CSeq number for the session indicated by the Session header in the
request. It provides both a numerical status code (according to
Section 8.1.1) and a human readable reason phrase.
Example:
Request-Status: cseq=63 status=500 reason="Media data unavailable"
18.41. Require
The Require request-header field is used by clients to ensure that
the other end-point supports features that are required in respect to
this request. It can also be used to query if the other end-point
supports certain features, however, the use of the Supported
(Section 18.49) is much more effective in this purpose. The server
MUST respond to this header by using the Unsupported header to
negatively acknowledge those feature-tags which are NOT supported.
The response MUST use the error code 551 (Option Not Supported).
This header does not apply to proxies, for the same functionality in
respect to proxies see Proxy-Require header (Section 18.35) with the
exception of media modifying proxies. Media modifying proxies, due
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to their nature of handling media in a way that is very similar to a
server, do need to understand also the server features to correctly
serve the client.
This is to make sure that the client-server interaction will
proceed without delay when all features are understood by both
sides, and only slow down if features are not understood (as in
the example below). For a well-matched client-server pair, the
interaction proceeds quickly, saving a round-trip often required
by negotiation mechanisms. In addition, it also removes state
ambiguity when the client requires features that the server does
not understand.
Example (Not complete):
C->S: SETUP rtsp://server.com/foo/bar/baz.rm RTSP/2.0
CSeq: 302
Require: funky-feature
Funky-Parameter: funkystuff
S->C: RTSP/2.0 551 Option not supported
CSeq: 302
Unsupported: funky-feature
In this example, "funky-feature" is the feature-tag which indicates
to the client that the fictional Funky-Parameter field is required.
The relationship between "funky-feature" and Funky-Parameter is not
communicated via the RTSP exchange, since that relationship is an
immutable property of "funky-feature" and thus should not be
transmitted with every exchange.
Proxies and other intermediary devices MUST ignore this header. If a
particular extension requires that intermediate devices support it,
the extension should be tagged in the Proxy-Require field instead
(see Section 18.35). See discussion in the proxies section
(Section 15.1) about when to consider that a feature requires proxy
support.
18.42. Retry-After
The Retry-After response-header field can be used with a 503 (Service
Unavailable) response to indicate how long the service is expected to
be unavailable to the requesting client. This field MAY also be used
with any 3xx (Redirection) response to indicate the minimum time the
user-agent is asked to wait before issuing the redirected request.
The value of this field can be either an RTSP-date or an integer
number of seconds (in decimal) after the time of the response.
Example:
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Retry-After: Fri, 31 Dec 1999 23:59:59 GMT
Retry-After: 120
In the latter example, the delay is 2 minutes.
18.43. RTP-Info
The RTP-Info general header field is used to set RTP-specific
parameters in the PLAY and GET_PARAMETER responses or a PLAY_NOTIFY
and GET_PARAMETER requests. For streams using RTP as transport
protocol the RTP-Info header SHOULD be part of a 200 response to
PLAY.
The exclusion of the RTP-Info in a PLAY response for RTP
transported media will result in that a client needs to
synchronize the media streams using RTCP. This may have negative
impact as the RTCP can be lost, and does not need to be
particularly timely in its arrival. Also functionality as
informing the client from which packet a seek has occurred is
affected.
The RTP-Info MAY be included in SETUP responses to provide
synchronization information when changing transport parameters, see
Section 13.3. The RTP-Info header and the Range header MAY be
included in a GET_PARAMETER request from client to server without any
values to request the current playback point and corresponding RTP
synchronization information. When the RTP-Info header is included in
a Request also the Range header MUST be included (Note, Range header
only MAY be used). The server response SHALL include both the Range
header and the RTP-Info header. If the session is in Play state,
then the value of the Range header SHALL be filled in with the
current playback point and with the corresponding RTP-Info values.
If the server is another state, no values are included in the RTP-
Info header. The header is included in PLAY_NOTIFY requests with the
Notify-Reason of end-of-stream to provide RTP information about the
end of the stream.
The header can carry the following parameters:
url: Indicates the stream URI for which the following RTP parameters
correspond, this URI MUST be the same as used in the SETUP
request for this media stream. Any relative URI MUST use the
Request-URI as base URI. This parameter MUST be present.
ssrc: The Synchronization source (SSRC) that the RTP timestamp and
sequence number provided applies to. This parameter MUST be
present.
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seq: Indicates the sequence number of the first packet of the stream
that is direct result of the request. This allows clients to
gracefully deal with packets when seeking. The client uses
this value to differentiate packets that originated before the
seek from packets that originated after the seek. Note that a
client may not receive the packet with the expressed sequence
number, and instead packets with a higher sequence number, due
to packet loss or reordering. This parameter is RECOMMENDED to
be present.
rtptime: MUST indicate the RTP timestamp value corresponding to the
start time value in the Range response header, or if not
explicitly given the implied start point. The client uses this
value to calculate the mapping of RTP time to NPT or other
media timescale. This parameter SHOULD be present to ensure
inter-media synchronization is achieved. There exists no
requirement that any received RTP packet will have the same RTP
timestamp value as the one in the parameter used to establish
synchronization.
A mapping from RTP timestamps to NTP timestamps (wallclock) is
available via RTCP. However, this information is not sufficient
to generate a mapping from RTP timestamps to media clock time
(NPT, etc.). Furthermore, in order to ensure that this
information is available at the necessary time (immediately at
startup or after a seek), and that it is delivered reliably, this
mapping is placed in the RTSP control channel.
In order to compensate for drift for long, uninterrupted
presentations, RTSP clients should additionally map NPT to NTP,
using initial RTCP sender reports to do the mapping, and later
reports to check drift against the mapping.
Example:
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Range:npt=3.25-15
RTP-Info:url="rtsp://example.com/foo/audio" ssrc=0A13C760:seq=45102;
rtptime=12345678,url="rtsp://example.com/foo/video"
ssrc=9A9DE123:seq=30211;rtptime=29567112
Lets assume that Audio uses a 16kHz RTP timestamp clock and Video
a 90kHz RTP timestamp clock. Then the media synchronization is
depicted in the following way.
NPT 3.0---3.1---3.2-X-3.3---3.4---3.5---3.6
Audio PA A
Video V PV
X: NPT time value = 3.25, from Range header.
A: RTP timestamp value for Audio from RTP-Info header (12345678).
V: RTP timestamp value for Video from RTP-Info header (29567112).
PA: RTP audio packet carrying an RTP timestamp of 12344878. Which
corresponds to NPT = (12344878 - A) / 16000 + 3.25 = 3.2
PV: RTP video packet carrying an RTP timestamp of 29573412. Which
corresponds to NPT = (29573412 - V) / 90000 + 3.25 = 3.32
18.44. Scale
A scale value of 1 indicates normal play at the normal forward
viewing rate. If not 1, the value corresponds to the rate with
respect to normal viewing rate. For example, a ratio of 2 indicates
twice the normal viewing rate ("fast forward") and a ratio of 0.5
indicates half the normal viewing rate. In other words, a ratio of 2
has content time increase at twice the playback time. For every
second of elapsed (wallclock) time, 2 seconds of content time will be
delivered. A negative value indicates reverse direction. For
certain media transports this may require certain considerations to
work consistent, see Appendix C.1 for description on how RTP handles
this.
The transmitted data rate SHOULD NOT be changed by selection of a
different scale value. The resulting bit-rate should be reasonably
close to the nominal bit-rate of the content for Scale = 1. The
server has to actively manipulate the data when needed to meet the
bitrate constraints. Implementation of scale changes depends on the
server and media type. For video, a server may, for example, deliver
only key frames or selected frames. For audio, it may time-scale the
audio while preserving pitch or, less desirably, deliver fragments of
audio, or completely mute the audio.
The server and content may restrict the range of scale values that it
supports. The supported values are indicated by the Media-Properties
header (Section 18.28). The client SHOULD only indicate request
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values to be supported. However, as the values may change as the
content progresses a requested value may no longer be valid when the
request arrives. Thus, a non-supported value in a request does not
generate an error, only forces the server to choose the closest
value. The response MUST always contain the actual scale value
chosen by the server.
If the server does not implement the possibility to scale, it will
not return a Scale header. A server supporting Scale operations for
PLAY MUST indicate this with the use of the "play.scale" feature-tag.
When indicating a negative scale for a reverse playback, the Range
header MUST indicate a decreasing range as described in
Section 18.38.
Example of playing in reverse at 3.5 times normal rate:
Scale: -3.5
Range: npt=15-10
18.45. Seek-Style
When a client sends a PLAY request with a Range header to perform a
random access to the media, the client does not know if the server
will pick the first media samples or the first random access point
prior to the request range. Depending on use case, the client may
have a strong preference. To express this preference and provide the
client with information on how the server actually acted on that
preference the Seek-Style header is defined.
Seek-Style is a general header that MAY be included in any PLAY
request to indicate the client's preference for any media stream that
has random access properties. The server MUST always include the
header in any PLAY response for media with random access properties
to indicate what policy was applied. A server that receives an
unknown Seek-Style policy MUST ignore it and select the server
default policy. A client receiving an unknown policy MUST ignore it
and use the Range header and any media synchronization information as
basis to determine what the server did.
This specification defines the following seek policies that may be
requested (see also Section 4.9.1):
RAP: Random Access Point (RAP) is the behavior of requesting the
server to locate the closest previous random access point that
exists in the media aggregate and deliver from that. By
requesting a RAP, media quality will be the best possible as all
media will be delivered from a point where full media state can be
established in the media decoder.
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CoRAP: Conditional Random Access Point (CoRAP) is a variant of the
above RAP behavior. This policy is primarily intended for cases
where there is larger distance between the random access points in
the media. CoRAP is conditioned on that there is a Random Access
Point closer to the requested start point than to the current
pause point. This policy assumes that the media state existing
prior to the pause is usable if delivery is continued. If the
client or server knows that this is not the fact the RAP policy
should be used. In other words: in most cases when the client
requests a start point prior to the current pause point, a valid
decoding dependency chain from the media delivered prior to the
pause and to the requested media unit will not exist. If the
server searched to a random access point the server MUST return
the CoRAP policy in the Seek-Style header and adjust the Range
header to reflect the position of the picked RAP. In case the
random access point is further away and the server selects to
continue from the current pause point it MUST include the "Next"
policy in the Seek-Style header and adjust the Range header start
point to the current pause point.
First-Prior: The first-prior policy will start delivery with the
media unit that has a playout time first prior to the requested
time. For discrete media that would only include media units that
would still be rendered at the request time. For continuous media
that is media that will be rendered during the requested start
time of the range.
Next: The next media units after the provided start time of the
range. For continuous framed media that would mean the first next
frame after the provided time. For discrete media the first unit
that is to be rendered after the provided time. The main usage
for this case is when the client knows it has all media up to a
certain point and would like to continue delivery so that a
complete non-interrupted media playback can be achieved. Example
of such scenarios include switching from a broadcast/multicast
delivery to a unicast based delivery. This policy MUST only be
used on the client's explicit request.
Please note that these expressed preferences exist for optimizing the
startup time or the media quality. The "Next" policy breaks the
normal definition of the Range header to enable a client to request
media with minimal overlap, although some may still occur for
aggregated sessions. RAP and First-Prior both fulfill the
requirement of providing media from the requested range and forward.
However, unless RAP is used, the media quality for many media codecs
using predictive methods can be severely degraded unless additional
data is available as, for example, already buffered, or through other
side channels.
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18.46. Server
The Server response-header field contains information about the
software used by the origin server to handle the request. The field
can contain multiple product tokens and comments identifying the
server and any significant subproducts. The product tokens are
listed in order of their significance for identifying the
application.
Example:
Server: PhonyServer/1.0
If the response is being forwarded through a proxy, the proxy
application MUST NOT modify the Server response-header. Instead, it
SHOULD include a Via field (Section 18.55). If the response is
generated by the proxy, the proxy application MUST return the Server
response-header as previously returned by the server.
18.47. Session
The Session request-header and response-header field identifies an
RTSP session. An RTSP session is created by the server as a result
of a successful SETUP request and in the response the session
identifier is given to the client. The RTSP session exists until
destroyed by a TEARDOWN, REDIRECT or timed out by the server.
The session identifier is chosen by the server (see Section 4.3) and
MUST be returned in the SETUP response. Once a client receives a
session identifier, it MUST be included in any request related to
that session. This means that the Session header MUST be included in
a request, using the following methods: PLAY, PAUSE, and TEARDOWN,
and MAY be included in SETUP, OPTIONS, SET_PARAMETER, GET_PARAMETER,
and REDIRECT, and MUST NOT be included in DESCRIBE. The Session
header MUST NOT be included in the following methods, if these
requests are pipelined and if the session identifier is not yet
known: PLAY, PAUSE, TEARDOWN, SETUP, OPTIONS SET_PARAMETER, and
GET_PARAMETER.
In an RTSP response the session header MUST be included in methods,
SETUP, PLAY, and PAUSE, and MAY be included in methods, TEARDOWN, and
REDIRECT, and if included in the request of the following methods it
MUST also be included in the response, OPTIONS, GET_PARAMETER, and
SET_PARAMETER, and MUST NOT be included in DESCRIBE responses.
Note that a session identifier identifies an RTSP session across
transport sessions or connections. RTSP requests for a given session
can use different URIs (Presentation and media URIs). Note, that
there are restrictions depending on the session which URIs that are
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acceptable for a given method. However, multiple "user" sessions for
the same URI from the same client will require use of different
session identifiers.
The session identifier is needed to distinguish several delivery
requests for the same URI coming from the same client.
The response 454 (Session Not Found) MUST be returned if the session
identifier is invalid.
The header MAY include the session timeout period. If not explicitly
provided this value is set to 60 seconds. As this affects how often
session keep-alives are needed values smaller than 30 seconds are not
recommended. However, larger than default values can be useful in
applications of RTSP that have inactive but established sessions for
longer time periods.
60 seconds was chosen as session timeout value due to: Resulting
in not too frequent keep-alive messages and having low sensitivity
to variations in request response timing. If one reduces the
timeout value to below 30 seconds the corresponding request
response timeout becomes a significant part of the session
timeout. 60 seconds also allows for reasonably rapid recovery of
committed server resources in case of client failure.
18.48. Speed
The Speed request-header field requests the server to deliver
specific amounts of nominal media time per unit of delivery time,
contingent on the server's ability and desire to serve the media
stream at the given speed. The client requests the delivery speed to
be within a given range with a lower and upper bound. The server
SHALL deliver at the highest possible speed within the range, but not
faster than the upper-bound, for which the underlying network path
can support the resulting transport data rates. As long as any speed
value within the given range can be provided the server SHALL NOT
modify the media quality. Only if the server is unable to deliver
media at the speed value provided by the lower bound shall it reduce
the media quality.
Implementation of the Speed functionality by the server is OPTIONAL.
The server can indicate its support through a feature-tag,
play.speed. The lack of a Speed header in the response is an
indication of lack of support of this functionality.
The speed parameter values are expressed as a positive decimal value,
e.g., a value of 2.0 indicates that data is to be delivered twice as
fast as normal. A speed value of zero is invalid. The range is
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specified in the form "lower bound - upper bound". The lower bound
value may be smaller or equal to the upper bound. All speeds may not
be possible to support. Therefore the server MAY modify the
requested values to the closest supported. The actual supported
speed MUST be included in the response. Note, however, that the use
cases may vary and that Speed value ranges such as 0.7 - 0.8,
0.3-2.0, 1.0-2.5, 2.5-2.5 all have their usage.
Example:
Speed: 1.0-2.5
Use of this header changes the bandwidth used for data delivery. It
is meant for use in specific circumstances where delivery of the
presentation at a higher or lower rate is desired. The main use
cases are buffer operations or local scale operations. Implementors
should keep in mind that bandwidth for the session may be negotiated
beforehand (by means other than RTSP), and therefore re-negotiation
may be necessary. To perform Speed operations the server needs to
ensure that the network path can support the resulting bit-rate.
Thus the media transport needs to support feedback so that the server
can react and adapt to the available bitrate.
18.49. Supported
The Supported header enumerates all the extensions supported by the
client or server using feature tags. The header carries the
extensions supported by the message sending client or server. The
Supported header MAY be included in any request. When present in a
request, the receiver MUST respond with its corresponding Supported
header. Note that the Supported header is also included in 4xx and
5xx responses.
The Supported header contains a list of feature-tags, described in
Section 4.7, that are understood by the client or server.
Example:
C->S: OPTIONS rtsp://example.com/ RTSP/2.0
Supported: foo, bar, blech
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
Supported: bar, blech, baz
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18.50. Terminate-Reason
The Terminate-Reason request header allows the server when sending a
REDIRECT or TEARDOWN request to provide a reason for the session
termination and any additional information. This specification
identifies three reasons for Redirections and may be extended in the
future:
Server-Admin: The server needs to be shutdown for some
administrative reason.
Session-Timeout: A client's session is kept alive for extended
periods of time and the server has determined that it needs to
reclaim the resources associated with this session.
Internal-Error An internal error that is impossible to recover from
has occurred forcing the server to terminate the session.
The Server may provide additional parameters containing information
around the redirect. This specification defines the following ones.
time: Provides a wallclock time when the server will stop provide
any service.
user-msg: An UTF-8 text string with a message from the server to the
user. This message SHOULD be displayed to the user.
18.51. Timestamp
The Timestamp general-header describes when the agent sent the
request. The value of the timestamp is of significance only to the
agent and may use any timescale. The responding agent MUST echo the
exact same value and MAY, if it has accurate information about this,
add a floating point number indicating the number of seconds that has
elapsed since it has received the request. The timestamp can be used
by the agent to compute the round-trip time to the responding agent
so that it can adjust the timeout value for retransmissions when
running over an unreliable protocol. It also resolves retransmission
ambiguities for unreliable transport of RTSP.
Note that the present specification provides only for reliable
transport of RTSP messages. The Timestamp general-header is
specified in case the protocol is extended in the future to use
unreliable transport.
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18.52. Transport
The Transport request and response header indicates which transport
protocol is to be used and configures its parameters such as
destination address, compression, multicast time-to-live and
destination port for a single stream. It sets those values not
already determined by a presentation description.
A Transport request header MAY contain a list of transport options
acceptable to the client, in the form of multiple transport
specification entries. Transport specifications are comma separated,
listed in decreasing order of preference. Parameters may be added to
each transport specification, separated by a semicolon. The server
MUST return a Transport response-header in the response to indicate
the values actually chosen if any. If the transport specification is
not supported, no transport header is returned and the request MUST
be responded using the status code 461 (Unsupported Transport)
(Section 17.4.26). In case more than one transport specification was
present in the request, the server MUST return the single (transport-
spec) which was actually chosen, if any. The number of transport-
spec entries is expected to be limited as the client will get
guidance on what configurations that are possible from the
presentation description.
The Transport header MAY also be used in subsequent SETUP requests to
change transport parameters. A server MAY refuse to change
parameters of an existing stream.
A transport specification may only contain one of any given parameter
within it. Parameters MAY be given in any order. Additionally, it
may only contain either of the unicast or the multicast transport
type parameter. All parameters need to be understood in a transport
specification, if not, the transport specification MUST be ignored.
An RTSP proxy of any type that uses or modifies the transport
specification, e.g. access proxy or security proxy, MUST remove
specifications with unknown parameters before forwarding the RTSP
message. If that results in no remaining transport specification the
proxy SHALL send a 461 (Unsupported Transport) (Section 17.4.26)
response without any Transport header.
The Transport header is restricted to describing a single media
stream. (RTSP can also control multiple streams as a single
entity.) Making it part of RTSP rather than relying on a
multitude of session description formats greatly simplifies
designs of firewalls.
The general syntax for the transport specifier is a list of slash
separated tokens:
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Value1/Value2/Value3...
Which for RTP transports take the form:
RTP/profile/lower-transport.
The default value for the "lower-transport" parameters is specific to
the profile. For RTP/AVP, the default is UDP.
There are two different methods for how to specify where the media
should be delivered for unicast transport:
dest_addr: The presence of this parameter and its values indicates
the destination address or addresses (host address and port
pairs for IP flows) necessary for the media transport.
No dest_addr: The lack of the dest_addr parameter indicates that the
server MUST send media to same address for which the RTSP
messages originates.
The choice of method for indicating where the media is to be
delivered depends on the use case. In some cases the only allowed
method will be to use no explicit address indication and have the
server deliver media to the source of the RTSP messages.
For Multicast there is several methods for specifying addresses but
they are different in how they work compared with unicast:
dest_addr with client picked address: The address and relevant
parameters, like TTL (scope), for the actual multicast group to
deliver the media to. There are security implications
(Section 21) with this method that need to be addressed if
using this method because a RTSP server can be used as a DoS
attacker on an existing multicast group.
dest_addr using Session Description Information: The information
included in the transport header can all be coming from the
session description, e.g. the SDP c= and m= line. This
mitigates some of the security issues of the previous methods
as it is the session provider that picks the multicast group
and scope. The client MUST include the information if it is
available in the session description.
No dest_addr: The behavior when no explicit multicast group is
present in a request is not defined.
An RTSP proxy will need to take care. If the media is not desired to
be routed through the proxy, the proxy will need to introduce the
destination indication.
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Below are the configuration parameters associated with transport:
General parameters:
unicast / multicast: This parameter is a mutually exclusive
indication of whether unicast or multicast delivery will be
attempted. One of the two values MUST be specified. Clients
that are capable of handling both unicast and multicast
transmission needs to indicate such capability by including two
full transport-specs with separate parameters for each.
layers: The number of multicast layers to be used for this media
stream. The layers are sent to consecutive addresses starting
at the dest_addr address. If the parameter is not included, it
defaults to a single layer.
dest_addr: A general destination address parameter that can contain
one or more address specifications. Each combination of
protocol/profile/lower transport needs to have the format and
interpretation of its address specification defined. For RTP/
AVP/UDP and RTP/AVP/TCP, the address specification is a tuple
containing a host address and port. Note, only a single
destination parameter per transport spec is intended. The
usage of multiple destinations to distribute a single media to
multiple entities is unspecified.
The client originating the RTSP request MAY specify the
destination address of the stream recipient with the host
address part of the tuple. When the destination address is
specified, the recipient may be a different party than the
originator of the request. To avoid becoming the unwitting
perpetrator of a remote-controlled denial-of-service attack, a
server MUST perform security checks (see Section 21.2.1) and
SHOULD log such attempts before allowing the client to direct a
media stream to a recipient address not chosen by the server.
Implementations cannot rely on TCP as reliable means of client
identification. If the server does not allow the host address
part of the tuple to be set, it MUST return 463 (Destination
Prohibited).
The host address part of the tuple MAY be empty, for example
":58044", in cases when only destination port is desired to be
specified. Responses to requests including the Transport
header with a dest_addr parameter SHOULD include the full
destination address that is actually used by the server. The
server MUST NOT remove address information present already in
the request when responding unless the protocol requires it.
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src_addr: A general source address parameter that can contain one or
more address specifications. Each combination of protocol/
profile/lower transport needs to have the format and
interpretation of its address specification defined. For RTP/
AVP/UDP and RTP/AVP/TCP, the address specification is a tuple
containing a host address and port.
This parameter MUST be specified by the server if it transmits
media packets from another address than the one RTSP messages
are sent to. This will allow the client to verify source
address and give it a destination address for its RTCP feedback
packets, if RTP is used. The address or addresses indicated in
the src_addr parameter SHOULD be used both for sending and
receiving of the media streams data packets. The main reasons
are threefold: First, indicating the port and source address(s)
lets the receiver know where from the packets is expected to
originate. Secondly, traversal of NATs is greatly simplified
when traffic is flowing symmetrically over a NAT binding.
Thirdly, certain NAT traversal mechanisms, needs to know to
which address and port to send so called "binding packets" from
the receiver to the sender, thus creating an address binding in
the NAT that the sender to receiver packet flow can use.
This information may also be available through SDP.
However, since this is more a feature of transport than
media initialization, the authoritative source for this
information should be in the SETUP response.
mode: The mode parameter indicates the methods to be supported for
this session. Currently defined valid values are "PLAY". If
not provided, the default is "PLAY". The "RECORD" value was
defined in RFC 2326 and is in this specification unspecified
but reserved. RECORD and other values may be specified in the
future.
interleaved: The interleaved parameter implies mixing the media
stream with the control stream in whatever protocol is being
used by the control stream, using the mechanism defined in
Section 14. The argument provides the channel number to be
used in the $ block (see Section 14) and MUST be present. This
parameter MAY be specified as an interval, e.g.,
interleaved=4-5 in cases where the transport choice for the
media stream requires it, e.g., for RTP with RTCP. The channel
number given in the request is only a guidance from the client
to the server on what channel number(s) to use. The server MAY
set any valid channel number in the response. The declared
channel(s) are bi-directional, so both end-parties MAY send
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data on the given channel. One example of such usage is the
second channel used for RTCP, where both server and client send
RTCP packets on the same channel.
This allows RTP/RTCP to be handled similarly to the way
that it is done with UDP, i.e., one channel for RTP and
the other for RTCP.
MIKEY: This parameter is used in conjunction with transport
specifications that can utilize MIKEY [RFC3830] for security
context establishment. So far only the SRTP based RTP profiles
SAVP and SAVPF can utilize MIKEY and this is defined in
Appendix C.1.4.1. This parameter can be included both in
request and response messages. The binary MIKEY message SHALL
be BASE64 [RFC4648] encoded before being included in the value
part of the parameter.
Multicast-specific:
ttl: multicast time-to-live for IPv4. When included in requests the
value indicate the TTL value that the client requests the
server to use. In a response, the value actually being used by
the server is returned. A server will need to consider what
values that are reasonable and also the authority of the user
to set this value. Corresponding functions are not needed for
IPv6 as the scoping is part of the IPv6 multicast address
[RFC4291].
RTP-specific:
These parameters MAY only be used if the media transport protocol is
RTP.
ssrc: The ssrc parameter, if included in a SETUP response, indicates
the RTP SSRC [RFC3550] value(s) that will be used by the media
server for RTP packets within the stream. It is expressed as
an eight digit hexadecimal value.
The ssrc parameter MUST NOT be specified in requests. The
functionality of specifying the ssrc parameter in a SETUP
request is deprecated as it is incompatible with the
specification of RTP in RFC 3550[RFC3550]. If the parameter is
included in the Transport header of a SETUP request, the server
SHOULD ignore it, and choose appropriate SSRCs for the stream.
The server SHOULD set the ssrc parameter in the Transport
header of the response.
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RTCP-mux: Use to negotiate the usage of RTP and RTCP multiplexing
[RFC5761] on a single underlying transport stream / flow. The
presence of this parameter in a SETUP request indicates the
client's support and requires the server to use RTP and RTCP
multiplexing. The client SHALL only include one transport
stream in the Transport header specification. To provide the
server with a choice between using RTP/RTCP multiplexing or
not, two different transport header specifications must be
included.
The parameters setup and connection defined below MAY only be used if
the media transport protocol of the lower-level transport is
connection-oriented (such as TCP). However, these parameters MUST
NOT be used when interleaving data over the RTSP control connection.
setup: Clients use the setup parameter on the Transport line in a
SETUP request, to indicate the roles it wishes to play in a TCP
connection. This parameter is adapted from [RFC4145]. We
discuss the use of this parameter in RTP/AVP/TCP non-
interleaved transport in Appendix C.2.2; the discussion below
is limited to syntactic issues. Clients may specify the
following values for the setup parameter:
["active:"] The client will initiate an outgoing connection.
["passive":] The client will accept an incoming connection.
["actpass":] The client is willing to accept an incoming
connection or to initiate an outgoing connection.
If a client does not specify a setup value, the "active" value
is assumed.
In response to a client SETUP request where the setup parameter
is set to "active", a server's 2xx reply MUST assign the setup
parameter to "passive" on the Transport header line.
In response to a client SETUP request where the setup parameter
is set to "passive", a server's 2xx reply MUST assign the setup
parameter to "active" on the Transport header line.
In response to a client SETUP request where the setup parameter
is set to "actpass", a server's 2xx reply MUST assign the setup
parameter to "active" or "passive" on the Transport header
line.
Note that the "holdconn" value for setup is not defined for
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RTSP use, and MUST NOT appear on a Transport line.
connection: Clients use the setup parameter on the Transport line in
a SETUP request, to indicate the SETUP request prefers the
reuse of an existing connection between client and server (in
which case the client sets the "connection" parameter to
"existing"), or that the client requires the creation of a new
connection between client and server (in which cast the client
sets the "connection" parameter to "new"). Typically, clients
use the "new" value for the first SETUP request for a URL, and
"existing" for subsequent SETUP requests for a URL.
If a client SETUP request assigns the "new" value to
"connection", the server response MUST also assign the "new"
value to "connection" on the Transport line.
If a client SETUP request assigns the "existing" value to
"connection", the server response MUST assign a value of
"existing" or "new" to "connection" on the Transport line, at
its discretion.
The default value of "connection" is "existing", for all SETUP
requests (initial and subsequent).
The combination of transport protocol, profile and lower transport
needs to be defined. A number of combinations are defined in the
Appendix C.
Below is a usage example, showing a client advertising the capability
to handle multicast or unicast, preferring multicast. Since this is
a unicast-only stream, the server responds with the proper transport
parameters for unicast.
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C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/2.0
CSeq: 302
Transport: RTP/AVP;multicast;mode="PLAY",
RTP/AVP;unicast;dest_addr="192.0.2.5:3456"/
"192.0.2.5:3457";mode="PLAY"
Accept-Ranges: NPT, SMPTE, UTC
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 302
Date: Thu, 23 Jan 1997 15:35:06 GMT
Session: 47112344
Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:3456"/
"192.0.2.5:3457";src_addr="192.0.2.224:6256"/
"192.0.2.224:6257";mode="PLAY"
Accept-Ranges: NPT
Media-Properties: Random-Access=0.6, Dynamic,
Time-Limited=20081128T165900
18.53. Unsupported
The Unsupported response-header lists the features not supported by
the responding RTSP agent. In the case where the feature was
specified via the Proxy-Require field (Section 18.35), if there is a
proxy on the path between the client and the server, the proxy MUST
send a response message with a status code of 551 (Option Not
Supported). The request MUST NOT be forwarded.
See Section 18.41 for a usage example.
18.54. User-Agent
The User-Agent general-header field contains information about the
user agent originating the request. This is for statistical
purposes, the tracing of protocol violations, and automated
recognition of user agents for the sake of tailoring responses to
avoid particular user agent limitations. User agents SHOULD include
this field with requests. The field can contain multiple product
tokens and comments identifying the agent and any subproducts which
form a significant part of the user agent. By convention, the
product tokens are listed in order of their significance for
identifying the application.
Example:
User-Agent: PhonyClient/1.2
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18.55. Via
The Via general-header field MUST be used by proxies to indicate the
intermediate protocols and recipients between the user agent and the
server on requests, and between the origin server and the client on
responses. The field is intended to be used for tracking message
forwards, avoiding request loops, and identifying the protocol
capabilities of all senders along the request/response chain.
Multiple Via field values represents each proxy that has forwarded
the message. Each recipient MUST append its information such that
the end result is ordered according to the sequence of forwarding
applications.
Proxies (e.g., Access Proxy or Translator Proxy) SHOULD NOT, by
default, forward the names and ports of hosts within the private/
protected region. This information SHOULD only be propagated if
explicitly enabled. If not enabled, the via-received of any host
behind the firewall/NAT SHOULD be replaced by an appropriate
pseudonym for that host.
For organizations that have strong privacy requirements for hiding
internal structures, a proxy MAY combine an ordered subsequence of
Via header field entries with identical sent-protocol values into a
single such entry. Applications MUST NOT combine entries which have
different received-protocol values.
18.56. WWW-Authenticate
The WWW-Authenticate response-header field MUST be included in 401
(Unauthorized) response messages. The field value consists of at
least one challenge that indicates the authentication scheme(s) and
parameters applicable to the Request-URI.
The HTTP access authentication process is described in [RFC2617].
User agents are advised to take special care in parsing the WWW-
Authenticate field value as it might contain more than one challenge,
or if more than one WWW-Authenticate header field is provided, the
contents of a challenge itself can contain a comma-separated list of
authentication parameters.
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19. Security Framework
The RTSP security framework consists of two high level components:
the pure authentication mechanisms based on HTTP authentication, and
the message transport protection based on TLS, which is independent
of RTSP. Because of the similarity in syntax and usage between RTSP
servers and HTTP servers, the security for HTTP is re-used to a large
extent.
19.1. RTSP and HTTP Authentication
RTSP and HTTP share common authentication schemes, and thus follow
the same usage guidelines as specified in [RFC2617] and also in
[H15]. Servers SHOULD implement both basic and digest [RFC2617]
authentication. Clients MUST implement both basic and digest
authentication [RFC2617] so that a server that requires the client to
authenticate can trust that the capability is present.
It should be stressed that using the HTTP authentication alone does
not provide full control message security. Therefore, in
environments requiring tighter security for the control messages, TLS
SHOULD be used, see Section 19.2.
19.2. RTSP over TLS
RTSP agents MUST implement RTSP over TLS as defined in this section
and the next Section 19.3. RTSP MUST follow the same guidelines with
regards to TLS [RFC5246] usage as specified for HTTP, see [RFC2818].
RTSP over TLS is separated from unsecured RTSP both on URI level and
port level. Instead of using the "rtsp" scheme identifier in the
URI, the "rtsps" scheme identifier MUST be used to signal RTSP over
TLS. If no port is given in a URI with the "rtsps" scheme, port 322
MUST be used for TLS over TCP/IP.
When a client tries to setup an insecure channel to the server (using
the "rtsp" URI), and the policy for the resource requires a secure
channel, the server MUST redirect the client to the secure service by
sending a 301 redirect response code together with the correct
Location URI (using the "rtsps" scheme). A user or client MAY
upgrade a non secured URI to a secured by changing the scheme from
"rtsp" to "rtsps". A server implementing support for "rtsps" MUST
allow this.
It should be noted that TLS allows for mutual authentication (when
using both server and client certificates). Still, one of the more
common ways TLS is used is to only provide server side authentication
(often to avoid client certificates). TLS is then used in addition
to HTTP authentication, providing transport security and server
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authentication, while HTTP Authentication is used to authenticate the
client.
RTSP includes the possibility to keep a TCP session up between the
client and server, throughout the RTSP session lifetime. It may be
convenient to keep the TCP session, not only to save the extra setup
time for TCP, but also the extra setup time for TLS (even if TLS uses
the resume function, there will be almost two extra round trips).
Still, when TLS is used, such behavior introduces extra active state
in the server, not only for TCP and RTSP, but also for TLS. This may
increase the vulnerability to DoS attacks.
In addition to these recommendations, Section 19.3 gives further
recommendations of TLS usage with proxies.
19.3. Security and Proxies
The nature of a proxy is often to act as a "man-in-the-middle", while
security is often about preventing the existence of a "man-in-the-
middle". This section provides clients with the possibility to use
proxies even when applying secure transports (TLS) between the RTSP
agents. The TLS proxy mechanism allows for server and proxy
identification using certificates. However, the client cannot be
identified based on certificates. The client needs to select between
using the procedure specified below or using a TLS connection
directly (by-passing any proxies) to the server. The choice may be
dependent on policies.
There are basically two categories of proxies, the transparent
proxies (of which the client is not aware) and the non-transparent
proxies (of which the client is aware), see Section 15 for an
introduction to RTSP proxies. An infrastructure based on proxies
requires that the trust model is such that both client and servers
can trust the proxies to handle the RTSP messages correctly. To be
able to trust a proxy, the client and server also needs to be aware
of the proxy. Hence, transparent proxies cannot generally be seen as
trusted and will not work well with security (unless they work only
at transport layer). In the rest of this section any reference to
proxy will be to a non-transparent proxy, which inspects or
manipulates the RTSP messages.
HTTP Authentication is built on the assumption of proxies and can
provide user-proxy authentication and proxy-proxy/server
authentication in addition to the client-server authentication.
When TLS is applied and a proxy is used, the client will connect to
the proxy's address when connecting to any RTSP server. This implies
that for TLS, the client will authenticate the proxy server and not
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the end server. Note that when the client checks the server
certificate in TLS, it MUST check the proxy's identity (URI or
possibly other known identity) against the proxy's identity as
presented in the proxy's Certificate message.
The problem is that for a proxy accepted by the client, the proxy
needs to be provided information on which grounds it should accept
the next-hop certificate. Both the proxy and the user may have rules
for this, and the user should have the possibility to select the
desired behavior. To handle this case, the Accept-Credentials header
(See Section 18.2) is used, where the client can request the proxy/
proxies to relay back the chain of certificates used to authenticate
any intermediate proxies as well as the server. The assumption that
the proxies are viewed as trusted, gives the user a possibility to
enforce policies to each trusted proxy of whether it should accept
the next agent in the chain. However, it should be noted that not
all deployments will return the chain of certificates used to
authenticate any intermediate proxies as well as the server. An
operator of such a deployment may want to hide its topology from the
client. It should be noted well that the client does not have any
insight into the proxy's operation. Even if the proxy is trusted, it
can still return an incomplete chain of certificates.
A proxy MUST use TLS for the next hop if the RTSP request includes a
"rtsps" URI. TLS MAY be applied on intermediate links (e.g. between
client and proxy, or between proxy and proxy), even if the resource
and the end server are not required to use it. The proxy MUST, when
initiating the next hop TLS connection, use the incoming TLS
connections cipher suite list, only modified by removing any cipher
suites that the proxy does not support. In case a proxy fails to
establish a TLS connection due to cipher suite mismatch between proxy
and next hop proxy or server, this is indicated using error code 472
(Failure to establish secure connection).
19.3.1. Accept-Credentials
The Accept-Credentials header can be used by the client to distribute
simple authorization policies to intermediate proxies. The client
includes the Accept-Credentials header to dictate how the proxy
treats the server/next proxy certificate. There are currently three
methods defined:
Any, which means that the proxy (or proxies) MUST accept whatever
certificate presented. This is of course not a recommended
option to use, but may be useful in certain circumstances (such
as testing).
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Proxy, which means that the proxy (or proxies) MUST use its own
policies to validate the certificate and decide whether to
accept it or not. This is convenient in cases where the user
has a strong trust relation with the proxy. Reasons why a
strong trust relation may exist are; personal/company proxy,
proxy has a out-of-band policy configuration mechanism.
User, which means that the proxy (or proxies) MUST send credential
information about the next hop to the client for authorization.
The client can then decide whether the proxy should accept the
certificate or not. See Section 19.3.2 for further details.
If the Accept-Credentials header is not included in the RTSP request
from the client, then the "Proxy" method MUST be used as default. If
another method than the "Proxy" is to be used, then the Accept-
Credentials header MUST be included in all of the RTSP requests from
the client. This is because it cannot be assumed that the proxy
always keeps the TLS state or the user's previous preference between
different RTSP messages (in particular if the time interval between
the messages is long).
With the "Any" and "Proxy" methods the proxy will apply the policy as
defined for each method. If the policy does not accept the
credentials of the next hop, the proxy MUST respond with a message
using status code 471 (Connection Credentials not accepted).
An RTSP request in the direction server to client MUST NOT include
the Accept-Credentials header. As for the non-secured communication,
the possibility for these requests depends on the presence of a
client established connection. However, if the server to client
request is in relation to a session established over a TLS secured
channel, it MUST be sent in a TLS secured connection. That secured
connection MUST also be the one used by the last client to server
request. If no such transport connection exists at the time when the
server desires to send the request, the server MUST discard the
message.
Further policies MAY be defined and registered, but should be done so
with caution.
19.3.2. User approved TLS procedure
For the "User" method, each proxy MUST perform the following
procedure for each RTSP request:
o Setup the TLS session to the next hop if not already present (i.e.
run the TLS handshake, but do not send the RTSP request).
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o Extract the peer certificate chain for the TLS session.
o Check if a matching identity and hash of the peer certificate is
present in the Accept-Credentials header. If present, send the
message to the next hop, and conclude these procedures. If not,
go to the next step.
o The proxy responds to the RTSP request with a 470 or 407 response
code. The 407 response code MAY be used when the proxy requires
both user and connection authorization from user or client. In
this message the proxy MUST include a Connection-Credentials
header, see Section 18.12 with the next hop's identity and
certificate.
The client MUST upon receiving a 470 or 407 response with Connection-
Credentials header take the decision on whether to accept the
certificate or not (if it cannot do so, the user SHOULD be
consulted). If the certificate is accepted, the client has to again
send the RTSP request. In that request the client has to include the
Accept-Credentials header including the hash over the DER encoded
certificate for all trusted proxies in the chain.
Example:
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C->P: SETUP rtsps://test.example.org/secret/audio RTSP/2.0
CSeq: 2
Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:4588"/
"192.0.2.5:4589"
Accept-Ranges: NPT, SMPTE, UTC
Accept-Credentials: User
P->C: RTSP/2.0 470 Connection Authorization Required
CSeq: 2
Connection-Credentials: "rtsps://test.example.org";
MIIDNTCCAp...
C->P: SETUP rtsps://test.example.org/secret/audio RTSP/2.0
CSeq: 3
Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:4588"/
"192.0.2.5:4589"
Accept-Credentials: User "rtsps://test.example.org";sha-256;
dPYD7txpoGTbAqZZQJ+vaeOkyH4=
Accept-Ranges: NPT, SMPTE, UTC
P->S: SETUP rtsps://test.example.org/secret/audio RTSP/2.0
CSeq: 3
Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:4588"/
"192.0.2.5:4589"
Via: RTSP/2.0 proxy.example.org
Accept-Credentials: User "rtsps://test.example.org";sha-256;
dPYD7txpoGTbAqZZQJ+vaeOkyH4=
Accept-Ranges: NPT, SMPTE, UTC
One implication of this process is that the connection for secured
RTSP messages may take significantly more round-trip times for the
first message. A complete extra message exchange between the proxy
connecting to the next hop and the client results because of the
process for approval for each hop. However, if each message contains
the chain of proxies that the requester accepts, the remaining
message exchange should not be delayed. The procedure of including
the credentials in each request rather than building state in each
proxy, avoids the need for revocation procedures.
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20. Syntax
The RTSP syntax is described in an Augmented Backus-Naur Form (ABNF)
as defined in RFC 5234 [RFC5234]. It uses the basic definitions
present in RFC 5234.
Please note that ABNF strings, e.g. "Accept", are case insensitive
as specified in section 2.3 of RFC 5234.
The RTSP syntax makes use of the ISO 10646 character set in UTF-8
encoding RFC 3629 [RFC3629].
20.1. Base Syntax
RTSP header values can be folded onto multiple lines if the
continuation line begins with a space or horizontal tab. All linear
white space, including folding, has the same semantics as SP. A
recipient MAY replace any linear white space with a single SP before
interpreting the field value or forwarding the message downstream.
This is intended to behave exactly as HTTP/1.1 as described in RFC
2616 [RFC2616]. The SWS construct is used when linear white space is
optional, generally between tokens and separators.
To separate the header name from the rest of value, a colon is used,
which, by the above rule, allows whitespace before, but no line
break, and whitespace after, including a line break. The HCOLON
defines this construct.
OCTET = %x00-FF ; any 8-bit sequence of data
CHAR = %x01-7F ; any US-ASCII character (octets 1 - 127)
UPALPHA = %x41-5A ; any US-ASCII uppercase letter "A".."Z"
LOALPHA = %x61-7A ;any US-ASCII lowercase letter "a".."z"
ALPHA = UPALPHA / LOALPHA
DIGIT = %x30-39 ; any US-ASCII digit "0".."9"
CTL = %x00-1F / %x7F ; any US-ASCII control character
; (octets 0 - 31) and DEL (127)
CR = %x0D ; US-ASCII CR, carriage return (13)
LF = %x0A ; US-ASCII LF, linefeed (10)
SP = %x20 ; US-ASCII SP, space (32)
HT = %x09 ; US-ASCII HT, horizontal-tab (9)
BACKSLASH = %x5C ; US-ASCII backslash (92)
CRLF = CR LF
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LWS = [CRLF] 1*( SP / HT ) ; Line-breaking White Space
SWS = [LWS] ; Separating White Space
HCOLON = *( SP / HT ) ":" SWS
TEXT = %x20-7E / %x80-FF ; any OCTET except CTLs
tspecials = "(" / ")" / "<" / ">" / "@"
/ "," / ";" / ":" / BACKSLASH / DQUOTE
/ "/" / "[" / "]" / "?" / "="
/ "{" / "}" / SP / HT
token = 1*(%x21 / %x23-27 / %x2A-2B / %x2D-2E / %x30-39
/ %x41-5A / %x5E-7A / %x7C / %x7E)
; 1*<any CHAR except CTLs or tspecials>
quoted-string = ( DQUOTE *qdtext DQUOTE )
qdtext = %x20-21 / %x23-5B / %x5D-7E / quoted-pair
/ UTF8-NONASCII
; No DQUOTE and no "\"
quoted-pair = "\\" / ( "\" DQUOTE )
ctext = %x20-27 / %x2A-7E
/ %x80-FF ; any OCTET except CTLs, "(" and ")"
generic-param = token [ EQUAL gen-value ]
gen-value = token / host / quoted-string
safe = "$" / "-" / "_" / "." / "+"
extra = "!" / "*" / "'" / "(" / ")" / ","
rtsp-extra = "!" / "*" / "'" / "(" / ")"
HEX = DIGIT / "A" / "B" / "C" / "D" / "E" / "F"
/ "a" / "b" / "c" / "d" / "e" / "f"
LHEX = DIGIT / "a" / "b" / "c" / "d" / "e" / "f"
; lowercase "a-f" Hex
reserved = ";" / "/" / "?" / ":" / "@" / "&" / "="
unreserved = ALPHA / DIGIT / safe / extra
rtsp-unreserved = ALPHA / DIGIT / safe / rtsp-extra
base64 = *base64-unit [base64-pad]
base64-unit = 4base64-char
base64-pad = (2base64-char "==") / (3base64-char "=")
base64-char = ALPHA / DIGIT / "+" / "/"
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SLASH = SWS "/" SWS ; slash
EQUAL = SWS "=" SWS ; equal
LPAREN = SWS "(" SWS ; left parenthesis
RPAREN = SWS ")" SWS ; right parenthesis
COMMA = SWS "," SWS ; comma
SEMI = SWS ";" SWS ; semicolon
COLON = SWS ":" SWS ; colon
MINUS = SWS "-" SWS ; minus/dash
LDQUOT = SWS DQUOTE ; open double quotation mark
RDQUOT = DQUOTE SWS ; close double quotation mark
RAQUOT = ">" SWS ; right angle quote
LAQUOT = SWS "<" ; left angle quote
TEXT-UTF8char = %x21-7E / UTF8-NONASCII
UTF8-NONASCII = UTF8-1 / UTF8-2 / UTF8-3 / UTF8-4
UTF8-CONT = %x80-BF
POS-FLOAT = 1*12DIGIT ["." 1*9DIGIT]
FLOAT = ["-"] POS-FLOAT
20.2. RTSP Protocol Definition
20.2.1. Generic Protocol elements
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RTSP-IRI = schemes ":" IRI-rest
IRI-rest = ihier-part [ "?" iquery ]
ihier-part = "//" iauthority ipath-abempty
RTSP-IRI-ref = RTSP-IRI / irelative-ref
irelative-ref = irelative-part [ "?" iquery ]
irelative-part = "//" iauthority ipath-abempty
/ ipath-absolute
/ ipath-noscheme
/ ipath-empty
iauthority = < As defined in RFC 3987>
ipath = ipath-abempty ; begins with "/" or is empty
/ ipath-absolute ; begins with "/" but not "//"
/ ipath-noscheme ; begins with a non-colon segment
/ ipath-rootless ; begins with a segment
/ ipath-empty ; zero characters
ipath-abempty = *( "/" isegment )
ipath-absolute = "/" [ isegment-nz *( "/" isegment ) ]
ipath-noscheme = isegment-nz-nc *( "/" isegment )
ipath-rootless = isegment-nz *( "/" isegment )
ipath-empty = 0<ipchar>
isegment = *ipchar [";" *ipchar]
isegment-nz = 1*ipchar [";" *ipchar]
/ ";" *ipchar
isegment-nz-nc = (1*ipchar-nc [";" *ipchar-nc])
/ ";" *ipchar-nc
; non-zero-length segment without any colon ":"
ipchar = iunreserved / pct-encoded / sub-delims / ":" / "@"
ipchar-nc = iunreserved / pct-encoded / sub-delims / "@"
iquery = < As defined in RFC 3987>
iunreserved = < As defined in RFC 3987>
pct-encoded = < As defined in RFC 3987>
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RTSP-URI = schemes ":" URI-rest
RTSP-REQ-URI = schemes ":" URI-req-rest
RTSP-URI-Ref = RTSP-URI / RTSP-Relative
RTSP-REQ-Ref = RTSP-REQ-URI / RTSP-REQ-Rel
schemes = "rtsp" / "rtsps" / scheme
scheme = < As defined in RFC 3986>
URI-rest = hier-part [ "?" query ]
URI-req-rest = hier-part [ "?" query ]
; Note fragment part not allowed in requests
hier-part = "//" authority path-abempty
RTSP-Relative = relative-part [ "?" query ]
RTSP-REQ-Rel = relative-part [ "?" query ]
relative-part = "//" authority path-abempty
/ path-absolute
/ path-noscheme
/ path-empty
authority = < As defined in RFC 3986>
query = < As defined in RFC 3986>
path = path-abempty ; begins with "/" or is empty
/ path-absolute ; begins with "/" but not "//"
/ path-noscheme ; begins with a non-colon segment
/ path-rootless ; begins with a segment
/ path-empty ; zero characters
path-abempty = *( "/" segment )
path-absolute = "/" [ segment-nz *( "/" segment ) ]
path-noscheme = segment-nz-nc *( "/" segment )
path-rootless = segment-nz *( "/" segment )
path-empty = 0<pchar>
segment = *pchar [";" *pchar]
segment-nz = ( 1*pchar [";" *pchar]) / (";" *pchar)
segment-nz-nc = ( 1*pchar-nc [";" *pchar-nc]) / (";" *pchar-nc)
; non-zero-length segment without any colon ":"
pchar = unreserved / pct-encoded / sub-delims / ":" / "@"
pchar-nc = unreserved / pct-encoded / sub-delims / "@"
sub-delims = "!" / "$" / "&" / "'" / "(" / ")"
/ "*" / "+" / "," / "="
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smpte-range = smpte-type ["=" smpte-range-spec]
; See section 3.4
smpte-range-spec = ( smpte-time "-" [ smpte-time ] )
/ ( "-" smpte-time )
smpte-type = "smpte" / "smpte-30-drop"
/ "smpte-25" / smpte-type-extension
; other timecodes may be added
smpte-type-extension = "smpte" token
smpte-time = 1*2DIGIT ":" 1*2DIGIT ":" 1*2DIGIT
[ ":" 1*2DIGIT [ "." 1*2DIGIT ] ]
npt-range = "npt" ["=" npt-range-spec]
npt-range-spec = ( npt-time "-" [ npt-time ] ) / ( "-" npt-time )
npt-time = "now" / npt-sec / npt-hhmmss
npt-sec = 1*19DIGIT [ "." 1*9DIGIT ]
npt-hhmmss = npt-hh ":" npt-mm ":" npt-ss [ "." 1*9DIGIT ]
npt-hh = 1*19DIGIT ; any positive number
npt-mm = 1*2DIGIT ; 0-59
npt-ss = 1*2DIGIT ; 0-59
utc-range = "clock" ["=" utc-range-spec]
utc-range-spec = ( utc-time "-" [ utc-time ] ) / ( "-" utc-time )
utc-time = utc-date "T" utc-clock "Z"
utc-date = 8DIGIT
utc-clock = 6DIGIT [ "." 1*9DIGIT ]
feature-tag = token
session-id = 1*256( ALPHA / DIGIT / safe )
extension-header = header-name HCOLON header-value
header-name = token
header-value = *(TEXT-UTF8char / UTF8-CONT / LWS)
20.2.2. Message Syntax
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RTSP-message = Request / Response ; RTSP/2.0 messages
Request = Request-Line
*((general-header
/ request-header
/ message-header) CRLF)
CRLF
[ message-body-data ]
Response = Status-Line
*((general-header
/ response-header
/ message-header) CRLF)
CRLF
[ message-body-data ]
Request-Line = Method SP Request-URI SP RTSP-Version CRLF
Status-Line = RTSP-Version SP Status-Code SP Reason-Phrase CRLF
Method = "DESCRIBE"
/ "GET_PARAMETER"
/ "OPTIONS"
/ "PAUSE"
/ "PLAY"
/ "PLAY_NOTIFY"
/ "REDIRECT"
/ "SETUP"
/ "SET_PARAMETER"
/ "TEARDOWN"
/ extension-method
extension-method = token
Request-URI = "*" / RTSP-REQ-URI
RTSP-Version = "RTSP/" 1*DIGIT "." 1*DIGIT
message-body-data = 1*OCTET
Status-Code = "100" ; Continue
/ "200" ; OK
/ "301" ; Moved Permanently
/ "302" ; Found
/ "303" ; See Other
/ "304" ; Not Modified
/ "305" ; Use Proxy
/ "400" ; Bad Request
/ "401" ; Unauthorized
/ "402" ; Payment Required
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/ "403" ; Forbidden
/ "404" ; Not Found
/ "405" ; Method Not Allowed
/ "406" ; Not Acceptable
/ "407" ; Proxy Authentication Required
/ "408" ; Request Time-out
/ "410" ; Gone
/ "411" ; Length Required
/ "412" ; Precondition Failed
/ "413" ; Request Message Body Too Large
/ "414" ; Request-URI Too Large
/ "415" ; Unsupported Media Type
/ "451" ; Parameter Not Understood
/ "452" ; reserved
/ "453" ; Not Enough Bandwidth
/ "454" ; Session Not Found
/ "455" ; Method Not Valid in This State
/ "456" ; Header Field Not Valid for Resource
/ "457" ; Invalid Range
/ "458" ; Parameter Is Read-Only
/ "459" ; Aggregate operation not allowed
/ "460" ; Only aggregate operation allowed
/ "461" ; Unsupported Transport
/ "462" ; Destination Unreachable
/ "463" ; Destination Prohibited
/ "464" ; Data Transport Not Ready Yet
/ "465" ; Notification Reason Unknown
/ "466" ; Key Management Error
/ "470" ; Connection Authorization Required
/ "471" ; Connection Credentials not accepted
/ "472" ; Failure to establish secure connection
/ "500" ; Internal Server Error
/ "501" ; Not Implemented
/ "502" ; Bad Gateway
/ "503" ; Service Unavailable
/ "504" ; Gateway Time-out
/ "505" ; RTSP Version not supported
/ "551" ; Option not supported
/ extension-code
extension-code = 3DIGIT
Reason-Phrase = 1*(TEXT-UTF8char / HT / SP)
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general-header = Cache-Control
/ Connection
/ CSeq
/ Date
/ Media-Properties
/ Media-Range
/ Pipelined-Requests
/ Proxy-Supported
/ Seek-Style
/ Server
/ Supported
/ Timestamp
/ User-Agent
/ Via
/ extension-header
request-header = Accept
/ Accept-Credentials
/ Accept-Encoding
/ Accept-Language
/ Authorization
/ Bandwidth
/ Blocksize
/ From
/ If-Match
/ If-Modified-Since
/ If-None-Match
/ Notify-Reason
/ Proxy-Require
/ Range
/ Referrer
/ Request-Status
/ Require
/ Scale
/ Session
/ Speed
/ Supported
/ Terminate-Reason
/ Transport
/ extension-header
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response-header = Accept-Credentials
/ Accept-Ranges
/ Connection-Credentials
/ MTag
/ Location
/ Proxy-Authenticate
/ Public
/ Range
/ Retry-After
/ RTP-Info
/ Scale
/ Session
/ Speed
/ Transport
/ Unsupported
/ WWW-Authenticate
/ extension-header
message-header = Allow
/ Content-Base
/ Content-Encoding
/ Content-Language
/ Content-Length
/ Content-Location
/ Content-Type
/ Expires
/ Last-Modified
/ extension-header
20.2.3. Header Syntax
Accept = "Accept" HCOLON
[ accept-range *(COMMA accept-range) ]
accept-range = media-type-range [SEMI accept-params]
media-type-range = ( "*/*"
/ ( m-type SLASH "*" )
/ ( m-type SLASH m-subtype )
) *( SEMI m-parameter )
accept-params = "q" EQUAL qvalue *(SEMI generic-param )
qvalue = ( "0" [ "." *3DIGIT ] )
/ ( "1" [ "." *3("0") ] )
Accept-Credentials = "Accept-Credentials" HCOLON cred-decision
cred-decision = ("User" [LWS cred-info])
/ "Proxy"
/ "Any"
/ (token [LWS 1*header-value])
; For future extensions
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cred-info = cred-info-data *(COMMA cred-info-data)
cred-info-data = DQUOTE RTSP-REQ-URI DQUOTE SEMI hash-alg
SEMI base64
hash-alg = "sha-256" / extension-alg
extension-alg = token
Accept-Encoding = "Accept-Encoding" HCOLON
[ encoding *(COMMA encoding) ]
encoding = codings [SEMI accept-params]
codings = content-coding / "*"
content-coding = token
Accept-Language = "Accept-Language" HCOLON
language *(COMMA language)
language = language-range [SEMI accept-params]
language-range = language-tag / "*"
language-tag = primary-tag *( "-" subtag )
primary-tag = 1*8ALPHA
subtag = 1*8ALPHA
Accept-Ranges = "Accept-Ranges" HCOLON acceptable-ranges
acceptable-ranges = (range-unit *(COMMA range-unit))
range-unit = "NPT" / "SMPTE" / "UTC" / extension-format
extension-format = token
Allow = "Allow" HCOLON Method *(COMMA Method)
Authorization = "Authorization" HCOLON credentials
credentials = ("Digest" LWS digest-response)
/ other-response
digest-response = dig-resp *(COMMA dig-resp)
dig-resp = username / realm / nonce / digest-uri
/ dresponse / algorithm / cnonce
/ opaque / message-qop
/ nonce-count / auth-param
username = "username" EQUAL username-value
username-value = quoted-string
digest-uri = "uri" EQUAL LDQUOT digest-uri-value RDQUOT
digest-uri-value = RTSP-REQ-URI
message-qop = "qop" EQUAL qop-value
cnonce = "cnonce" EQUAL cnonce-value
cnonce-value = nonce-value
nonce-count = "nc" EQUAL nc-value
nc-value = 8LHEX
dresponse = "response" EQUAL request-digest
request-digest = LDQUOT 32LHEX RDQUOT
auth-param = auth-param-name EQUAL
( token / quoted-string )
auth-param-name = token
other-response = auth-scheme LWS auth-param
*(COMMA auth-param)
auth-scheme = token
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Bandwidth = "Bandwidth" HCOLON 1*19DIGIT
Blocksize = "Blocksize" HCOLON 1*9DIGIT
Cache-Control = "Cache-Control" HCOLON cache-directive
*(COMMA cache-directive)
cache-directive = cache-rqst-directive
/ cache-rspns-directive
cache-rqst-directive = "no-cache"
/ "max-stale" [EQUAL delta-seconds]
/ "min-fresh" EQUAL delta-seconds
/ "only-if-cached"
/ cache-extension
cache-rspns-directive = "public"
/ "private"
/ "no-cache"
/ "no-transform"
/ "must-revalidate"
/ "proxy-revalidate"
/ "max-age" EQUAL delta-seconds
/ cache-extension
cache-extension = token [EQUAL (token / quoted-string)]
delta-seconds = 1*19DIGIT
Connection = "Connection" HCOLON connection-token
*(COMMA connection-token)
connection-token = "close" / token
Connection-Credentials = "Connection-Credentials" HCOLON cred-chain
cred-chain = DQUOTE RTSP-REQ-URI DQUOTE SEMI base64
Content-Base = "Content-Base" HCOLON RTSP-URI
Content-Encoding = "Content-Encoding" HCOLON
content-coding *(COMMA content-coding)
Content-Language = "Content-Language" HCOLON
language-tag *(COMMA language-tag)
Content-Length = "Content-Length" HCOLON 1*19DIGIT
Content-Location = "Content-Location" HCOLON RTSP-REQ-Ref
Content-Type = "Content-Type" HCOLON media-type
media-type = m-type SLASH m-subtype *(SEMI m-parameter)
m-type = discrete-type / composite-type
discrete-type = "text" / "image" / "audio" / "video"
/ "application" / extension-token
composite-type = "message" / "multipart" / extension-token
extension-token = ietf-token / x-token
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ietf-token = token
x-token = "x-" token
m-subtype = extension-token / iana-token
iana-token = token
m-parameter = m-attribute EQUAL m-value
m-attribute = token
m-value = token / quoted-string
CSeq = "CSeq" HCOLON cseq-nr
cseq-nr = 1*9DIGIT
Date = "Date" HCOLON RTSP-date
RTSP-date = rfc1123-date ; HTTP-date
rfc1123-date = wkday "," SP date1 SP time SP "GMT"
date1 = 2DIGIT SP month SP 4DIGIT
; day month year (e.g., 02 Jun 1982)
time = 2DIGIT ":" 2DIGIT ":" 2DIGIT
; 00:00:00 - 23:59:59
wkday = "Mon" / "Tue" / "Wed"
/ "Thu" / "Fri" / "Sat" / "Sun"
month = "Jan" / "Feb" / "Mar" / "Apr"
/ "May" / "Jun" / "Jul" / "Aug"
/ "Sep" / "Oct" / "Nov" / "Dec"
Expires = "Expires" HCOLON RTSP-date
From = "From" HCOLON from-spec
from-spec = ( name-addr / addr-spec ) *( SEMI from-param )
name-addr = [ display-name ] LAQUOT addr-spec RAQUOT
addr-spec = RTSP-REQ-URI / absolute-URI
absolute-URI = < As defined in RFC 3986>
display-name = *(token LWS) / quoted-string
from-param = tag-param / generic-param
tag-param = "tag" EQUAL token
If-Match = "If-Match" HCOLON ("*" / message-tag-list)
message-tag-list = message-tag *(COMMA message-tag)
message-tag = [ weak ] opaque-tag
weak = "W/"
opaque-tag = quoted-string
If-Modified-Since = "If-Modified-Since" HCOLON RTSP-date
If-None-Match = "If-None-Match" HCOLON ("*" / message-tag-list)
Last-Modified = "Last-Modified" HCOLON RTSP-date
Location = "Location" HCOLON RTSP-REQ-URI
Media-Properties = "Media-Properties" HCOLON [media-prop-list]
media-prop-list = media-prop-value *(COMMA media-prop-value)
media-prop-value = ("Random-Access" [EQUAL POS-FLOAT])
/ "Begining-Only"
/ "No-Seeking"
/ "Immutable"
/ "Dynamic"
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/ "Time-Progressing"
/ "Unlimited"
/ ("Time-Limited" EQUAL utc-time)
/ ("Time-Duration" EQUAL POS-FLOAT)
/ ("Scales" EQUAL scale-value-list)
/ media-prop-ext
media-prop-ext = token [EQUAL (1*rtsp-unreserved / quoted-string)]
scale-value-list = DQUOTE scale-entry *(COMMA scale-entry) DQUOTE
scale-entry = scale-value / (scale-value COLON scale-value)
scale-value = FLOAT
Media-Range = "Media-Range" HCOLON [ranges-list]
ranges-list = ranges-spec *(COMMA ranges-spec)
MTag = "MTag" HCOLON message-tag
Notify-Reason = "Notify-Reason" HCOLON Notify-Reas-val
Notify-Reas-val = "end-of-stream"
/ "media-properties-update"
/ "scale-change"
/ Notify-Reason-extension
Notify-Reason-extension = token
Pipelined-Requests = "Pipelined-Requests" HCOLON startup-id
startup-id = 1*8DIGIT
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Proxy-Authenticate = "Proxy-Authenticate" HCOLON challenge-list
challenge-list = challenge *(COMMA challenge)
challenge = ("Digest" LWS digest-cln *(COMMA digest-cln))
/ other-challenge
other-challenge = auth-scheme LWS auth-param
*(COMMA auth-param)
digest-cln = realm / domain / nonce
/ opaque / stale / algorithm
/ qop-options / auth-param
realm = "realm" EQUAL realm-value
realm-value = quoted-string
domain = "domain" EQUAL LDQUOT RTSP-REQ-Ref
*(1*SP RTSP-REQ-Ref ) RDQUOT
nonce = "nonce" EQUAL nonce-value
nonce-value = quoted-string
opaque = "opaque" EQUAL quoted-string
stale = "stale" EQUAL ( "true" / "false" )
algorithm = "algorithm" EQUAL ("MD5" / "MD5-sess" / token)
qop-options = "qop" EQUAL LDQUOT qop-value
*("," qop-value) RDQUOT
qop-value = "auth" / "auth-int" / token
Proxy-Require = "Proxy-Require" HCOLON feature-tag-list
feature-tag-list = feature-tag *(COMMA feature-tag)
Proxy-Supported = "Proxy-Supported" HCOLON [feature-tag-list]
Public = "Public" HCOLON Method *(COMMA Method)
Range = "Range" HCOLON ranges-spec
ranges-spec = npt-range / utc-range / smpte-range
/ range-ext
range-ext = extension-format ["=" range-value]
range-value = 1*(rtsp-unreserved / quoted-string / ":" )
Referrer = "Referrer" HCOLON (absolute-URI / RTSP-URI-Ref)
Request-Status = "Request-Status" HCOLON req-status-info
req-status-info = cseq-info LWS status-info LWS reason-info
cseq-info = "cseq" EQUAL cseq-nr
status-info = "status" EQUAL Status-Code
reason-info = "reason" EQUAL DQUOTE Reason-Phrase DQUOTE
Require = "Require" HCOLON feature-tag-list
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RTP-Info = "RTP-Info" HCOLON [rtsp-info-spec
*(COMMA rtsp-info-spec)]
rtsp-info-spec = stream-url 1*ssrc-parameter
stream-url = "url" EQUAL DQUOTE RTSP-REQ-Ref DQUOTE
ssrc-parameter = LWS "ssrc" EQUAL ssrc HCOLON
ri-parameter *(SEMI ri-parameter)
ri-parameter = ("seq" EQUAL 1*5DIGIT)
/ ("rtptime" EQUAL 1*10DIGIT)
/ generic-param
Retry-After = "Retry-After" HCOLON (RTSP-date / delta-seconds)
Scale = "Scale" HCOLON scale-value
Seek-Style = "Seek-Style" HCOLON Seek-S-values
Seek-S-values = "RAP"
/ "CoRAP"
/ "First-Prior"
/ "Next"
/ Seek-S-value-ext
Seek-S-value-ext = token
Server = "Server" HCOLON ( product / comment )
*(LWS (product / comment))
product = token [SLASH product-version]
product-version = token
comment = LPAREN *( ctext / quoted-pair) RPAREN
Session = "Session" HCOLON session-id
[ SEMI "timeout" EQUAL delta-seconds ]
Speed = "Speed" HCOLON lower-bound MINUS upper-bound
lower-bound = POS-FLOAT
upper-bound = POS-FLOAT
Supported = "Supported" HCOLON [feature-tag-list]
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Terminate-Reason = "Terminate-Reason" HCOLON TR-Info
TR-Info = TR-Reason *(SEMI TR-Parameter)
TR-Reason = "Session-Timeout"
/ "Server-Admin"
/ "Internal-Error"
/ token
TR-Parameter = TR-time / TR-user-msg / generic-param
TR-time = "time" EQUAL utc-time
TR-user-msg = "user-msg" EQUAL quoted-string
Timestamp = "Timestamp" HCOLON timestamp-value [LWS delay]
timestamp-value = *19DIGIT [ "." *9DIGIT ]
delay = *9DIGIT [ "." *9DIGIT ]
Transport = "Transport" HCOLON transport-spec
*(COMMA transport-spec)
transport-spec = transport-id *trns-parameter
transport-id = trans-id-rtp / other-trans
trans-id-rtp = "RTP/" profile ["/" lower-transport]
; no LWS is allowed inside transport-id
other-trans = token *("/" token)
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profile = "AVP" / "SAVP" / "AVPF" / "SAVPF" / token
lower-transport = "TCP" / "UDP" / token
trns-parameter = (SEMI ( "unicast" / "multicast" ))
/ (SEMI "interleaved" EQUAL channel ["-" channel])
/ (SEMI "ttl" EQUAL ttl)
/ (SEMI "layers" EQUAL 1*DIGIT)
/ (SEMI "ssrc" EQUAL ssrc *(SLASH ssrc))
/ (SEMI "mode" EQUAL mode-spec)
/ (SEMI "dest_addr" EQUAL addr-list)
/ (SEMI "src_addr" EQUAL addr-list)
/ (SEMI "setup" EQUAL contrans-setup)
/ (SEMI "connection" EQUAL contrans-con)
/ (SEMI "RTCP-mux")
/ (SEMI "MIKEY" EQUAL MIKEY-Value)
/ (SEMI trn-param-ext)
contrans-setup = "active" / "passive" / "actpass"
contrans-con = "new" / "existing"
trn-param-ext = par-name [EQUAL trn-par-value]
par-name = token
trn-par-value = *(rtsp-unreserved / quoted-string)
ttl = 1*3DIGIT ; 0 to 255
ssrc = 8HEX
channel = 1*3DIGIT ; 0 to 255
MIKEY-Value = base64
mode-spec = ( DQUOTE mode *(COMMA mode) DQUOTE )
mode = "PLAY" / token
addr-list = quoted-addr *(SLASH quoted-addr)
quoted-addr = DQUOTE (host-port / extension-addr) DQUOTE
host-port = ( host [":" port] )
/ ( ":" port )
extension-addr = 1*qdtext
host = < As defined in RFC 3986>
port = < As defined in RFC 3986>
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Unsupported = "Unsupported" HCOLON feature-tag-list
User-Agent = "User-Agent" HCOLON ( product / comment )
*(LWS (product / comment))
field-name-list = field-name *(COMMA field-name)
field-name = token
Via = "Via" HCOLON via-parm *(COMMA via-parm)
via-parm = sent-protocol LWS sent-by *( SEMI via-params )
via-params = via-ttl / via-maddr
/ via-received / via-extension
via-ttl = "ttl" EQUAL ttl
via-maddr = "maddr" EQUAL host
via-received = "received" EQUAL (IPv4address / IPv6address)
IPv4address = < As defined in RFC 3986>
IPv6address = < As defined in RFC 3986>
via-extension = generic-param
sent-protocol = protocol-name SLASH protocol-version
SLASH transport-prot
protocol-name = "RTSP" / token
protocol-version = token
transport-prot = "UDP" / "TCP" / "TLS" / other-transport
other-transport = token
sent-by = host [ COLON port ]
WWW-Authenticate = "WWW-Authenticate" HCOLON challenge-list
20.3. SDP extension Syntax
This section defines in ABNF the SDP extensions defined for RTSP.
See Appendix D for the definition of the extensions in text.
control-attribute = "a=control:" *SP RTSP-REQ-Ref CRLF
a-range-def = "a=range:" ranges-spec CRLF
a-mtag-def = "a=mtag:" message-tag CRLF
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21. Security Considerations
The security considerations and threats around RTSP and its usage can
be divided into considerations around the signaling protocol itself
and the issues related to the media stream delivery. However, when
it comes to mitigations of security threats, a threat depending on
the media stream delivery may in fact be mitigated by a mechanism in
the signaling protocol.
There are several chapters and appendix in this document that define
security solutions for the protocol. We will reference them when
discussing the threats below. But the reader should take special
notice of the Security Framework (Section 19) and the specification
of how to use SRTP and its key-mangement (Appendix C.1.4) to achieve
certain aspects of the media security.
21.1. Signaling Protocol Threats
This section focuses on issues related to the signaling protocol.
Because of the similarity in syntax and usage between RTSP servers
and HTTP servers, the security considerations outlined in [H15] apply
also.
Specifically, please note the following:
Abuse of Server Log Information: RTSP and HTTP servers will
presumably have similar logging mechanisms, and thus should be
equally guarded in protecting the contents of those logs, thus
protecting the privacy of the users of the servers. See
[H15.1.1] for HTTP server recommendations regarding server
logs.
Transfer of Sensitive Information: There is no reason to believe
that information transferred or controlled via RTSP may be any
less sensitive than that normally transmitted via HTTP.
Therefore, all of the precautions regarding the protection of
data privacy and user privacy apply to implementors of RTSP
clients, servers, and proxies. See [H15.1.2] for further
details.
Attacks Based On File and Path Names: Though RTSP URIs are opaque
handles that do not necessarily have file system semantics, it
is anticipated that many implementations will translate
portions of the Request-URIs directly to file system calls. In
such cases, file systems SHOULD follow the precautions outlined
in [H15.5], such as checking for ".." in path components.
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Personal Information: RTSP clients are often privy to the same
information that HTTP clients are (user name, location, etc.)
and thus should be equally sensitive. See [H15.1] for further
recommendations.
Privacy Issues Connected to Accept Headers: Since may of the same
"Accept" headers exist in RTSP as in HTTP, the same caveats
outlined in [H15.1.4] with regards to their use should be
followed.
DNS Spoofing: Presumably, given the longer connection times
typically associated to RTSP sessions relative to HTTP
sessions, RTSP client DNS optimizations should be less
prevalent. Nonetheless, the recommendations provided in
[H15.3] are still relevant to any implementation which attempts
to rely on a DNS-to-IP mapping to hold beyond a single use of
the mapping.
Location Headers and Spoofing: If a single server supports multiple
organizations that do not trust each another, then it MUST
check the values of Location and Content-Location header fields
in responses that are generated under control of said
organizations to make sure that they do not attempt to
invalidate resources over which they have no authority.
([H15.4])
In addition to the recommendations in the current HTTP specification
(RFC 2616 [RFC2616], as of this writing) and also of the previous RFC
2068 [RFC2068], future HTTP specifications may provide additional
guidance on security issues.
The following are added considerations for RTSP implementations.
Session hijacking: Since there is no or little relation between a
transport layer connection and an RTSP session, it is possible
for a malicious client to issue requests with random session
identifiers which could affect other clients of an unsuspecting
server. To mitigate this the server SHALL use a large, random
and non-sequential session identifier to minimize the
possibility of this kind of attack. However, unless the RTSP
signaling is always confidentiality protected, e.g. using TLS,
an on-path attacker will be able to hijack a session. To
prevent session hijacking client authentication needs to be
performed and only the authenticated client creating the
session SHALL be able to access that session.
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Authentication: Servers SHOULD implement both basic and digest
[RFC2617] authentication. In environments requiring tighter
security for the control messages, the transport layer
mechanism TLS [RFC5246] SHOULD be used.
Persistently suspicious behavior: RTSP servers SHOULD return error
code 403 (Forbidden) upon receiving a single instance of
behavior which is deemed a security risk. RTSP servers SHOULD
also be aware of attempts to probe the server for weaknesses
and entry points and MAY arbitrarily disconnect and ignore
further requests from clients which are deemed to be in
violation of local security policy.
TLS through proxies: If one uses the possibility to connect TLS in
multiple legs (Section 19.3) one really needs to be aware of
the trust model. That procedure requires full faith and trust
in all proxies, which will be identified, that one allows to
connect through. They are men in the middle and have access to
all that goes on over the TLS connection. Thus it is important
to consider if that trust model is acceptable in the actual
application. Further discussion of the actual trust model is
in Section 19.3.
Resource Exhaustion: As RTSP is a stateful protocol and establishes
resource usage on the server there is a clear possibility to
attack the server by trying to overbook these resources to
perform a denial of service attack. This attack can be both
against ongoing sessions and to prevent others from
establishing sessions. RTSP agents will need to have
mechanisms to prevent single peers from consuming extensive
amounts of resources. The methods for guarding against this
are varied and depends on the agent's role and capabilities and
policies. Each implementation has to carefully consider their
methods and policies to mitigate this threat. For example
regarding handling of connections there are recommendations in
Section 10.7.
The above threats and considerations have resulted in a set of
security functions and mechanisms built into or used by the protocol.
The signaling protocol relies on two security features defined in the
Security Framework (Section 19) namely client authentication using
HTTP authentication and TLS based transport protection of the
signaling messages. Both of these mechanisms are required to be
implemented by any RTSP agent.
A number of different security mitigations have been designed into
the protocol and will be present by following the specification as
written, for example by ensuring sufficient amount of entropy in the
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randomly generated session identifiers when not using client
authentication to prevent session hijacking. When client
authentication is used the protection against hijacking will be
strongly improved by scoping the accessible sessions to the one this
client identity has created. Some of the above threats are such that
the implementation of the RTSP functionality itself needs to consider
which policy and strategy it uses to mitigate them.
21.2. Media Stream Delivery Threats
The fact that RTSP establishes and controls a media stream delivery
results in a set of security issues related to the media streams.
This section will attempt to analyze general threats, however the
choice of media stream transport protocol, like RTP will result in
some differences in threats and what mechanisms that exist to
mitigate them. Thus it becomes important that each specification of
a new media stream transport and delivery protocol usable by RTSP
requires its own security analysis. This section includes one for
RTP.
The set of general threats from or by the media stream delivery
itself are:
Concentrated denial-of-service attack: The protocol offers the
opportunity for a remote-controlled denial-of-service (DoS)
attack, where the media stream is the hammer in that DoS attack.
See Section 21.2.1.
Media Confidentiality: The media delivery may contain content of any
type and it is not possible in general to determine how sensitive
this content is from a confidentiality point. Thus it is a strong
requirement that any media delivery protocol provides a method for
providing confidentiality of the actual media content. In
addition to the media level confidentiality it becomes critical
that no resource identifiers used in the signaling are exposed to
an attacker as they may have human understandable names, or may be
also available to the attacker so they can determine the content
the user was delivered. Thus also the signaling protocol must
provide confidentiality protection of any information related to
the media resource.
Media Integrity and Authentication: There are several reasons, such
as discrediting the target, misinformation of the target, why an
attacker will have interest to substitute the media stream sent
out from the RTSP server with one of the attacker's creation or
selection. Therefore it is important that the media protocol
provides mechanisms to verify the source authentication, integrity
and prevent replay attacks on the media stream.
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Scope of Multicast: If RTSP is used to control the transmission of
media onto a multicast network it is needed to consider the scope
that delivery has. RTSP supports the TTL Transport header
parameter to indicate this scope for IPv4. IPv6 has a different
mechanism for scope boundary. However, such scope control has
risks, as it may be set too large and distribute media beyond the
intended scope.
Below (Section 21.2.2) we do a protocol specific analysis of security
considerations for RTP based media transport. In that section we
also make clear the requirements on implementing security functions
for RTSP agents supporting media delivery over RTP.
21.2.1. Remote Denial of Service Attack
The attacker may initiate traffic flows to one or more IP addresses
by specifying them as the destination in SETUP requests. While the
attacker's IP address may be known in this case, this is not always
useful in prevention of more attacks or ascertaining the attackers
identity. Thus, an RTSP server MUST only allow client-specified
destinations for RTSP-initiated traffic flows if the server has
ensured that the specified destination address accepts receiving
media through different security mechanisms. Security mechanisms
that are acceptable in an increased generality are:
o Verification of the client's identity against a database of known
users using RTSP authentication mechanisms (preferably digest
authentication or stronger)
o A list of addresses that accept to be media destinations,
especially considering user identity
o Media path based verification
The server SHOULD NOT allow the destination field to be set unless a
mechanism exists in the system to authorize the request originator to
direct streams to the recipient. It is preferred that this
authorization be performed by the media recipient (destination)
itself and the credentials passed along to the server. However, in
certain cases, such as when the recipient address is a multicast
group, or when the recipient is unable to communicate with the server
in an out-of-band manner, this may not be possible. In these cases
the server may chose another method such as a server-resident
authorization list to ensure that the request originator has the
proper credentials to request stream delivery to the recipient.
One solution that performs the necessary verification of acceptance
of media suitable for unicast based delivery is the ICE based NAT
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traversal method described in [I-D.ietf-mmusic-rtsp-nat]. This
mechanism uses random passwords and username so that the probability
of unintended indication as a valid media destination is very low.
In addition the server includes in its STUN requests a cookie
(consisting of random material) that the destination echoes back,
thus the solution also safe-guards against having an off-path
attacker being able to spoof the STUN checks. This leaves this
solution vulnerable only to on-path attackers that can see the STUN
requests go to the target of attack and thus forge a response.
For delivery to multicast addresses there is a need for another
solution which is not specified in this memo.
21.2.2. RTP Security analysis
RTP is a commonly used media transport protocol and has been the most
common choice for RTSP 1.0 implementations. The core RTP protocol
has been in use for a long time and it has well-known security
properties and the RTP security consideration (Section 9 of
[RFC3550]) needs to be reviewed. In perspective of the usage of RTP
in context of RTSP the following properties should be noted:
Stream Additions: RTP has support for multiple simultaneous media
streams in each RTP session. As some use cases require support
for non-synchronized adding and removal of media streams and their
identifiers an attacker can easily insert additional media streams
into a session context that according to protocol design is
intended to be played out. Another threat vector is one of denial
of service by exhausting the resources of the RTP session
receiver, for example by using a large number of SSRC identifiers
simultaneously. The strong mitigation of this is to ensure that
one cryptographically authenticates any incoming packet flow to
the RTP session. Weak mitigations like blocking additional media
streams in session contexts easily lead to a denial of service
vulnerability in addition to preventing certain RTP extensions or
use cases which rely on multiple media streams, such as RTP
retransmission [RFC4588] to function.
Forged Feedback: The built in RTP control Protocol (RTCP) also
offers a large attack surface for a couple of different types of
attacks. One venue is to send RTCP feedback to the media sender
indicating large amounts of packet loss and thus trigger a media
bit-rate adaptation response from the sender resulting in lowered
media quality and potentially shut down of the media stream.
Another attack is to perform a resource exhaustion attack on the
receiver by using many SSRC identifiers to create large state
tables and increase the RTCP related processing demands.
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RTP/RTCP Extensions: RTP and RTCP extensions generally provide
additional and sometimes extremely powerful tools to do denial of
service or service disruption. For example the Code Control
Message [RFC5104] RTCP extensions enables both locking down the
bit-rate to low values and disrupt video quality by requesting
Intra frames.
Taking into account the above general discussion in Section 21.2 and
the RTP specific discussion in this section it is clear that strong
security mechanism to protect RTP is necessary to support. Therefore
this specification has the following requirements on RTP security
functions for all RTSP agents that handles media streams and where
the media stream transport is done using RTP.
RTSP agents supporting RTP MUST implement Secure RTP (SRTP) [RFC3711]
and thus the SAVP profile. In addition the secure AVP profile
(SAVPF) [RFC5124] MUST also be supported if the AVPF profile is
implemented. This specification requires no additional crypto
transforms or configuration values beyond the mandatory to implement
in RFC3711, i.e. AES-CM and HMAC-SHA1. The default key-management
mechanism which MUST be implemented is the one defined in the MIKEY
Key Establishment (Appendix C.1.4.1). The MIKEY implementation MUST
implement the necessary functions for MIKEY-RSA-R mode [RFC4738] and
in addition the SRTP parameter negotiation necessary to negotiate the
supported SRTP transforms and parameters.
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22. IANA Considerations
This section sets up a number of registries for RTSP 2.0 that should
be maintained by IANA. These registries are separate from any
registries existing for RTSP 1.0. For each registry there is a
description of what it is required to contain, what specification is
needed when adding an entry with IANA, and finally the entries that
this document needs to register. See also the Section 2.7 "Extending
RTSP". There is also an IANA registration of three SDP attributes.
Registries or entries in registries which have been made for RTSP 1.0
are not moved to RTSP 2.0. The registries and entries in registries
of RTSP 1.0 and RTSP 2.0 are independent. If any registry or entry
in a registry is also required in RTSP 2.0, it MUST follow the below
defined procedure to allocate the registry or entry in a registry.
The sections describing how to register an item uses some of the
requirements level described in RFC 5226 [RFC5226], namely "First
Come, First Served", "Expert Review, "Specification Required", and
"Standards Action".
In case a registry requires a contact person, the authors are the
contact person for any entries created by this document.
A registration request to IANA MUST contain the following
information:
o A name of the item to register according to the rules specified by
the intended registry.
o Indication of who has change control over the feature (for
example, IETF, ISO, ITU-T, other international standardization
bodies, a consortium, a particular company or group of companies,
or an individual);
o A reference to a further description, if available, for example
(in decreasing order of preference) an RFC, a published standard,
a published paper, a patent filing, a technical report, documented
source code or a computer manual;
o For proprietary features, contact information (postal and email
address);
22.1. Feature-tags
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22.1.1. Description
When a client and server try to determine what part and functionality
of the RTSP specification and any future extensions that its counter
part implements there is need for a namespace. This registry
contains named entries representing certain functionality.
The usage of feature-tags is explained in Section 11 and
Section 13.1.
22.1.2. Registering New Feature-tags with IANA
The registering of feature-tags is done on a first come, first served
basis.
The name of the feature MUST follow these rules: The name may be of
any length, but SHOULD be no more than twenty characters long. The
name MUST NOT contain any spaces, or control characters. The
registration MUST indicate if the feature-tag applies to clients,
servers, or proxies only or any combinations of these. Any
proprietary feature MUST have as the first part of the name a vendor
tag, which identifies the organization. The registry entries consist
of the feature tag, a one paragraph description of what it
represents, its applicability (server, client, proxy, any
combination) and a reference to its specification where applicable.
Examples for a vendor tag describing a proprietary feature are:
vendorA.specfeat01
vendorA.specfeat02
22.1.3. Registered entries
The following feature-tags are defined in this specification and
hereby registered. The change control belongs to the IETF.
play.basic: The implementation for delivery and playback operations
according to the core RTSP specification, as defined in this
memo. Applies for both clients, servers and proxies.
play.scale: Support of scale operations for media playback. Applies
only for servers.
play.speed: Support of the speed functionality for media delivery.
Applies only for servers.
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setup.rtp.rtcp.mux Support of the RTP and RTCP multiplexing as
discussed in Appendix C.1.6.4. Applies for both client and
servers and any media caching proxy.
This should be represented by IANA as a table with the feature tags,
contact person and their references.
22.2. RTSP Methods
22.2.1. Description
Methods are described in Section Section 13. Extending the protocol
with new methods allow for totally new functionality.
22.2.2. Registering New Methods with IANA
A new method MUST be registered through an IETF Standards Action.
The reason is that new methods may radically change the protocol's
behavior and purpose.
A specification for a new RTSP method MUST consist of the following
items:
o A method name which follows the ABNF rules for methods.
o A clear specification what a request using the method does and
what responses are expected. Which directions the method is used,
C->S or S->C or both. How the use of headers, if any, modifies
the behavior and effect of the method.
o A list or table specifying which of the IANA registered headers
that are allowed to be used with the method in request or/and
response. The list or table SHOULD follow the format of tables in
Section 18.
o Describe how the method relates to network proxies.
22.2.3. Registered Entries
This specification, RFCXXXX, registers 10 methods: DESCRIBE,
GET_PARAMETER, OPTIONS, PAUSE, PLAY, PLAY_NOTIFY, REDIRECT, SETUP,
SET_PARAMETER, and TEARDOWN. The initial table of the registry is
provided below.
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Method Directionality Reference
-----------------------------------------------------
DESCRIBE C->S [RFCXXXX]
GET_PARAMETER C->S, S->C [RFCXXXX]
OPTIONS C->S, S->C [RFCXXXX]
PAUSE C->S [RFCXXXX]
PLAY C->S [RFCXXXX]
PLAY_NOTIFY S->C [RFCXXXX]
REDIRECT S->C [RFCXXXX]
SETUP C->S [RFCXXXX]
SET_PARAMETER C->S, S->C [RFCXXXX]
TEARDOWN C->S, S->C [RFCXXXX]
22.3. RTSP Status Codes
22.3.1. Description
A status code is the three digit number used to convey information in
RTSP response messages, see Section 8. The number space is limited
and care should be taken not to fill the space.
22.3.2. Registering New Status Codes with IANA
A new status code registration follows the policy of IETF Review. A
specification for a new status code MUST specify the following:
o The registered number.
o A description of what the status code means and the expected
behavior of the sender and receiver of the code.
22.3.3. Registered Entries
RFCXXXX, registers the numbered status code defined in the ABNF entry
"Status-Code" except "extension-code" (that defines the syntax
allowed for future extensions) in Section 20.2.2.
22.4. RTSP Headers
22.4.1. Description
By specifying new headers a method(s) can be enhanced in many
different ways. An unknown header will be ignored by the receiving
agent. If the new header is vital for a certain functionality, a
feature-tag for the functionality can be created and demanded to be
used by the counter-part with the inclusion of a Require header
carrying the feature-tag.
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22.4.2. Registering New Headers with IANA
Registrations in the registry can be done following the Expert Review
policy. A specification SHOULD be provided, preferably an IETF RFC
or other Standards Developing Organization specification. The
minimal information in a registration request is the header name and
the contact information.
The specification SHOULD contain the following information:
o The name of the header.
o An ABNF specification of the header syntax.
o A list or table specifying when the header may be used,
encompassing all methods, their request or response, the direction
(C->S or S->C).
o How the header is to be handled by proxies.
o A description of the purpose of the header.
22.4.3. Registered entries
All headers specified in Section 18 in RFCXXXX are to be registered.
The Registry is to include header name and reference.
Furthermore the following legacy RTSP headers defined in other
specifications are registered with header name, reference and
description according to below list. Note: These references may not
fulfill all of the above rules for registrations due to their legacy
status.
o x-wap-profile defined in [TS-26234]. The x-wap-profile request
header contains one or more absolute URLs to the requesting
agent's device capability profile.
o x-wap-profile-diff defined in [TS-26234]. The x-wap-profile-diff
request header contains a subset of a device capability profile.
o x-wap-profile-warning defined in [TS-26234]. The x-wap-profile-
warning is a response header that contains error codes explaining
to what extent the server has been able to match the terminal
request in regards to device capability profile as described using
x-wap-profile and x-wap-profile-diff headers.
o x-predecbufsize defined in [TS-26234]. This response header
provides an RTSP agent with the TS 26.234 Annex G hypothetical
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pre-decoder buffer size.
o x-initpredecbufperiod defined in [TS-26234]. This response header
provides an RTSP agent with the TS 26.234 Annex G hypothetical
pre-decoder buffering period.
o x-initpostdecbufperiod defined in [TS-26234]. This response
header provides an RTSP agent with the TS 26.234 Annex G post-
decoder buffering period.
o 3gpp-videopostdecbufsize defined in [TS-26234]. This response
header provides an RTSP agent with the TS 26.234 defined post-
decoder buffer size usable for H.264 (AVC) video streams.
o 3GPP-Link-Char defined in [TS-26234]. This request header
provides the RTSP server with the RTSP client's link
charateristics as deterimined from the radio interface. The
information that can be provided are guaranteed bit-rate, maximum
bit-rate and maximum transfer delay.
o 3GPP-Adaptation defined in [TS-26234]. This general header is
part of the bit-rate adaptation solution specified for PSS. It
provides the RTSP client's buffer sizes and target buffer levels
to the server and responses are used to acknowledge the support
and values.
o 3GPP-QoE-Metrics defined in [TS-26234]. This general header is
used by PSS RTSP agents to negotiate the quality of experince
metrics that a client should gather and report to the server.
o 3GPP-QoE-Feedback defined in [TS-26234]. This request header is
used by RTSP clients supporting PSS to report the actual values of
the metrics gathered in its quality of experince metering.
The use of "x-" is NOT RECOMMENDED but the above headers in the
register list were defined prior to the clarification.
22.5. Accept-Credentials
The security framework's TLS connection mechanism has two registrable
entities.
22.5.1. Accept-Credentials policies
In Section 19.3.1 three policies for how to handle certificates are
specified. Further policies may be defined and MUST be registered
with IANA using the following rules:
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o Registering requires an IETF Standards Action
o A registration is required to name a contact person.
o Name of the policy.
o A describing text that explains how the policy works for handling
the certificates.
This specification registers the following values:
Any
Proxy
User
22.5.2. Accept-Credentials hash algorithms
The Accept-Credentials header (See Section 18.2) allows for the usage
of other algorithms for hashing the DER records of accepted entities.
The registration of any future algorithm is expected to be extremely
rare and could also cause interoperability problems. Therefore the
bar for registering new algorithms is intentionally placed high.
Any registration of a new hash algorithm MUST fulfill the following
requirement:
o Follow the IETF Standards Action policy.
o A definition of the algorithm and its identifier meeting the
"token" ABNF requirement.
The registered value is:
Hash Alg. Id Reference
------------------------
sha-256 [RFCXXXX]
22.6. Cache-Control Cache Directive Extensions
There exists a number of cache directives which can be sent in the
Cache-Control header. A registry for these cache directives MUST be
defined with the following rules:
o Registering requires an IETF Standards Action or IESG Approval.
o A registration is required to contain a contact person.
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o Name of the directive and a definition of the value, if any.
o Specification if it is a request or response directive.
o A describing text that explains how the cache directive is used
for RTSP controlled media streams.
This specification registers the following values:
no-cache:
public:
private:
no-transform:
only-if-cached:
max-stale:
min-fresh:
must-revalidate:
proxy-revalidate:
max-age:
The registry should be represented as: Name of the directive, contact
person and reference.
22.7. Media Properties
22.7.1. Description
The media streams being controlled by RTSP can have many different
properties. The media properties required to cover the use cases
that were in mind when writing the specification are defined.
However, it can be expected that further innovation will result in
new use cases or media streams with properties not covered by the
ones specified here. Thus new media properties can be specified. As
new media properties may need a substantial amount of new definitions
to correctly specify behavior for this property the bar is intended
to be high.
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22.7.2. Registration Rules
Registering a new media property MUST fulfill the following
requirements
o Follow the Specification Required policy and get the approval of
the designated Expert.
o Have an ABNF definition of the media property value name that
meets "media-prop-ext" definition
o A Contact Person for the Registration
o Description of all changes to the behavior of the RTSP protocol as
result of these changes.
22.7.3. Registered Values
This specification registers the 9 values listed in Section 18.28.
The registry should be represented as: Name of the media property,
contact person and reference.
22.8. Notify-Reason header
22.8.1. Description
Notify-Reason values are used for indicating the reason the
notification was sent. Each reason has its associated rules on what
headers and information that may or must be included in the
notification. New notification behaviors need to be specified to
enable interoperable usage, thus a specification of each new value is
required.
22.8.2. Registration Rules
Registrations for new Notify-Reason value MUST fulfill the following
requirements
o Follow the Specification Required policy and get the approval of
the designated Expert.
o An ABNF definition of the Notify reason value name that meets
"Notify-Reason-extension" definition
o A Contact Person for the Registration
o Description of which headers shall be included in the request and
response, when it should be sent, and any effect it has on the
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server client state.
22.8.3. Registered Values
This specification registers 3 values defined in the Notify-Reas-val
ABNF, Section 20.2.3:
end-of-stream: This Notify-Reason value indicates the end of a media
stream.
media-properties-update: This Notify-Reason value allows the server
to indicate that the properties of the media has changed during
the playout.
scale-change: This Notify-Reason value allows the server to notify
the client about a change in the Scale of the media.
The registry entries should be represented in the registry as: Name,
short description, contact and reference.
22.9. Range header formats
22.9.1. Description
The Range header (Section 18.38) allows for different range formats.
New ones may be registered, but moderation should be applied as it
makes interoperability more difficult.
22.9.2. Registration Rules
A registration MUST fulfill the following requirements:
o Follow the Specification Required policy.
o An ABNF definition of the range format that fulfills the "range-
ext" definition.
o A Contact person for the registration.
o Rules for how one handles the range when using a negative Scale.
22.9.3. Registered Values
The registry should be represented as: Name of the range format,
contact person and reference. This specification registers the
following values.
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npt: Normal Play Time
clock: UTC Clock format
smpte: SMPTE Timestamps
smpte-30-drop: SMPTE Timestamps
smpte-25: SMPTE Timestamps
22.10. Terminate-Reason Header
The Terminate-Reason header (Section 18.50) has two registries for
extensions.
22.10.1. Redirect Reasons
Registrations are done under the policy of Expert Review. The
registered value needs to follow the Terminate-Reason ABNF, i.e., be
a token. The specification needs to provide a definition of what
procedures are to be followed when a client receives this redirect
reason. This specification registers three values:
o Session-Timeout
o Server-Admin
o Internal-Error
The registry should be represented as: Name of the Redirect Reason,
contact person and reference.
22.10.2. Terminate-Reason Header Parameters
Registrations are done under the policy of Specification Required.
The registrations must define a syntax for the parameter that also
follows the syntax allowed by the RTSP 2.0 specification. A contact
person is also required. This specification registers:
o time
o user-msg
The registry should be represented as: Name of the Terminate Reason,
contact person and reference.
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22.11. RTP-Info header parameters
22.11.1. Description
The RTP-Info header (Section 18.43) carries one or more parameter
value pairs with information about a particular point in the RTP
stream. RTP extensions or new usages may need new types of
information. As RTP information that could be needed is likely to be
generic enough and to maximize the interoperability, new registration
requires Specification Required.
22.11.2. Registration Rules
Registrations for new RTP-Info value MUST fulfill the following
requirements
o Follow the Specification Required policy and get the approval of
the designated Expert.
o Have an ABNF definition that meets the "generic-param" definition
o A Contact Person for the Registration
22.11.3. Registered Values
This specification registers the following parameter value pairs:
o url
o ssrc
o seq
o rtptime
The registry should be represented as: Name of the parameter, contact
person and reference.
22.12. Seek-Style Policies
22.12.1. Description
New seek policies may be registered, however, a large number of these
will complicate implementation substantially. The impact of unknown
policies is that the server will not honor the unknown and use the
server default policy instead.
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22.12.2. Registration Rules
Registrations of new Seek-Style polices MUST fulfill the following
requirements
o Follow the Specification Required policy.
o Have an ABNF definition of the Seek-Style policy name that meets
"Seek-S-value-ext" definition
o A Contact Person for the Registration
o Description of which headers shall be included in the request and
response, when it should be sent, and any affect it has on the
server client state.
22.12.3. Registered Values
This specification registers 4 values:
o RAP
o CoRAP
o First-Prior
o Next
The registry should be represented as: Name of the Seek-Style Policy,
short description, contact person and reference.
22.13. Transport Header Registries
The transport header contains a number of parameters which have
possibilities for future extensions. Therefore registries for these
need to be defined.
22.13.1. Transport Protocol Specification
A registry for the parameter transport-protocol specification MUST be
defined with the following rules:
o Registering uses the policy of Specification Required.
o A contact person or organization with address and email.
o A value definition that are following the ABNF syntax definition
of "transport-id" Section 20.2.3.
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o A describing text that explains how the registered value are used
in RTSP.
The registry should be represented as: The protocol ID string,
contact person and reference.
This specification registers the following values:
RTP/AVP: Use of the RTP [RFC3550] protocol for media transport in
combination with the "RTP profile for audio and video
conferences with minimal control" [RFC3551] over UDP. The
usage is explained in RFC XXXX, Appendix C.1.
RTP/AVP/UDP: the same as RTP/AVP.
RTP/AVPF: Use of the RTP [RFC3550] protocol for media transport in
combination with the "Extended RTP Profile for RTCP-based
Feedback (RTP/AVPF)" [RFC4585] over UDP. The usage is
explained in RFC XXXX, Appendix C.1.
RTP/AVPF/UDP: the same as RTP/AVPF.
RTP/SAVP: Use of the RTP [RFC3550] protocol for media transport in
combination with the "The Secure Real-time Transport Protocol
(SRTP)" [RFC3711] over UDP. The usage is explained in RFC
XXXX, Appendix C.1.
RTP/SAVP/UDP: the same as RTP/SAVP.
RTP/SAVPF: Use of the RTP[RFC3550] protocol for media transport in
combination with the Extended Secure RTP Profile for Real-time
Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF)
[RFC5124] over UDP. The usage is explained in RFC XXXX,
Appendix C.1.
RTP/SAVPF/UDP: the same as RTP/SAVPF.
RTP/AVP/TCP: Use of the RTP [RFC3550] protocol for media transport
in combination with the "RTP profile for audio and video
conferences with minimal control" [RFC3551] over TCP. The
usage is explained in RFC XXXX, Appendix C.2.2.
RTP/AVPF/TCP: Use of the RTP [RFC3550] protocol for media transport
in combination with the "Extended RTP Profile for RTCP-based
Feedback (RTP/AVPF)" [RFC4585] over TCP. The usage is
explained in RFC XXXX, Appendix C.2.2.
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RTP/SAVP/TCP: Use of the RTP [RFC3550] protocol for media transport
in combination with the "The Secure Real-time Transport
Protocol (SRTP)" [RFC3711] over TCP. The usage is explained in
RFC XXXX, Appendix C.2.2.
RTP/SAVPF/TCP: Use of the RTP [RFC3550] protocol for media transport
in combination with the "Extended Secure RTP Profile for Real-
time Transport Control Protocol (RTCP)-Based Feedback (RTP/
SAVPF)" [RFC5124] over TCP. The usage is explained in RFC
XXXX, Appendix C.2.2.
22.13.2. Transport modes
A registry for the transport parameter mode MUST be defined with the
following rules:
o Registering requires an IETF Standards Action.
o A contact person or organization with address and email.
o A value definition that are following the ABNF "token" definition
Section 20.2.3.
o A describing text that explains how the registered value are used
in RTSP.
This specification registers 1 value:
PLAY: See RFC XXXX.
22.13.3. Transport Parameters
A registry for parameters that may be included in the Transport
header MUST be defined with the following rules:
o Registering uses the Specification Required policy.
o A value definition that are following the ABNF "token" definition
Section 20.2.3.
o A describing text that explains how the registered value are used
in RTSP.
This specification registers all the transport parameters defined in
Section 18.52. This is a copy of this list:
o unicast
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o multicast
o interleaved
o ttl
o layers
o ssrc
o mode
o dest_addr
o src_addr
o setup
o connection
o RTCP-mux
o MIKEY
22.14. URI Schemes
This specification defines two URI schemes ("rtsp" and "rtsps") and
reserves a third one ("rtspu"). These URI schemes are defined in
existing registries which are not created by RTSP. Registrations are
following RFC 4395[RFC4395].
22.14.1. The rtsp URI Scheme
URI scheme name: rtsp
Status: Permanent
URI scheme syntax: See Section 20.2.1 of RFC XXXX.
URI scheme semantics: The rtsp scheme is used to indicate resources
accessible through the usage of the Real-time Streaming
Protocol (RTSP). RTSP allows different operations on the
resource identified by the URI, but the primary purpose is the
streaming delivery of the resource to a client. However, the
operations that are currently defined are: DESCRIBE,
GET_PARAMETER, OPTIONS, PLAY, PLAY_NOTIFY, PAUSE, REDIRECT,
SETUP, SET_PARAMETER, and TEARDOWN.
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Encoding considerations: IRIs in this scheme are defined and needs
to be encoded as RTSP URIs when used within the RTSP protocol.
That encoding is done according to RFC 3987.
Applications/protocols that use this URI scheme name: RTSP 1.0 (RFC
2326), RTSP 2.0 (RFC XXXX)
Interoperability considerations: The extensions in the URI syntax
performed between RTSP 1.0 and 2.0 can create interoperability
issues. The changes are:
Support for IPV6 literal in host part and future IP
literals through RFC 3986 defined mechanism.
A new relative format to use in the RTSP protocol
elements that is not required to start with "/".
Security considerations: All the security threats identified in
Section 7 of RFC 3986 apply also to this scheme. They need to
be reviewed and considered in any implementation utilizing this
scheme.
Contact: Magnus Westerlund, magnus.westerlund@ericsson.com
Author/Change controller: IETF
References: RFC 2326, RFC 3986, RFC 3987, RFC XXXX
22.14.2. The rtsps URI Scheme
URI scheme name: rtsps
Status: Permanent
URI scheme syntax: See Section 20.2.1 of RFC XXXX.
URI scheme semantics: The rtsps scheme is used to indicate resources
accessible through the usage of the Real-time Streaming
Protocol (RTSP) over TLS. RTSP allows different operations on
the resource identified by the URI, but the primary purpose is
the streaming delivery of the resource to a client. However,
the operations that are currently defined are: DESCRIBE,
GET_PARAMETER, OPTIONS, PLAY, PLAY_NOTIFY, PAUSE, REDIRECT,
SETUP, SET_PARAMETER, and TEARDOWN.
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Encoding considerations: IRIs in this scheme are defined and needs
to be encoded as RTSP URIs when used within the RTSP protocol.
That encoding is done according to RFC 3987.
Applications/protocols that use this URI scheme name: RTSP 1.0 (RFC
2326), RTSP 2.0 (RFC XXXX)
Interoperability considerations: The extensions in the URI syntax
performed between RTSP 1.0 and 2.0 can create interoperability
issues. The changes are:
Support for IPV6 literal in host part and future IP
literals through RFC 3986 defined mechanism.
A new relative format to use in the RTSP protocol
elements that is not required to start with "/".
Security considerations: All the security threats identified in
Section 7 of RFC 3986 apply also to this scheme. They need to
be reviewed and considered in any implementation utilizing this
scheme.
Contact: Magnus Westerlund, magnus.westerlund@ericsson.com
Author/Change controller: IETF
References: RFC 2326, RFC 3986, RFC 3987, RFC XXXX
22.14.3. The rtspu URI Scheme
URI scheme name: rtspu
Status: Permanent
URI scheme syntax: See Section 3.2 of RFC 2326.
URI scheme semantics: The rtspu scheme is used to indicate resources
accessible through the usage of the Real-time Streaming
Protocol (RTSP) over unreliable datagram transport. RTSP
allows different operations on the resource identified by the
URI, but the primary purpose is the streaming delivery of the
resource to a client. However, the operations that are
currently defined are: DESCRIBE, GET_PARAMETER, OPTIONS,
REDIRECT,PLAY, PLAY_NOTIFY, PAUSE, SETUP, SET_PARAMETER, and
TEARDOWN.
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Encoding considerations: IRIs in this scheme are not defined.
Applications/protocols that use this URI scheme name: RTSP 1.0 (RFC
2326)
Interoperability considerations: The definition of the transport
mechanism of RTSP over UDP has interoperability issues. That
makes the usage of this scheme problematic.
Security considerations: All the security threats identified in
Section 7 of RFC 3986 apply also to this scheme. They needs to
be reviewed and considered in any implementation utilizing this
scheme.
Contact: Magnus Westerlund, magnus.westerlund@ericsson.com
Author/Change controller: IETF
References: RFC 2326
22.15. SDP attributes
This specification defines three SDP [RFC4566] attributes that it is
requested that IANA register.
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SDP Attribute ("att-field"):
Attribute name: range
Long form: Media Range Attribute
Type of name: att-field
Type of attribute: Media and session level
Subject to charset: No
Purpose: RFC XXXX
Reference: RFC XXXX, RFC 2326
Values: See ABNF definition.
Attribute name: control
Long form: RTSP control URI
Type of name: att-field
Type of attribute: Media and session level
Subject to charset: No
Purpose: RFC XXXX
Reference: RFC XXXX, RFC 2326
Values: Absolute or Relative URIs.
Attribute name: mtag
Long form: Message Tag
Type of name: att-field
Type of attribute: Media and session level
Subject to charset: No
Purpose: RFC XXXX
Reference: RFC XXXX
Values: See ABNF definition
22.16. Media Type Registration for text/parameters
Type name: text
Subtype name: parameters
Required parameters:
Optional parameters: charset: The charset parameter is applicable to
the encoding of the parameter values. The default charset is
UTF-8, if the 'charset' parameter is not present.
Encoding considerations: 8bit
Security considerations: This format may carry any type of
parameters. Some can have security requirements, like privacy,
confidentiality or integrity requirements. The format has no
built in security protection. For the usage it was defined the
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transport can be protected between server and client using TLS.
However, care must be taken to consider if also the proxies are
trusted with the parameters in case hop-by-hop security is used.
If stored as file in file systemi, the necessary precautions need
to be taken in relation to the parameters requirements including
object security such as S/MIME [RFC5751].
Interoperability considerations: This media type was mentioned as a
fictional example in [RFC2326], but was not formally specified.
This has resulted in usage of this media type which may not match
its formal definition.
Published specification: RFC XXXX, Appendix F.
Applications that use this media type: Applications that use RTSP
and have additional parameters they like to read and set using the
RTSP GET_PARAMETER and SET_PARAMETER methods.
Additional information:
Magic number(s):
File extension(s):
Macintosh file type code(s):
Person & email address to contact for further information: Magnus
Westerlund (magnus.westerlund@ericsson.com)
Intended usage: Common
Restrictions on usage: None
Author: Magnus Westerlund (magnus.westerlund@ericsson.com)
Change controller: IETF
Addition Notes:
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23. References
23.1. Normative References
[FIPS-pub-180-2]
National Institute of Standards and Technology (NIST),
"Federal Information Processing Standards Publications
(FIPS PUBS) 180-2: Secure Hash Standard", August 2002.
[RFC0768] Postel, J., "User Datagram Protocol", STD 6, RFC 768,
August 1980.
[RFC0793] Postel, J., "Transmission Control Protocol", STD 7,
RFC 793, September 1981.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC2460] Deering, S. and R. Hinden, "Internet Protocol, Version 6
(IPv6) Specification", RFC 2460, December 1998.
[RFC2616] Fielding, R., Gettys, J., Mogul, J., Frystyk, H.,
Masinter, L., Leach, P., and T. Berners-Lee, "Hypertext
Transfer Protocol -- HTTP/1.1", RFC 2616, June 1999.
[RFC2617] Franks, J., Hallam-Baker, P., Hostetler, J., Lawrence, S.,
Leach, P., Luotonen, A., and L. Stewart, "HTTP
Authentication: Basic and Digest Access Authentication",
RFC 2617, June 1999.
[RFC2818] Rescorla, E., "HTTP Over TLS", RFC 2818, May 2000.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
July 2003.
[RFC3629] Yergeau, F., "UTF-8, a transformation format of ISO
10646", STD 63, RFC 3629, November 2003.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
[RFC3830] Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K.
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Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830,
August 2004.
[RFC3986] Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform
Resource Identifier (URI): Generic Syntax", STD 66,
RFC 3986, January 2005.
[RFC3987] Duerst, M. and M. Suignard, "Internationalized Resource
Identifiers (IRIs)", RFC 3987, January 2005.
[RFC4086] Eastlake, D., Schiller, J., and S. Crocker, "Randomness
Requirements for Security", BCP 106, RFC 4086, June 2005.
[RFC4291] Hinden, R. and S. Deering, "IP Version 6 Addressing
Architecture", RFC 4291, February 2006.
[RFC4395] Hansen, T., Hardie, T., and L. Masinter, "Guidelines and
Registration Procedures for New URI Schemes", BCP 35,
RFC 4395, February 2006.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
[RFC4567] Arkko, J., Lindholm, F., Naslund, M., Norrman, K., and E.
Carrara, "Key Management Extensions for Session
Description Protocol (SDP) and Real Time Streaming
Protocol (RTSP)", RFC 4567, July 2006.
[RFC4571] Lazzaro, J., "Framing Real-time Transport Protocol (RTP)
and RTP Control Protocol (RTCP) Packets over Connection-
Oriented Transport", RFC 4571, July 2006.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
July 2006.
[RFC4648] Josefsson, S., "The Base16, Base32, and Base64 Data
Encodings", RFC 4648, October 2006.
[RFC4738] Ignjatic, D., Dondeti, L., Audet, F., and P. Lin, "MIKEY-
RSA-R: An Additional Mode of Key Distribution in
Multimedia Internet KEYing (MIKEY)", RFC 4738,
November 2006.
[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for
Real-time Transport Control Protocol (RTCP)-Based Feedback
(RTP/SAVPF)", RFC 5124, February 2008.
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[RFC5226] Narten, T. and H. Alvestrand, "Guidelines for Writing an
IANA Considerations Section in RFCs", BCP 26, RFC 5226,
May 2008.
[RFC5234] Crocker, D. and P. Overell, "Augmented BNF for Syntax
Specifications: ABNF", STD 68, RFC 5234, January 2008.
[RFC5246] Dierks, T. and E. Rescorla, "The Transport Layer Security
(TLS) Protocol Version 1.2", RFC 5246, August 2008.
[RFC5280] Cooper, D., Santesson, S., Farrell, S., Boeyen, S.,
Housley, R., and W. Polk, "Internet X.509 Public Key
Infrastructure Certificate and Certificate Revocation List
(CRL) Profile", RFC 5280, May 2008.
[RFC5646] Phillips, A. and M. Davis, "Tags for Identifying
Languages", BCP 47, RFC 5646, September 2009.
[RFC5751] Ramsdell, B. and S. Turner, "Secure/Multipurpose Internet
Mail Extensions (S/MIME) Version 3.2 Message
Specification", RFC 5751, January 2010.
[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port", RFC 5761, April 2010.
[RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description
Protocol (SDP) Grouping Framework", RFC 5888, June 2010.
[RFC6838] Freed, N., Klensin, J., and T. Hansen, "Media Type
Specifications and Registration Procedures", BCP 13,
RFC 6838, January 2013.
[TS-26234]
Third Generation Partnership Project (3GPP), "Transparent
end-to-end Packet-switched Streaming Service (PSS);
Protocols and codecs; Technical Specification 26.234",
December 2002.
23.2. Informative References
[I-D.ietf-mmusic-rtsp-nat]
Goldberg, J., Westerlund, M., and T. Zeng, "A Network
Address Translator (NAT) Traversal mechanism for media
controlled by Real-Time Streaming Protocol (RTSP)",
draft-ietf-mmusic-rtsp-nat-14 (work in progress),
November 2012.
[ISO.13818-6.1995]
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International Organization for Standardization,
"Information technology - Generic coding of moving
pictures and associated audio information - part 6:
Extension for digital storage media and control",
ISO Draft Standard 13818-6, November 1995.
[ISO.8601.2000]
International Organization for Standardization, "Data
elements and interchange formats - Information interchange
- Representation of dates and times", ISO/IEC Standard
8601, December 2000.
[RFC1123] Braden, R., "Requirements for Internet Hosts - Application
and Support", STD 3, RFC 1123, October 1989.
[RFC2068] Fielding, R., Gettys, J., Mogul, J., Nielsen, H., and T.
Berners-Lee, "Hypertext Transfer Protocol -- HTTP/1.1",
RFC 2068, January 1997.
[RFC2326] Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time
Streaming Protocol (RTSP)", RFC 2326, April 1998.
[RFC2663] Srisuresh, P. and M. Holdrege, "IP Network Address
Translator (NAT) Terminology and Considerations",
RFC 2663, August 1999.
[RFC2822] Resnick, P., "Internet Message Format", RFC 2822,
April 2001.
[RFC2974] Handley, M., Perkins, C., and E. Whelan, "Session
Announcement Protocol", RFC 2974, October 2000.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
June 2002.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264,
June 2002.
[RFC4145] Yon, D. and G. Camarillo, "TCP-Based Media Transport in
the Session Description Protocol (SDP)", RFC 4145,
September 2005.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
July 2006.
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[RFC4855] Casner, S., "Media Type Registration of RTP Payload
Formats", RFC 4855, February 2007.
[RFC4856] Casner, S., "Media Type Registration of Payload Formats in
the RTP Profile for Audio and Video Conferences",
RFC 4856, February 2007.
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
"Codec Control Messages in the RTP Audio-Visual Profile
with Feedback (AVPF)", RFC 5104, February 2008.
[RFC5583] Schierl, T. and S. Wenger, "Signaling Media Decoding
Dependency in the Session Description Protocol (SDP)",
RFC 5583, July 2009.
[RFC5905] Mills, D., Martin, J., Burbank, J., and W. Kasch, "Network
Time Protocol Version 4: Protocol and Algorithms
Specification", RFC 5905, June 2010.
[RFC6298] Paxson, V., Allman, M., Chu, J., and M. Sargent,
"Computing TCP's Retransmission Timer", RFC 6298,
June 2011.
[Stevens98]
Stevens, W., "Unix Networking Programming - Volume 1,
second edition", 1998.
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Appendix A. Examples
This section contains several different examples trying to illustrate
possible ways of using RTSP. The examples can also help with the
understanding of how functions of RTSP work. However, remember that
these are examples and the normative and syntax description in the
other sections take precedence. Please also note that many of the
examples contain syntax illegal line breaks to accommodate the
formatting restriction that the RFC series impose.
A.1. Media on Demand (Unicast)
This is an example of media on demand streaming of a media stored in
a container file. For purposes of this example, a container file is
a storage entity in which multiple continuous media types pertaining
to the same end-user presentation are present. In effect, the
container file represents an RTSP presentation, with each of its
components being RTSP controlled media streams. Container files are
a widely used means to store such presentations. While the
components are transported as independent streams, it is desirable to
maintain a common context for those streams at the server end.
This enables the server to keep a single storage handle open
easily. It also allows treating all the streams equally in case
of any priorization of streams by the server.
It is also possible that the presentation author may wish to prevent
selective retrieval of the streams by the client in order to preserve
the artistic effect of the combined media presentation. Similarly,
in such a tightly bound presentation, it is desirable to be able to
control all the streams via a single control message using an
aggregate URI.
The following is an example of using a single RTSP session to control
multiple streams. It also illustrates the use of aggregate URIs. In
a container file it is also desirable to not write any URI parts
which are not kept, when the container is distributed, like the host
and most of the path element. Therefore this example also uses the
"*" and relative URI in the delivered SDP.
Also this presentation description (SDP) is not cachable, as the
Expires header is set to an equal value with date indicating
immediate expiration of its valididty.
Client C requests a presentation from media server M. The movie is
stored in a container file. The client has obtained an RTSP URI to
the container file.
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C->M: DESCRIBE rtsp://example.com/twister.3gp RTSP/2.0
CSeq: 1
User-Agent: PhonyClient/1.2
M->C: RTSP/2.0 200 OK
CSeq: 1
Server: PhonyServer/1.0
Date: Thu, 24 Jan 1997 15:35:06 GMT
Content-Type: application/sdp
Content-Length: 271
Content-Base: rtsp://example.com/twister.3gp/
Expires: 24 Jan 1997 15:35:06 GMT
v=0
o=- 2890844256 2890842807 IN IP4 198.51.100.5
s=RTSP Session
i=An Example of RTSP Session Usage
e=adm@example.com
c=IN IP4 0.0.0.0
a=control: *
a=range:npt=0-0:10:34.10
t=0 0
m=audio 0 RTP/AVP 0
a=control: trackID=1
m=video 0 RTP/AVP 26
a=control: trackID=4
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C->M: SETUP rtsp://example.com/twister.3gp/trackID=1 RTSP/2.0
CSeq: 2
User-Agent: PhonyClient/1.2
Require: play.basic
Transport: RTP/AVP;unicast;dest_addr=":8000"/":8001"
Accept-Ranges: NPT, SMPTE, UTC
M->C: RTSP/2.0 200 OK
CSeq: 2
Server: PhonyServer/1.0
Transport: RTP/AVP;unicast; ssrc=93CB001E;
dest_addr="192.0.2.53:8000"/"192.0.2.53:8001";
src_addr="198.51.100.5:9000"/"198.51.100.5:9001"
Session: 12345678
Expires: 24 Jan 1997 15:35:12 GMT
Date: 24 Jan 1997 15:35:12 GMT
Accept-Ranges: NPT
Media-Properties: Random-Access=0.02, Immutable, Unlimited
C->M: SETUP rtsp://example.com/twister.3gp/trackID=4 RTSP/2.0
CSeq: 3
User-Agent: PhonyClient/1.2
Require: play.basic
Transport: RTP/AVP;unicast;dest_addr=":8002"/":8003"
Session: 12345678
Accept-Ranges: NPT, SMPTE, UTC
M->C: RTSP/2.0 200 OK
CSeq: 3
Server: PhonyServer/1.0
Transport: RTP/AVP;unicast; ssrc=A813FC13;
dest_addr="192.0.2.53:8002"/"192.0.2.53:8003";
src_addr="198.51.100.5:9002"/"198.51.100.5:9003";
Session: 12345678
Expires: 24 Jan 1997 15:35:13 GMT
Date: 24 Jan 1997 15:35:13 GMT
Accept-Range: NPT
Media-Properties: Random-Access=0.8, Immutable, Unlimited
C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0
CSeq: 4
User-Agent: PhonyClient/1.2
Range: npt=30-
Seek-Style: RAP
Session: 12345678
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M->C: RTSP/2.0 200 OK
CSeq: 4
Server: PhonyServer/1.0
Date: 24 Jan 1997 15:35:14 GMT
Session: 12345678
Range: npt=30-634.10
Seek-Style: RAP
RTP-Info: url="rtsp://example.com/twister.3gp/trackID=4"
ssrc=0D12F123:seq=12345;rtptime=3450012,
url="rtsp://example.com/twister.3gp/trackID=1"
ssrc=4F312DD8:seq=54321;rtptime=2876889
C->M: PAUSE rtsp://example.com/twister.3gp/ RTSP/2.0
CSeq: 5
User-Agent: PhonyClient/1.2
Session: 12345678
# Pause happens 0.87 seconds after starting to play
M->C: RTSP/2.0 200 OK
CSeq: 5
Server: PhonyServer/1.0
Date: 24 Jan 1997 15:36:01 GMT
Session: 12345678
Range: npt=30.87-634.10
C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0
CSeq: 6
User-Agent: PhonyClient/1.2
Range: npt=30.87-634.10
Seek-Style: Next
Session: 12345678
M->C: RTSP/2.0 200 OK
CSeq: 6
Server: PhonyServer/1.0
Date: 24 Jan 1997 15:36:01 GMT
Session: 12345678
Range: npt=30.87-634.10
Seek-Style: Next
RTP-Info: url="rtsp://example.com/twister.3gp/trackID=4"
ssrc=0D12F123:seq=12555;rtptime=6330012,
url="rtsp://example.com/twister.3gp/trackID=1"
ssrc=4F312DD8:seq=55021;rtptime=3132889
C->M: TEARDOWN rtsp://example.com/twister.3gp/ RTSP/2.0
CSeq: 7
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User-Agent: PhonyClient/1.2
Session: 12345678
M->C: RTSP/2.0 200 OK
CSeq: 7
Server: PhonyServer/1.0
Date: 24 Jan 1997 15:49:34 GMT
A.2. Media on Demand using Pipelining
This example is basically the example above (Appendix A.1), but now
utilizing pipelining to speed up the setup. It requires only two
round trip times until the media starts flowing. First of all, the
session description is retrieved to determine what media resources
need to be setup. In the second step, one sends the necessary SETUP
requests and the PLAY request to initiate media delivery.
Client C requests a presentation from media server M. The movie is
stored in a container file. The client has obtained an RTSP URI to
the container file.
C->M: DESCRIBE rtsp://example.com/twister.3gp RTSP/2.0
CSeq: 1
User-Agent: PhonyClient/1.2
M->C: RTSP/2.0 200 OK
CSeq: 1
Server: PhonyServer/1.0
Date: Thu, 23 Jan 1997 15:35:06 GMT
Content-Type: application/sdp
Content-Length: 271
Content-Base: rtsp://example.com/twister.3gp/
Expires: 24 Jan 1997 15:35:06 GMT
v=0
o=- 2890844256 2890842807 IN IP4 192.0.2.5
s=RTSP Session
i=An Example of RTSP Session Usage
e=adm@example.com
c=IN IP4 0.0.0.0
a=control: *
a=range:npt=0-0:10:34.10
t=0 0
m=audio 0 RTP/AVP 0
a=control: trackID=1
m=video 0 RTP/AVP 26
a=control: trackID=4
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C->M: SETUP rtsp://example.com/twister.3gp/trackID=1 RTSP/2.0
CSeq: 2
User-Agent: PhonyClient/1.2
Require: play.basic
Transport: RTP/AVP;unicast;dest_addr=":8000"/":8001"
Accept-Ranges: NPT, SMPTE, UTC
Pipelined-Requests: 7654
C->M: SETUP rtsp://example.com/twister.3gp/trackID=4 RTSP/2.0
CSeq: 3
User-Agent: PhonyClient/1.2
Require: play.basic
Transport: RTP/AVP;unicast;dest_addr=":8002"/":8003"
Accept-Ranges: NPT, SMPTE, UTC
Pipelined-Requests: 7654
C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0
CSeq: 4
User-Agent: PhonyClient/1.2
Range: npt=0-
Seek-Style: RAP
Pipelined-Requests: 7654
M->C: RTSP/2.0 200 OK
CSeq: 2
Server: PhonyServer/1.0
Transport: RTP/AVP;unicast;
dest_addr="192.0.2.53:8000"/"192.0.2.53:8001";
src_addr="198.51.100.5:9000"/"198.51.100.5:9001";
ssrc=93CB001E
Session: 12345678
Expires: 24 Jan 1997 15:35:12 GMT
Date: 23 Jan 1997 15:35:12 GMT
Accept-Ranges: NPT
Pipelined-Requests: 7654
Media-Properties: Random-Access=0.2, Immutable, Unlimited
M->C: RTSP/2.0 200 OK
CSeq: 3
Server: PhonyServer/1.0
Transport: RTP/AVP;unicast;
dest_addr="192.0.2.53:8002"/"192.0.2.53:8003;
src_addr="198.51.100.5:9002"/"198.51.100.5:9003";
ssrc=A813FC13
Session: 12345678
Expires: 24 Jan 1997 15:35:13 GMT
Date: 23 Jan 1997 15:35:13 GMT
Accept-Range: NPT
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Pipelined-Requests: 7654
Media-Properties: Random-Access=0.8, Immutable, Unlimited
M->C: RTSP/2.0 200 OK
CSeq: 4
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:35:14 GMT
Session: 12345678
Range: npt=0-623.10
Seek-Style: RAP
RTP-Info: url="rtsp://example.com/twister.3gp/trackID=4"
ssrc=0D12F123:seq=12345;rtptime=3450012,
url="rtsp://example.com/twister.3gp/trackID=1"
ssrc=4F312DD8:seq=54321;rtptime=2876889
Pipelined-Requests: 7654
A.3. Media on Demand (Unicast)
An alternative example of media on demand with a bit more tweaks is
the following. Client C requests a movie distributed from two
different media servers A (audio.example.com) and V (
video.example.com). The media description is stored on a web server
W. The media description contains descriptions of the presentation
and all its streams, including the codecs that are available, dynamic
RTP payload types, the protocol stack, and content information such
as language or copyright restrictions. It may also give an
indication about the timeline of the movie.
In this example, the client is only interested in the last part of
the movie.
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C->W: GET /twister.sdp HTTP/1.1
Host: www.example.com
Accept: application/sdp
W->C: HTTP/1.1 200 OK
Date: Thu, 23 Jan 1997 15:35:06 GMT
Content-Type: application/sdp
Content-Length: 278
Expires: 23 Jan 1998 15:35:06 GMT
v=0
o=- 2890844526 2890842807 IN IP4 198.51.100.5
s=RTSP Session
e=adm@example.com
c=IN IP4 0.0.0.0
a=range:npt=0-1:49:34
t=0 0
m=audio 0 RTP/AVP 0
a=control:rtsp://audio.example.com/twister/audio.en
m=video 0 RTP/AVP 31
a=control:rtsp://video.example.com/twister/video
C->A: SETUP rtsp://audio.example.com/twister/audio.en RTSP/2.0
CSeq: 1
User-Agent: PhonyClient/1.2
Transport: RTP/AVP/UDP;unicast;dest_addr=":3056"/":3057",
RTP/AVP/TCP;unicast;interleaved=0-1
Accept-Ranges: NPT, SMPTE, UTC
A->C: RTSP/2.0 200 OK
CSeq: 1
Session: 12345678
Transport: RTP/AVP/UDP;unicast;
dest_addr="192.0.2.53:3056"/"192.0.2.53:3057";
src_addr="198.51.100.5:5000"/"198.51.100.5:5001"
Date: 23 Jan 1997 15:35:12 GMT
Server: PhonyServer/1.0
Expires: 24 Jan 1997 15:35:12 GMT
Cache-Control: public
Accept-Ranges: NPT, SMPTE
Media-Properties: Random-Access=0.02, Immutable, Unlimited
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C->V: SETUP rtsp://video.example.com/twister/video RTSP/2.0
CSeq: 1
User-Agent: PhonyClient/1.2
Transport: RTP/AVP/UDP;unicast;
dest_addr="192.0.2.53:3058"/"192.0.2.53:3059",
RTP/AVP/TCP;unicast;interleaved=0-1
Accept-Ranges: NPT, SMPTE, UTC
V->C: RTSP/2.0 200 OK
CSeq: 1
Session: 23456789
Transport: RTP/AVP/UDP;unicast;
dest_addr="192.0.2.53:3058"/"192.0.2.53:3059";
src_addr="198.51.100.5:5002"/"198.51.100.5:5003"
Date: 23 Jan 1997 15:35:12 GMT
Server: PhonyServer/1.0
Cache-Control: public
Expires: 24 Jan 1997 15:35:12 GMT
Accept-Ranges: NPT, SMPTE
Media-Properties: Random-Access=1.2, Immutable, Unlimited
C->V: PLAY rtsp://video.example.com/twister/video RTSP/2.0
CSeq: 2
User-Agent: PhonyClient/1.2
Session: 23456789
Range: smpte=0:10:00-
V->C: RTSP/2.0 200 OK
CSeq: 2
Session: 23456789
Range: smpte=0:10:00-1:49:23
Seek-Style: First-Prior
RTP-Info: url="rtsp://video.example.com/twister/video"
ssrc=A17E189D:seq=12312232;rtptime=78712811
Server: PhonyServer/2.0
Date: 23 Jan 1997 15:35:13 GMT
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C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/2.0
CSeq: 2
User-Agent: PhonyClient/1.2
Session: 12345678
Range: smpte=0:10:00-
A->C: RTSP/2.0 200 OK
CSeq: 2
Session: 12345678
Range: smpte=0:10:00-1:49:23
Seek-Style: First-Prior
RTP-Info: url="rtsp://audio.example.com/twister/audio.en"
ssrc=3D124F01:seq=876655;rtptime=1032181
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:35:13 GMT
C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en RTSP/2.0
CSeq: 3
User-Agent: PhonyClient/1.2
Session: 12345678
A->C: RTSP/2.0 200 OK
CSeq: 3
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:36:52 GMT
C->V: TEARDOWN rtsp://video.example.com/twister/video RTSP/2.0
CSeq: 3
User-Agent: PhonyClient/1.2
Session: 23456789
V->C: RTSP/2.0 200 OK
CSeq: 3
Server: PhonyServer/2.0
Date: 23 Jan 1997 15:36:52 GMT
Even though the audio and video track are on two different servers
that may start at slightly different times and may drift with respect
to each other over time, the client can perform initial
synchronization of the two media using RTP-Info and Range received in
the PLAY responses. If the two servers are time synchronized the
RTCP packets can also be used to maintain synchronization.
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A.4. Single Stream Container Files
Some RTSP servers may treat all files as though they are "container
files", yet other servers may not support such a concept. Because of
this, clients needs to use the rules set forth in the session
description for Request-URIs, rather than assuming that a consistent
URI may always be used throughout. Below is an example of how a
multi-stream server might expect a single-stream file to be served:
C->S: DESCRIBE rtsp://foo.example.com/test.wav RTSP/2.0
Accept: application/x-rtsp-mh, application/sdp
CSeq: 1
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 1
Content-base: rtsp://foo.example.com/test.wav/
Content-type: application/sdp
Content-length: 163
Server: PhonyServer/1.0
Date: Thu, 23 Jan 1997 15:35:06 GMT
Expires: 23 Jan 1997 17:00:00 GMT
v=0
o=- 872653257 872653257 IN IP4 192.0.2.5
s=mu-law wave file
i=audio test
c=IN IP4 0.0.0.0
t=0 0
a=control: *
m=audio 0 RTP/AVP 0
a=control:streamid=0
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C->S: SETUP rtsp://foo.example.com/test.wav/streamid=0 RTSP/2.0
Transport: RTP/AVP/UDP;unicast;
dest_addr=":6970"/":6971";mode="PLAY"
CSeq: 2
User-Agent: PhonyClient/1.2
Accept-Ranges: NPT, SMPTE, UTC
S->C: RTSP/2.0 200 OK
Transport: RTP/AVP/UDP;unicast;
dest_addr="192.0.2.53:6970"/"192.0.2.53:6971";
src_addr="198.51.100.5:6970"/"198.51.100.5:6971";
mode="PLAY";ssrc=EAB98712
CSeq: 2
Session: 2034820394
Expires: 23 Jan 1997 16:00:00 GMT
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:35:07 GMT
Accept-Ranges: NPT
Media-Properties: Random-Acces=0.5, Immutable, Unlimited
C->S: PLAY rtsp://foo.example.com/test.wav/ RTSP/2.0
CSeq: 3
User-Agent: PhonyClient/1.2
Session: 2034820394
S->C: RTSP/2.0 200 OK
CSeq: 3
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:35:08 GMT
Session: 2034820394
Range: npt=0-600
Seek-Style: RAP
RTP-Info: url="rtsp://foo.example.com/test.wav/streamid=0"
ssrc=0D12F123:seq=981888;rtptime=3781123
Note the different URI in the SETUP command, and then the switch back
to the aggregate URI in the PLAY command. This makes complete sense
when there are multiple streams with aggregate control, but is less
than intuitive in the special case where the number of streams is
one. However, the server has declared the aggregated control URI in
the SDP and therefore this is legal.
In this case, it is also required that servers accept implementations
that use the non-aggregated interpretation and use the individual
media URI, like this:
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C->S: PLAY rtsp://example.com/test.wav/streamid=0 RTSP/2.0
CSeq: 3
User-Agent: PhonyClient/1.2
Session: 2034820394
A.5. Live Media Presentation Using Multicast
The media server M chooses the multicast address and port. Here, it
is assumed that the web server only contains a pointer to the full
description, while the media server M maintains the full description.
C->W: GET /sessions.html HTTP/1.1
Host: www.example.com
W->C: HTTP/1.1 200 OK
Content-Type: text/html
<html>
...
<a href "rtsp://live.example.com/concert/audio">
Streamed Live Music performance </a>
...
</html>
C->M: DESCRIBE rtsp://live.example.com/concert/audio RTSP/2.0
CSeq: 1
Supported: play.basic, play.scale
User-Agent: PhonyClient/1.2
M->C: RTSP/2.0 200 OK
CSeq: 1
Content-Type: application/sdp
Content-Length: 183
Server: PhonyServer/1.0
Date: Thu, 23 Jan 1997 15:35:06 GMT
Supported: play.basic
v=0
o=- 2890844526 2890842807 IN IP4 192.0.2.5
s=RTSP Session
t=0 0
m=audio 3456 RTP/AVP 0
c=IN IP4 233.252.0.54/16
a=control: rtsp://live.example.com/concert/audio
a=range:npt=0-
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C->M: SETUP rtsp://live.example.com/concert/audio RTSP/2.0
CSeq: 2
Transport: RTP/AVP;multicast;
dest_addr="233.252.0.54:3456"/"233.252.0.54:3457";ttl=16
Accept-Ranges: NPT, SMPTE, UTC
User-Agent: PhonyClient/1.2
M->C: RTSP/2.0 200 OK
CSeq: 2
Server: PhonyServer/1.0
Date: Thu, 23 Jan 1997 15:35:06 GMT
Transport: RTP/AVP;multicast;
dest_addr="233.252.0.54:3456"/"233.252.0.54:3457";ttl=16
;ssrc=4D12AB92/0DF876A3
Session: 0456804596
Accept-Ranges: NPT, UTC
Media-Properties: No-Seeking, Time-Progressing, Time-Duration=0
C->M: PLAY rtsp://live.example.com/concert/audio RTSP/2.0
CSeq: 3
Session: 0456804596
User-Agent: PhonyClient/1.2
M->C: RTSP/2.0 200 OK
CSeq: 3
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:35:07 GMT
Session: 0456804596
Seek-Style: Next
Range:npt=1256-
RTP-Info: url="rtsp://live.example.com/concert/audio"
ssrc=0D12F123:seq=1473; rtptime=80000
A.6. Capability Negotiation
This example illustrates how the client and server determine their
capability to support a special feature, in this case "play.scale".
The server, through the clients request and the included Supported
header, learns the client supports RTSP 2.0, and also supports the
playback time scaling feature of RTSP. The server's response
contains the following feature related information to the client; it
supports the basic media delivery functions (play.basic), the
extended functionality of time scaling of content (play.scale), and
one "example.com" proprietary feature (com.example.flight). The
client also learns the methods supported (Public header) by the
server for the indicated resource.
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C->S: OPTIONS rtsp://media.example.com/movie/twister.3gp RTSP/2.0
CSeq: 1
Supported: play.basic, play.scale
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 1
Public:OPTIONS,SETUP,PLAY,PAUSE,TEARDOWN,DESCRIBE,GET_PARAMETER
Allow: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN, DESCRIBE
Server: PhonyServer/2.0
Supported: play.basic, play.scale, com.example.flight
When the client sends its SETUP request it tells the server that it
requires support of the play.scale feature for this session by
including the Require header.
C->S: SETUP rtsp://media.example.com/twister.3gp/trackID=1 RTSP/2.0
CSeq: 3
User-Agent: PhonyClient/1.2
Transport: RTP/AVP/UDP;unicast;
dest_addr="192.0.2.53:3056"/"192.0.2.53:3057",
RTP/AVP/TCP;unicast;interleaved=0-1
Require: play.scale
Accept-Ranges: NPT, SMPTE, UTC
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 3
Session: 12345678
Transport: RTP/AVP/UDP;unicast;
dest_addr="192.0.2.53:3056"/"192.0.2.53:3057";
src_addr="198.51.100.5:5000"/"198.51.100.5:5001"
Server: PhonyServer/2.0
Accept-Ranges: NPT, SMPTE
Media-Properties: Random-Access=0.8, Immutable, Unlimited
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Appendix B. RTSP Protocol State Machine
The RTSP session state machine describes the behavior of the protocol
from RTSP session initialization through RTSP session termination.
The State machine is defined on a per session basis which is uniquely
identified by the RTSP session identifier. The session may contain
one or more media streams depending on state. If a single media
stream is part of the session it is in non-aggregated control. If
two or more is part of the session it is in aggregated control.
The below state machine is an informative description of the
protocols behavior. In case of ambiguity with the earlier parts of
this specification, the description in the earlier parts take
precedence.
B.1. States
The state machine contains three states, described below. For each
state there exists a table which shows which requests and events are
allowed and whether they will result in a state change.
Init: Initial state no session exists.
Ready: Session is ready to start playing.
Play: Session is playing, i.e. sending media stream data in the
direction S->C.
B.2. State variables
This representation of the state machine needs more than its state to
work. A small number of variables are also needed and they are
explained below.
NRM: The number of media streams part of this session.
RP: Resume point, the point in the presentation time line at which
a request to continue playing will resume from. A time format
for the variable is not mandated.
B.3. Abbreviations
To make the state tables more compact a number of abbreviations are
used, which are explained below.
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IFI: IF Implemented.
md: Media
PP: Pause Point, the point in the presentation time line at which
the presentation was paused.
Prs: Presentation, the complete multimedia presentation.
RedP: Redirect Point, the point in the presentation time line at
which a REDIRECT was specified to occur.
SES: Session.
B.4. State Tables
This section contains a table for each state. The table contains all
the requests and events that this state is allowed to act on. The
events which are method names are, unless noted, requests with the
given method in the direction client to server (C->S). In some cases
there exist one or more requisite. The response column tells what
type of response actions should be performed. Possible actions that
are requested for an event include: response codes, e.g., 200,
headers that need to be included in the response, setting of state
variables, or setting of other session related parameters. The new
state column tells which state the state machine changes to.
The response to a valid request meeting the requisites is normally a
2xx (SUCCESS) unless otherwise noted in the response column. The
exceptions need to be given a response according to the response
column. If the request does not meet the requisite, is erroneous or
some other type of error occurs, the appropriate response code is to
be sent. If the response code is a 4xx the session state is
unchanged. A response code of 3rr will result in that the session is
ended and its state is changed to Init. A response code of 304
results in no state change. However, there are restrictions to when
a 3rr response may be used. A 5xx response does not result in any
change of the session state, except if the error is not possible to
recover from. A unrecoverable error results in the ending of the
session. As it in the general case can't be determined if it was a
unrecoverable error or not the client will be required to test. In
the case that the next request after a 5xx is responded with 454
(Session Not Found) the client knows that the session has ended. For
any request message that cannot be responded to within the time
defined in Section 10.4, a 100 response must be sent.
The server will timeout the session after the period of time
specified in the SETUP response, if no activity from the client is
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detected. Therefore there exists a timeout event for all states
except Init.
In the case that NRM = 1 the presentation URI is equal to the media
URI or a specified presentation URI. For NRM > 1 the presentation
URI needs to be other than any of the medias that are part of the
session. This applies to all states.
+---------------+-----------------+---------------------------------+
| Event | Prerequisite | Response |
+---------------+-----------------+---------------------------------+
| DESCRIBE | Needs REDIRECT | 3rr, Redirect |
| | | |
| DESCRIBE | | 200, Session description |
| | | |
| OPTIONS | Session ID | 200, Reset session timeout |
| | | timer |
| | | |
| OPTIONS | | 200 |
| | | |
| SET_PARAMETER | Valid parameter | 200, change value of parameter |
| | | |
| GET_PARAMETER | Valid parameter | 200, return value of parameter |
+---------------+-----------------+---------------------------------+
Table 13: None state-machine changing events
The methods in Table 13 do not have any effect on the state machine
or the state variables. However, some methods do change other
session related parameters, for example SET_PARAMETER which will set
the parameter(s) specified in its body. Also all of these methods
that allow Session header will also update the keep-alive timer for
the session.
+------------------+----------------+-----------+-------------------+
| Action | Requisite | New State | Response |
+------------------+----------------+-----------+-------------------+
| SETUP | | Ready | NRM=1, RP=0.0 |
| | | | |
| SETUP | Needs Redirect | Init | 3rr Redirect |
| | | | |
| S -> C: REDIRECT | No Session hdr | Init | Terminate all SES |
+------------------+----------------+-----------+-------------------+
Table 14: State: Init
The initial state of the state machine, see Table 14 can only be left
by processing a correct SETUP request. As seen in the table the two
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state variables are also set by a correct request. This table also
shows that a correct SETUP can in some cases be redirected to another
URI and/or server by a 3rr response.
+-------------+------------------------+---------+------------------+
| Action | Requisite | New | Response |
| | | State | |
+-------------+------------------------+---------+------------------+
| SETUP | New URI | Ready | NRM +=1 |
| | | | |
| SETUP | URI Setup prior | Ready | Change transport |
| | | | param |
| | | | |
| TEARDOWN | Prs URI, | Init | No session hdr, |
| | | | NRM = 0 |
| | | | |
| TEARDOWN | md URI,NRM=1 | Init | No Session hdr, |
| | | | NRM = 0 |
| | | | |
| TEARDOWN | md URI,NRM>1 | Ready | Session hdr, NRM |
| | | | -= 1 |
| | | | |
| PLAY | Prs URI, No range | Play | Play from RP |
| | | | |
| PLAY | Prs URI, Range | Play | According to |
| | | | range |
| | | | |
| PLAY | md URI, NRM=1, Range | Play | According to |
| | | | range |
| | | | |
| PLAY | md URI, NRM=1 | Play | Play from RP |
| | | | |
| PAUSE | Prs URI | Ready | Return PP |
| | | | |
| SC:REDIRECT | Terminate-Reason | Ready | Set RedP |
| | | | |
| SC:REDIRECT | No Terminate-Reason | Init | Session is |
| | time parameter | | removed |
| | | | |
| Timeout | | Init | |
| | | | |
| RedP | | Init | TEARDOWN of |
| reached | | | session |
+-------------+------------------------+---------+------------------+
Table 15: State: Ready
In the Ready state, see Table 15, some of the actions are depending
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on the number of media streams (NRM) in the session, i.e., aggregated
or non-aggregated control. A SETUP request in the Ready state can
either add one more media stream to the session or, if the media
stream (same URI) already is part of the session, change the
transport parameters. TEARDOWN is depending on both the Request-URI
and the number of media streams within the session. If the Request-
URI is the presentations URI the whole session is torn down. If a
media URI is used in the TEARDOWN request and more than one media
exists in the session, the session will remain and a session header
is returned in the response. If only a single media stream remains
in the session when performing a TEARDOWN with a media URI the
session is removed. The number of media streams remaining after
tearing down a media stream determines the new state.
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+----------------+-----------------------+--------+-----------------+
| Action | Requisite | New | Response |
| | | State | |
+----------------+-----------------------+--------+-----------------+
| PAUSE | Prs URI | Ready | Set RP to |
| | | | present point |
| | | | |
| End of media | All media | Play | Set RP = End of |
| | | | media |
| | | | |
| End of range | | Play | Set RP = End of |
| | | | range |
| | | | |
| PLAY | Prs URI, No range | Play | Play from |
| | | | present point |
| | | | |
| PLAY | Prs URI, Range | Play | According to |
| | | | range |
| | | | |
| SC:PLAY_NOTIFY | | Play | 200 |
| | | | |
| SETUP | New URI | Play | 455 |
| | | | |
| SETUP | Setuped URI | Play | 455 |
| | | | |
| SETUP | Setuped URI, IFI | Play | Change |
| | | | transport |
| | | | param. |
| | | | |
| TEARDOWN | Prs URI | Init | No session hdr |
| | | | |
| TEARDOWN | md URI,NRM=1 | Init | No Session hdr, |
| | | | NRM=0 |
| | | | |
| TEARDOWN | md URI | Play | 455 |
| | | | |
| SC:REDIRECT | Terminate Reason with | Play | Set RedP |
| | Time parameter | | |
| | | | |
| SC:REDIRECT | | Init | Session is |
| | | | removed |
| | | | |
| RedP reached | | Init | TEARDOWN of |
| | | | session |
| | | | |
| Timeout | | Init | Stop Media |
| | | | playout |
+----------------+-----------------------+--------+-----------------+
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Table 16: State: Play
The Play state table, see Table 16, contains a number of requests
that need a presentation URI (labeled as Prs URI) to work on (i.e.,
the presentation URI has to be used as the Request-URI). This is due
to the exclusion of non-aggregated stream control in sessions with
more than one media stream.
To avoid inconsistencies between the client and server, automatic
state transitions are avoided. This can be seen at for example "End
of media" event when all media has finished playing, the session
still remains in Play state. An explicit PAUSE request needs to be
sent to change the state to Ready. It may appear that there exist
automatic transitions in "RedP reached" and "PP reached". However,
they are requested and acknowledged before they take place. The time
at which the transition will happen is known by looking at the range
header. If the client sends a request close in time to these
transitions it needs to be prepared for receiving error messages, as
the state may or may not have changed.
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Appendix C. Media Transport Alternatives
This section defines how certain combinations of protocols, profiles
and lower transports are used. This includes the usage of the
Transport header's source and destination address parameters
"src_addr" and "dest_addr".
C.1. RTP
This section defines the interaction of RTSP with respect to the RTP
protocol [RFC3550]. It also defines any necessary media transport
signaling with regards to RTP.
The available RTP profiles and lower layer transports are described
below along with rules on signaling the available combinations.
C.1.1. AVP
The usage of the "RTP Profile for Audio and Video Conferences with
Minimal Control" [RFC3551] when using RTP for media transport over
different lower layer transport protocols is defined below in regards
to RTSP.
One such case is defined within this document: the use of embedded
(interleaved) binary data as defined in Section 14. The usage of
this method is indicated by including the "interleaved" parameter.
When using embedded binary data the "src_addr" and "dest_addr" MUST
NOT be used. This addressing and multiplexing is used as defined
with use of channel numbers and the interleaved parameter.
C.1.2. AVP/UDP
This part describes sending of RTP [RFC3550] over lower transport
layer UDP [RFC0768] according to the profile "RTP Profile for Audio
and Video Conferences with Minimal Control" defined in RFC 3551
[RFC3551]. This profile requires one or two uni- or bi-directional
UDP flows per media stream. The first UDP flow is for RTP and the
second is for RTCP. Multiplexing of RTP and RTCP (Appendix C.1.6.4)
MAY be used, in which case a single UDP flow is used for both parts.
Embedding of RTP data with the RTSP messages, in accordance with
Section 14, SHOULD NOT be performed when RTSP messages are
transported over unreliable transport protocols, like UDP [RFC0768].
The RTP/UDP and RTCP/UDP flows can be established using the Transport
header's "src_addr", and "dest_addr" parameters.
In RTSP PLAY mode, the transmission of RTP packets from client to
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server is unspecified. The behavior in regards to such RTP packets
MAY be defined in future.
The "src_addr" and "dest_addr" parameters are used in the following
way for media delivery and playback mode, i.e. Mode=PLAY:
o The "src_addr" and "dest_addr" parameters MUST contain either 1 or
2 address specifications. Note that two address specifications
MAY be provided even if RTP and RTCP multiplexing is negotiated.
o Each address specification for RTP/AVP/UDP or RTP/AVP/TCP MUST
contain either:
* both an address and a port number, or
* a port number without an address.
o The first address specification given in either of the parameters
applies to the RTP stream. The second specification if present
applies to the RTCP stream, unless in case RTP and RTCP
multiplexing is negotiated where both RTP and RTCP will use the
first specification.
o The RTP/UDP packets from the server to the client MUST be sent to
the address and port given by the first address specification of
the "dest_addr" parameter.
o The RTCP/UDP packets from the server to the client MUST be sent to
the address and port given by the second address specification of
the "dest_addr" parameter, unless RTP and RTCP multiplexing has
been negotiated, in which case RTCP MUST be sent to the first
address specification. If no second pair is specified and RTP and
RTCP multiplexing has not been negotiated, RTCP MUST NOT be sent.
o The RTCP/UDP packets from the client to the server MUST be sent to
the address and port given by the second address specification of
the "src_addr" parameter, unless RTP and RTCP multiplexing has
been negotiated, in which case RTCP MUST be sent to the first
address specification. If no second pair is specified and RTP and
RTCP multiplexing has not been negotiated, RTCP MUST NOT be sent.
o The RTP/UDP packets from the client to the server MUST be sent to
the address and port given by the first address specification of
the "src_addr" parameter.
o RTP and RTCP Packets SHOULD be sent from the corresponding
receiver port, i.e. RTCP packets from the server should be sent
from the "src_addr" parameters second address port pair, unless
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RTP and RTCP multiplexing has been negotiated in which case the
first address port pair is used.
C.1.3. AVPF/UDP
The RTP profile "Extended RTP Profile for RTCP-based Feedback (RTP/
AVPF)" [RFC4585] MAY be used as RTP profiles in sessions using RTP.
All that is defined for AVP MUST also apply for AVPF.
The usage of AVPF is indicated by the media initialization protocol
used. In the case of SDP it is indicated by media lines (m=)
containing the profile RTP/AVPF. That SDP MAY also contain further
AVPF related SDP attributes configuring the AVPF session regarding
reporting interval and feedback messages to be used [RFC4585]. This
configuration MUST be followed.
C.1.4. SAVP/UDP
The RTP profile "The Secure Real-time Transport Protocol (SRTP)"
[RFC3711] is an RTP profile (SAVP) that MAY be used in RTSP sessions
using RTP. All that is defined for AVP MUST also apply for SAVP.
The usage of SRTP requires that a security context is established.
The default key-management unless otherwise signalled SHALL be MIKEY
in RSA-R mode as defined in Appendix C.1.4.1, and not according to
the procedure defined in "Key Management Extensions for Session
Description Protocol (SDP) and Real Time Streaming Protocol (RTSP)"
[RFC4567]. The reason is that RFC 4567 sends the initial MIKEY
message in SDP, thus both requiring the usage of the DESCRIBE method
and forcing the server to keep state for clients performing DESCRIBE
in anticipation that they might require key management.
MIKEY is selected as default method for establishing SRTP
cryptographic context within an RTSP session as it can be embedded in
the RTSP messages, while still ensuring confidentiality of content of
the keying material, even when using hop-by-hop TLS security for the
RTSP messages. This method does also support pipelining of the RTSP
messages.
C.1.4.1. MIKEY Key Establishment
This method for using MIKEY [RFC3830] to establish the SRTP
cryptographic context is initiated in the client's SETUP request, and
the server's response to the SETUP carries the MIKEY response. This
ensures that the crypto context establishment happens simultaneously
with the establishment of the media stream being protected. By using
MIKEY's RSA-R mode [RFC4738] the client can be the initiator and
still allow the server to set the parameters in accordance with the
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actual media stream.
The SRTP cryptographic context establishment is done according to the
following process:
1. The client determines that SAVP or SAVPF shall be used from
media description format, e.g. SDP. If no other key management
method is explicitly signalled, then MIKEY SHALL be used as
defined herein. This specification does not specify an explicit
method for indicating this SRTP cryptographic context
establishment method, but future specifications may.
2. The client SHALL establish a TLS connection for RTSP messages,
directly or hop by hop with the server. If hop-by-hop TLS
security is used, the User method SHALL be indicated in the
Accept-Credentials header. We do note that using hop-by-hop
does allow the proxy to insert itself as a man in the middle
also in the MIKEY exchange by providing one of its certificates,
rather than the server's in the Connection-Credentials header.
The client SHALL therefore validate the server certificate.
3. The client retrieves the server's certificate from a direct TLS
connection, or if hop by hop from Connection-Credentials header.
The client then checks that the server certificate is valid and
belongs to the server.
4. The client forms the MIKEY Initiator message using RSA-R mode in
unicast mode as specified in [RFC4738]. The client SHOULD use
the same certificate for TLS and in MIKEY to enable the server
to bind the two together. The client's certificate SHALL be
included in the MIKEY message. The client SHALL indicate its
SRTP capabilities in the message.
5. The MIKEY message from the previous step is base64 [RFC4648]
encoded and becomes the value of the MIKEY parameter that is
included in the transport specification(s) that specifies a SRTP
based profile (SAVP, SAVPF) in the SETUP request.
6. Any proxy encountering the MIKEY parameter SHALL forward it
without modification. A proxy requiring to understand transport
specification which doesn't support SAVP/SAVPF with MIKEY will
discard the whole transport specification. Most types of
proxies can easily support SAVP and SAVPF with MIKEY. If
possible bypassing the proxy should be tried.
7. The server upon receiving the SETUP request, will need to decide
upon the transport specification to use, if multiple are
included by the client. In the determination of which transport
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specifications that are supported and preferred, the server
SHOULD decode the MIKEY message to take the embedded SRTP
parameters into account. If all transport specs require SRTP
but no MIKEY parameter or other supported keying method is
included, the server SHALL respond with 403.
8. Upon generating a response the following outcomes can occur:
* A transport spec not using SRTP and MIKEY is selected. Thus
the response will not contain any MIKEY parameter.
* A transport spec using SRTP and MIKEY is selected but an
error is encountered in the MIKEY processing. In that case
an RTSP error response code of 466 "Key Management Error"
SHALL be used. A MIKEY message describing the error MAY be
included.
* A transport spec using SRTP and MIKEY is selected and a MIKEY
response message can be created. The server SHOULD use the
same certificate for TLS and in MIKEY to enable client to
bind the two together. If a different certificate is used it
SHALL be included in the MIKEY message. It is RECOMMENDED
that the envelope key cache type is set to 'Cache' and that a
single envelope key is reused for all MIKEY messages to the
client. That message is included in the MIKEY parameter part
of the single selected transport specification in the SETUP
response. The server will set the SRTP parameters as
preferred for this media stream within the supported range by
the client.
9. The server transmits the SETUP response back to the client.
10. The client receives the SETUP response and if the response code
indicates a successful request it decodes the MIKEY message and
establishes the SRTP cryptographic context from the parameters
in the MIKEY response.
In the above method the client's certificate may be self-signed in
cases where the client's identity is not necessary to authenticate
and the security goal is only to ensure that the RTSP signaling
client is the same as the one receiving the SRTP security context.
C.1.5. SAVPF/UDP
The RTP profile "Extended Secure RTP Profile for RTCP-based Feedback
(RTP/SAVPF)" [RFC5124] is an RTP profile (SAVPF) that MAY be used in
RTSP sessions using RTP. All that is defined for AVPF MUST also
apply for SAVPF.
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The usage of SRTP requires that a cryptographic context is
established. The default mechanism for establishing that security
association is to use MIKEY[RFC3830] with RTSP as defined in
Appendix C.1.4.1.
C.1.6. RTCP usage with RTSP
RTCP has several usages when RTP is used for media transport as
explained below. Due to that RTCP MUST be supported if an RTSP agent
handles RTP.
C.1.6.1. Media synchronization
RTCP provides media synchronization and clock drift compensation.
The initial media synchronization is available from RTP-Info header.
However, to be able to handle any clock drift between the media
streams, RTCP is needed.
C.1.6.2. RTSP Session keep-alive
RTCP traffic from the RTSP client to the RTSP server MUST function as
keep-alive. This requires an RTSP server supporting RTP to use the
received RTCP packets as indications that the client desires the
related RTSP session to be kept alive.
C.1.6.3. Bit-rate adaption
RTCP Receiver reports and any additional feedback from the client
MUST be used to adapt the bit-rate used over the transport for all
cases when RTP is sent over UDP. An RTP sender without reserved
resources MUST NOT use more than its fair share of the available
resources. This can be determined by comparing on short to medium
term (some seconds) the used bit-rate and adapt it so that the RTP
sender sends at a bit-rate comparable to what a TCP sender would
achieve on average over the same path.
C.1.6.4. RTP and RTCP Multiplexing
RTSP can be used to negotiate the usage of RTP and RTCP multiplexing
as described in [RFC5761]. This allows servers and client to reduce
the amount of resources required for the session by only requiring
one underlying transport stream per media stream instead of two when
using RTP and RTCP. This lessens the server port consumption and
also the necessary state and keep-alive work when operating across
Network and Address Translators [RFC2663].
Content must be prepared with some consideration for RTP and RTCP
multiplexing, mainly ensuring that the RTP payload types used do not
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collide with the ones used for RTCP packet types. This option likely
needs explicit support from the content unless the RTP payload types
can be remapped by the server and that is correctly reflected in the
session description. Beyond that support of this feature should come
at little cost and much gain.
It is recommended that if the content and server support RTP and RTCP
multiplexing that this is indicated in the session description, for
example using the SDP attribute "a=rtcp-mux". If the SDP message
contains the a=rtcp-mux attribute for a media stream, the server MUST
support RTP and RTCP multiplexing. If indicated or otherwise desired
by the client it can include the Transport parameter "RTCP-mux" in
any transport specification where it desires to use RTCP-mux. The
server will indicate if it supports RTCP-mux. Servers and Clients
SHOULD support RTP and RTCP multiplexing.
For capability exchange, an RTSP feature tag for RTP and RTCP
multiplexing is defined: "setup.rtp.rtcp.mux".
To minimize the risk of negotiation failure while using RTP and RTCP
multiplexing some recommendations are here provided. If the session
description includes explicit indication of support (a=rtcp-mux in
SDP), then a RTSP agent can safely create a SETUP request with a
transport specification with only a single dest_addr parameter
address specification. If no such explicit indication is provided,
then even if the feature tag "setup.rtp.rtcp.mux" is provided in a
Supported header by the RTSP server or the feature tag included in
the Required header in the SETUP request, the media resource may not
support RTP and RTCP multiplexing. Thus, to maximize the probability
of successful negotiation the RTSP agent is recommended to include
two dest_addr parameter address specifications in the first or first
set (if pipelining is used) of SETUP request(s) for any media
resource aggregate. That way the RTSP server can either accept RTP
and RTCP multiplexing and only use the first address specification,
and if not use both specifications. The RTSP agent after having
received the response for a successful negotiation of the usage of
RTP and RTCP multiplexing, can then release the resources associated
with the second address specification.
C.2. RTP over TCP
Transport of RTP over TCP can be done in two ways: over independent
TCP connections using RFC 4571 [RFC4571] or interleaved in the RTSP
control connection. In both cases the protocol MUST be "rtp" and the
lower layer MUST be TCP. The profile may be any of the above
specified ones; AVP, AVPF, SAVP or SAVPF.
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C.2.1. Interleaved RTP over TCP
The use of embedded (interleaved) binary data transported on the RTSP
connection is possible as specified in Section 14. When using this
declared combination of interleaved binary data the RTSP messages
MUST be transported over TCP. TLS may or may not be used. If TLS is
used both RTSP messages and the binary data will be protected by TLS.
One should, however, consider that this will result in all media
streams go through any proxy. Using independent TCP connections can
avoid that issue.
C.2.2. RTP over independent TCP
In this Appendix, we describe the sending of RTP [RFC3550] over lower
transport layer TCP [RFC0793] according to "Framing Real-time
Transport Protocol (RTP) and RTP Control Protocol (RTCP) Packets over
Connection-Oriented Transport" [RFC4571]. This Appendix adapts the
guidelines for using RTP over TCP within SIP/SDP [RFC4145] to work
with RTSP.
A client codes the support of RTP over independent TCP by specifying
an RTP/AVP/TCP transport option without an interleaved parameter in
the Transport line of a SETUP request. This transport option MUST
include the "unicast" parameter.
If the client wishes to use RTP with RTCP, two address specifications
needs to be included in the dest_addr parameter. If the client
wishes to use RTP without RTCP, one address specification is included
in the dest_addr parameter. If the client wishes to multiplex RTP
and RTCP on a single transport flow (see Appendix C.1.6.4), one or
two address specifications are included in the dest_addr parameter in
addition to the RTCP-mux transport parameter. Two address
specifications are allowed to allow successful negotiation when
server or content can't support RTP and RTCP multiplexing. Ordering
rules of dest_addr ports follow the rules for RTP/AVP/UDP.
If the client wishes to play the active role in initiating the TCP
connection, it MAY set the "setup" parameter (See Section 18.52) on
the Transport line to be "active", or it MAY omit the setup
parameter, as active is the default. If the client signals the
active role, the ports in the address specifications in the dest_addr
parameter MUST be set to 9 (the discard port).
If the client wishes to play the passive role in TCP connection
initiation, it MUST set the "setup" parameter on the Transport line
to be "passive". If the client is able to assume the active or the
passive role, it MUST set the "setup" parameter on the Transport line
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to be "actpass". In either case, the dest_addr parameter's address
specification port value for RTP MUST be set to the TCP port number
on which the client is expecting to receive the TCP connection for
RTP, and the dest_addr's address specification port value for RTCP
MUST be set to the TCP port number on which the client is expecting
to receive the TCP connection for RTCP. In the case that the client
wishes to multiplex RTP and RTCP on a single transport flow, the
RTCP-mux parameter is included and one or two dest_addr parameter
address specifications are included, as mentioned earlier in this
section.
If upon receipt of a non-interleaved RTP/AVP/TCP SETUP request, a
server decides to accept this requested option, the 2xx reply MUST
contain a Transport option that specifies RTP/AVP/TCP (without using
the interleaved parameter, and with using the unicast parameter).
The dest_addr parameter value MUST be echoed from the parameter value
in the client request unless the destination address (only port) was
not provided in which case the server MAY include the source address
of the RTSP TCP connection with the port number unchanged.
In addition, the server reply MUST set the setup parameter on the
Transport line, to indicate the role the server will play in the
connection setup. Permissible values are "active" (if a client set
"setup" to "passive" or "actpass") and "passive" (if a client set
"setup" to "active" or "actpass").
If a server sets "setup" to "passive", the "src_addr" in the reply
MUST indicate the ports the server is willing to receive an TCP
connection for RTP and (if the client requested an TCP connection for
RTCP by specifying two dest_addr address specifications) an TCP/RTCP
connection. If a server sets "setup" to "active", the ports
specified in "src_addr" address specifications MUST be set to 9. The
server MAY use the "ssrc" parameter, following the guidance in
Section 18.52. The server sets only one address specification in the
case that the client has indicated only a single address
specification or in case RTP and RTCP multiplexing was requested and
accepted by server. Port ordering for src_addr follows the rules for
RTP/AVP/UDP.
Servers MUST support taking the passive role and MAY support taking
the active role. Servers with a public IP address take the passive
role, thus enabling clients behind NATs and Firewalls a better chance
of successful connect to the server by actively connecting outwards.
Therefore the clients are RECOMMENDED to take the active role.
After sending (receiving) a 2xx reply for a SETUP method for a non-
interleaved RTP/AVP/TCP media stream, the active party SHOULD
initiate the TCP connection as soon as possible. The client MUST NOT
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send a PLAY request prior to the establishment of all the TCP
connections negotiated using SETUP for the session. In case the
server receives a PLAY request in a session that has not yet
established all the TCP connections, it MUST respond using the 464
"Data Transport Not Ready Yet" (Section 17.4.29) error code.
Once the PLAY request for a media resource transported over non-
interleaved RTP/AVP/TCP occurs, media begins to flow from server to
client over the RTP TCP connection, and RTCP packets flow
bidirectionally over the RTCP TCP connection. Unless RTP and RTCP
multiplexing has been negotiated in which case RTP and RTCP will flow
over a common TCP connection. As in the RTP/UDP case, client to
server traffic on a RTP only TCP session is unspecified by this memo.
The packets that travel on these connections MUST be framed using the
protocol defined in [RFC4571], not by the framing defined for
interleaving RTP over the RTSP control connection defined in
Section 14.
A successful PAUSE request for a media being transported over RTP/
AVP/TCP pauses the flow of packets over the connections, without
closing the connections. A successful TEARDOWN request signals that
the TCP connections for RTP and RTCP are to be closed by the RTSP
client as soon as possible.
Subsequent SETUP requests on an already-SETUP RTP/AVP/TCP URI may be
ambiguous in the following way: does the client wish to open up new
TCP connection for RTP or RTCP for the URI, or does the client wish
to continue using the existing TCP connections? The client SHOULD
use the "connection" parameter (defined in Section 18.52) on the
Transport line to make its intention clear (by setting "connection"
to "new" if new connections are needed, and by setting "connection"
to "existing" if the existing connections are to be used). After a
2xx reply for a SETUP request for a new connection, parties should
close the pre-existing connections, after waiting a suitable period
for any stray RTP or RTCP packets to arrive.
The usage of SRTP, i.e., either SAVP or SAVPF profiles, requires that
a security association is established. The default mechanism for
establishing that security association is to use MIKEY[RFC3830] with
RTSP as defined Appendix C.1.4.1.
Below, we rewrite part of the example media on demand example shown
in Appendix A.1 to use RTP/AVP/TCP non-interleaved:
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C->M: DESCRIBE rtsp://example.com/twister.3gp RTSP/2.0
CSeq: 1
User-Agent: PhonyClient/1.2
M->C: RTSP/2.0 200 OK
CSeq: 1
Server: PhonyServer/1.0
Date: Thu, 23 Jan 1997 15:35:06 GMT
Content-Type: application/sdp
Content-Length: 227
Content-Base: rtsp://example.com/twister.3gp/
Expires: 24 Jan 1997 15:35:06 GMT
v=0
o=- 2890844256 2890842807 IN IP4 198.51.100.34
s=RTSP Session
i=An Example of RTSP Session Usage
e=adm@example.com
c=IN IP4 0.0.0.0
a=control: *
a=range:npt=0-0:10:34.10
t=0 0
m=audio 0 RTP/AVP 0
a=control: trackID=1
C->M: SETUP rtsp://example.com/twister.3gp/trackID=1 RTSP/2.0
CSeq: 2
User-Agent: PhonyClient/1.2
Require: play.basic
Transport: RTP/AVP/TCP;unicast;dest_addr=":9"/":9";
setup=active;connection=new
Accept-Ranges: NPT, SMPTE, UTC
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M->C: RTSP/2.0 200 OK
CSeq: 2
Server: PhonyServer/1.0
Transport: RTP/AVP/TCP;unicast;
dest_addr=":9"/":9";
src_addr="198.51.100.5:53478"/"198.51.100:54091";
setup=passive;connection=new;ssrc=93CB001E
Session: 12345678
Expires: 24 Jan 1997 15:35:12 GMT
Date: 23 Jan 1997 15:35:12 GMT
Accept-Ranges: NPT
Media-Properties: Random-Access=0.8, Immutable, Unlimited
C->M: TCP Connection Establishment x2
C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0
CSeq: 4
User-Agent: PhonyClient/1.2
Range: npt=30-
Session: 12345678
M->C: RTSP/2.0 200 OK
CSeq: 4
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:35:14 GMT
Session: 12345678
Range: npt=30-623.10
Seek-Style: First-Prior
RTP-Info: url="rtsp://example.com/twister.3gp/trackID=1"
ssrc=4F312DD8:seq=54321;rtptime=2876889
C.3. Handling Media Clock Time Jumps in the RTP Media Layer
RTSP allows media clients to control selected, non-contiguous
sections of media presentations, rendering those streams with an RTP
media layer [RFC3550]. Two cases occur, the first is when a new PLAY
request replaces an old ongoing request and the new request results
in a jump in the media. This should produce in the RTP layer a
continuous media stream. A client may also directly following a
completed PLAY request perform a new PLAY request. This will result
in some gap in the media layer. The below text will look into both
cases.
A PLAY request that replaces an ongoing request allows the media
layer rendering the RTP stream without being affected by jumps in
media clock time. The RTP timestamps for the new media range is set
so that they become continuous with the previous media range in the
previous request. The RTP sequence number for the first packet in
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the new range will be the next following the last packet in the
previous range, i.e. monotonically increasing. The goal is to allow
the media rendering layer to work without interruption or
reconfiguration across the jumps in media clock. This should be
possible in all cases of replaced PLAY requests for media that has
random-access properties. In this case care is needed to align
frames or similar media dependent structures.
In cases where jumps in media clock time are a result of RTSP
signaling operations arriving after a completed PLAY operation, the
request timing will result in that media becomes non-continuous. The
server becomes unable to send the media so that it arrives timely and
still carry timestamps to make the media stream continuous. In these
cases the server will produce RTP streams where there are gaps in the
RTP timeline for the media. In such cases, if the media has frame
structure, aligning the timestamp for the next frame with the
previous structure reduces the burden to render this media. The gap
should represent the time the server hasn't been serving media, e.g.
the time between the end of the media stream or a PAUSE request and
the new PLAY request. In these cases the RTP sequence number would
normally be monotonically increasing across the gap.
For RTSP sessions with media that lacks random access properties,
such as live streams, any media clock jump is commonly the result of
a correspondingly long pause of delivery. The RTP timestamp will
have increased in direct proportion to the duration of the paused
delivery. Note also that in this case the RTP sequence number should
be the next packet number. If not, the RTCP packet loss reporting
will indicate as loss all packets not received between the point of
pausing and later resuming. This may trigger congestion avoidance
mechanisms. An allowed exception from the above recommendation on
monotonically increasing RTP sequence number is live media streams,
likely being relayed. In this case, when the client resumes
delivery, it will get the media that is currently being delivered to
the server itself. For this type of basic delivery of live streams
to multiple users over unicast, individual rewriting of RTP sequence
numbers becomes quite a burden. For solutions that anyway caches
media, timeshifts, etc, the rewriting should be a minor issue.
The goal when handling jumps in media clock time is that the provided
stream is continuous without gaps in RTP timestamp or sequence
number. However, when delivery has been halted for some reason the
RTP timestamp when resuming MUST represent the duration the delivery
was halted. RTP sequence number MUST generally be the next number,
i.e. monotonically increasing modulo 65536. For media resources with
the properties Time-Progressing and Time-Duration=0.0 the server MAY
create RTP media streams with RTP sequence number jumps in them due
to the client first halting delivery and later resuming it (PAUSE and
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then later PLAY). However, servers utilizing this exception must
take into consideration the resulting RTCP receiver reports that
likely contain loss reports for all the packets part of the
discontinuity. A client cannot rely on that a server will align when
resuming playing even if it is RECOMMENDED. The RTP-Info header will
provide information on how the server acts in each case.
We cannot assume that the RTSP client can communicate with the RTP
media agent, as the two may be independent processes. If the RTP
timestamp shows the same gap as the NPT, the media agent will
assume that there is a pause in the presentation. If the jump in
NPT is large enough, the RTP timestamp may roll over and the media
agent may believe later packets to be duplicates of packets just
played out. Having the RTP timestamp jump will also affect the
RTCP measurements based on this.
As an example, assume an RTP timestamp frequency of 8000 Hz, a
packetization interval of 100 ms and an initial sequence number and
timestamp of zero.
C->S: PLAY rtsp://example.com/fizzle RTSP/2.0
CSeq: 4
Session: abcdefgh
Range: npt=10-15
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 4
Session: abcdefgh
Range: npt=10-15
RTP-Info: url="rtsp://example.com/fizzle/audiotrack"
ssrc=0D12F123:seq=0;rtptime=0
The ensuing RTP data stream is depicted below:
S -> C: RTP packet - seq = 0, rtptime = 0, NPT time = 10s
S -> C: RTP packet - seq = 1, rtptime = 800, NPT time = 10.1s
. . .
S -> C: RTP packet - seq = 49, rtptime = 39200, NPT time = 14.9s
Upon the completion of the requested delivery the server sends a
PLAY_NOTIFY
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S->C: PLAY_NOTIFY rtsp://example.com/fizzle RTSP/2.0
CSeq: 5
Notify-Reason: end-of-stream
Request-Status: cseq=4 status=200 reason="OK"
Range: npt=-15
RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
ssrc=0D12F123:seq=49;rtptime=39200
Session: abcdefgh
C->S: RTSP/2.0 200 OK
CSeq: 5
User-Agent: PhonyClient/1.2
Upon the completion of the play range, the client follows up with a
request to PLAY from a new NPT.
C->S: PLAY rtsp://example.com/fizzle RTSP/2.0
CSeq: 6
Session: abcdefg
Range: npt=18-20
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 6
Session: abcdefg
Range: npt=18-20
RTP-Info: url="rtsp://example.com/fizzle/audiotrack"
ssrc=0D12F123:seq=50;rtptime=40100
The ensuing RTP data stream is depicted below:
S->C: RTP packet - seq = 50, rtptime = 40100, NPT time = 18s
S->C: RTP packet - seq = 51, rtptime = 40900, NPT time = 18.1s
. . .
S->C: RTP packet - seq = 69, rtptime = 55300, NPT time = 19.9s
In this example, first, NPT 10 through 15 is played, then the client
requests the server to skip ahead and play NPT 18 through 20. The
first segment is presented as RTP packets with sequence numbers 0
through 49 and timestamp 0 through 39,200. The second segment
consists of RTP packets with sequence number 50 through 69, with
timestamps 40,100 through 55,200. While there is a gap in the NPT,
there is no gap in the sequence number space of the RTP data stream.
The RTP timestamp gap is present in the above example due to the time
it takes to perform the second play request, in this case 12.5 ms
(100/8000).
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C.4. Handling RTP Timestamps after PAUSE
During a PAUSE / PLAY interaction in an RTSP session, the duration of
time for which the RTP transmission was halted MUST be reflected in
the RTP timestamp of each RTP stream. The duration can be calculated
for each RTP stream as the time elapsed from when the last RTP packet
was sent before the PAUSE request was received and when the first RTP
packet was sent after the subsequent PLAY request was received. The
duration includes all latency incurred and processing time required
to complete the request.
The RTP RFC [RFC3550] states that: The RTP timestamp for each unit
[packet] would be related to the wallclock time at which the unit
becomes current on the virtual presentation timeline.
In order to satisfy the requirements of [RFC3550], the RTP
timestamp space needs to increase continuously with real time.
While this is not optimal for stored media, it is required for RTP
and RTCP to function as intended. Using a continuous RTP
timestamp space allows the same timestamp model for both stored
and live media and allows better opportunity to integrate both
types of media under a single control.
As an example, assume a clock frequency of 8000 Hz, a packetization
interval of 100 ms and an initial sequence number and timestamp of
zero.
C->S: PLAY rtsp://example.com/fizzle RTSP/2.0
CSeq: 4
Session: abcdefg
Range: npt=10-15
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 4
Session: abcdefg
Range: npt=10-15
RTP-Info: url="rtsp://example.com/fizzle/audiotrack"
ssrc=0D12F123:seq=0;rtptime=0
The ensuing RTP data stream is depicted below:
S -> C: RTP packet - seq = 0, rtptime = 0, NPT time = 10s
S -> C: RTP packet - seq = 1, rtptime = 800, NPT time = 10.1s
S -> C: RTP packet - seq = 2, rtptime = 1600, NPT time = 10.2s
S -> C: RTP packet - seq = 3, rtptime = 2400, NPT time = 10.3s
The client then sends a PAUSE request:
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C->S: PAUSE rtsp://example.com/fizzle RTSP/2.0
CSeq: 5
Session: abcdefg
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 5
Session: abcdefg
Range: npt=10.4-15
20 seconds elapse and then the client sends a PLAY request. In
addition the server requires 15 ms to process the request:
C->S: PLAY rtsp://example.com/fizzle RTSP/2.0
CSeq: 6
Session: abcdefg
User-Agent: PhonyClient/1.2
S->C: RTSP/2.0 200 OK
CSeq: 6
Session: abcdefg
Range: npt=10.4-15
RTP-Info: url="rtsp://example.com/fizzle/audiotrack"
ssrc=0D12F123:seq=4;rtptime=164400
The ensuing RTP data stream is depicted below:
S -> C: RTP packet - seq = 4, rtptime = 164400, NPT time = 10.4s
S -> C: RTP packet - seq = 5, rtptime = 165200, NPT time = 10.5s
S -> C: RTP packet - seq = 6, rtptime = 166000, NPT time = 10.6s
First, NPT 10 through 10.3 is played, then a PAUSE is received by the
server. After 20 seconds a PLAY is received by the server which
takes 15 ms to process. The duration of time for which the session
was paused is reflected in the RTP timestamp of the RTP packets sent
after this PLAY request.
A client can use the RTSP range header and RTP-Info header to map NPT
time of a presentation with the RTP timestamp.
Note: In RFC 2326 [RFC2326], this matter was not clearly defined and
was misunderstood commonly. However, for RTSP 2.0 it is expected
that this will be handled correctly and no exception handling will be
required.
Note further: It may be required to reset some of the state to ensure
the correct media decoding and the usual jitter-buffer handling when
issuing a PLAY request.
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C.5. RTSP / RTP Integration
For certain datatypes, tight integration between the RTSP layer and
the RTP layer will be necessary. This by no means precludes the
above restrictions. Combined RTSP/RTP media clients should use the
RTP-Info field to determine whether incoming RTP packets were sent
before or after a seek or before or after a PAUSE.
C.6. Scaling with RTP
For scaling (see Section 18.44), RTP timestamps should correspond to
the rendering timing. For example, when playing video recorded at 30
frames/second at a scale of two and speed (Section 18.48) of one, the
server would drop every second frame to maintain and deliver video
packets with the normal timestamp spacing of 3,000 per frame, but NPT
would increase by 1/15 second for each video frame.
Note: The above scaling puts requirements on the media codec or a
media stream to support it. For example motion JPEG or other non-
predictive video coding can easier handle the above example.
C.7. Maintaining NPT synchronization with RTP timestamps
The client can maintain a correct display of NPT (Normal Play Time)
by noting the RTP timestamp value of the first packet arriving after
repositioning. The sequence parameter of the RTP-Info
(Section 18.43) header provides the first sequence number of the next
segment.
C.8. Continuous Audio
For continuous audio, the server SHOULD set the RTP marker bit at the
beginning of serving a new PLAY request or at jumps in timeline.
This allows the client to perform playout delay adaptation.
C.9. Multiple Sources in an RTP Session
Note that more than one SSRC MAY be sent in the media stream. If it
happens all sources are expected to be rendered simultaneously.
C.10. Usage of SSRCs and the RTCP BYE Message During an RTSP Session
The RTCP BYE message indicates the end of use of a given SSRC. If
all sources leave an RTP session, it can, in most cases, be assumed
to have ended. Therefore, a client or server MUST NOT send an RTCP
BYE message until it has finished using a SSRC. A server SHOULD keep
using a SSRC until the RTP session is terminated. Prolonging the use
of a SSRC allows the established synchronization context associated
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with that SSRC to be used to synchronize subsequent PLAY requests
even if the PLAY response is late.
An SSRC collision with the SSRC that transmits media does also have
consequences, as it will normally force the media sender to change
its SSRC in accordance with the RTP specification [RFC3550].
However, an RTSP server may wait and see if the client changes and
thus resolve the conflict to minimize the impact. As media sender
SSRC change will result in a loss of synchronization context, and
require any receiver to wait for RTCP sender reports for all media
requiring synchronization before being able to play out synchronized.
Due to these reasons a client joining a session should take care to
not select the same SSRC(s) as the server indicates in the ssrc
Transport header parameter. Any SSRC signalled in the Transport
header MUST be avoided. A client detecting a collision prior to
sending any RTP or RTCP messages SHALL also select a new SSRC.
C.11. Future Additions
It is the intention that any future protocol or profile regarding
media delivery and lower transport should be easy to add to RTSP.
This section provides the necessary steps that needs to be meet.
The following things needs to be considered when adding a new
protocol or profile for use with RTSP:
o The protocol or profile needs to define a name tag representing
it. This tag is required to be an ABNF "token" to be possible to
use in the Transport header specification.
o The useful combinations of protocol, profiles and lower layer
transport for this extension needs to be defined. For each
combination declare the necessary parameters to use in the
Transport header.
o For new media protocols the interaction with RTSP needs to be
addressed. One important factor will be the media
synchronization. It may be necessary to have new headers similar
to RTP info to carry this information.
o Discuss congestion control for media, especially if transport
without built in congestion control is used.
See the IANA section (Section 22) for information how to register new
attributes.
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Appendix D. Use of SDP for RTSP Session Descriptions
The Session Description Protocol (SDP, [RFC4566]) may be used to
describe streams or presentations in RTSP. This description is
typically returned in reply to a DESCRIBE request on an URI from a
server to a client, or received via HTTP from a server to a client.
This appendix describes how an SDP file determines the operation of
an RTSP session. Thus, it is worth pointing out that the
interpretation of the SDP is done in the context of the SDP receiver,
which is the one being configured. This is the same as in SAP
[RFC2974]; this differs from SDP Offer/Answer [RFC3264] where each
SDP is interpreted in the context of the agent providing it.
SDP as is provides no mechanism by which a client can distinguish,
without human guidance, between several media streams to be rendered
simultaneously and a set of alternatives (e.g., two audio streams
spoken in different languages). The SDP extension "Grouping of Media
Lines in the Session Description Protocol (SDP)" [RFC5888] provides
such functionality to some degree. Appendix D.4 describes the usage
of SDP media line grouping for RTSP.
D.1. Definitions
The terms "session-level", "media-level" and other key/attribute
names and values used in this appendix are to be used as defined in
SDP[RFC4566]:
D.1.1. Control URI
The "a=control:" attribute is used to convey the control URI. This
attribute is used both for the session and media descriptions. If
used for individual media, it indicates the URI to be used for
controlling that particular media stream. If found at the session
level, the attribute indicates the URI for aggregate control
(presentation URI). The session level URI MUST be different from any
media level URI. The presence of a session level control attribute
MUST be interpreted as support for aggregated control. The control
attribute MUST be present on media level unless the presentation only
contains a single media stream, in which case the attribute MAY be
present on the session level only and then also apply to that single
media stream.
ABNF for the attribute is defined in Section 20.3.
Example:
a=control:rtsp://example.com/foo
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This attribute MAY contain either relative or absolute URIs,
following the rules and conventions set out in RFC 3986 [RFC3986].
Implementations MUST look for a base URI in the following order:
1. the RTSP Content-Base field;
2. the RTSP Content-Location field;
3. the RTSP Request-URI.
If this attribute contains only an asterisk (*), then the URI MUST be
treated as if it were an empty embedded URI, and thus inherit the
entire base URI.
Note, RFC 2326 was very unclear on the processing of relative URI
and several RTSP 1.0 implementations at the point of publishing
this document did not perform RFC 3986 processing to determine the
resulting URI, instead simple concatenation is common. To avoid
this issue completely it is recommended to use absolute URI in the
SDP.
The URI handling for SDPs from container files need special
consideration. For example let's assume that a container file has
the URI: "rtsp://example.com/container.mp4". Let's further assume
this URI is the base URI, and that there is an absolute media level
URI: "rtsp://example.com/container.mp4/trackID=2". A relative media
level URI that resolves in accordance with RFC 3986 [RFC3986] to the
above given media URI is: "container.mp4/trackID=2". It is usually
not desirable to need to include in or modify the SDP stored within
the container file with the server local name of the container file.
To avoid this, one can modify the base URI used to include a trailing
slash, e.g. "rtsp://example.com/container.mp4/". In this case the
relative URI for the media will only need to be: "trackID=2".
However, this will also mean that using "*" in the SDP will result in
control URI including the trailing slash, i.e.
"rtsp://example.com/container.mp4/".
Note: The usage of TrackID in the above is not a standardized
form, but one example out of several similar strings such as
TrackID, Track_ID, StreamID that is used by different server
vendors to indicate a particular piece of media inside a container
file.
D.1.2. Media Streams
The "m=" field is used to enumerate the streams. It is expected that
all the specified streams will be rendered with appropriate
synchronization. If the session is over multicast, the port number
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indicated SHOULD be used for reception. The client MAY try to
override the destination port, through the Transport header. The
servers MAY allow this, the response will indicate if allowed or not.
If the session is unicast, the port numbers are the ones RECOMMENDED
by the server to the client, about which receiver ports to use; the
client MUST still include its receiver ports in its SETUP request.
The client MAY ignore this recommendation. If the server has no
preference, it SHOULD set the port number value to zero.
The "m=" lines contain information about which transport protocol,
profile, and possibly lower-layer is to be used for the media stream.
The combination of transport, profile and lower layer, like RTP/AVP/
UDP needs to be defined for how to be used with RTSP. The currently
defined combinations are defined in Appendix C, further combinations
MAY be specified.
Example:
m=audio 0 RTP/AVP 31
D.1.3. Payload Type(s)
The payload type(s) are specified in the "m=" line. In case the
payload type is a static payload type from RFC 3551 [RFC3551], no
other information may be required. In case it is a dynamic payload
type, the media attribute "rtpmap" is used to specify what the media
is. The "encoding name" within the "rtpmap" attribute may be one of
those specified in [RFC4856], or a media type registered with IANA
according to [RFC4855], or an experimental encoding as specified in
SDP [RFC4566]). Codec-specific parameters are not specified in this
field, but rather in the "fmtp" attribute described below.
The selection of the RTP payload type numbers used may be required to
consider RTP and RTCP Multiplexing [RFC5761] if that is to be
supported by the server.
D.1.4. Format-Specific Parameters
Format-specific parameters are conveyed using the "fmtp" media
attribute. The syntax of the "fmtp" attribute is specific to the
encoding(s) that the attribute refers to. Note that some of the
format specific parameters may be specified outside of the fmtp
parameters, like for example the "ptime" attribute for most audio
encodings.
D.1.5. Directionality of media stream
The SDP attributes "a=sendrecv", "a=recvonly" and "a=sendonly"
provide instructions about the direction the media streams flow
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within a session. When using RTSP the SDP can be delivered to a
client using either RTSP DESCRIBE or a number of RTSP external
methods, like HTTP, FTP, and email. Based on this the SDP applies to
how the RTSP client will see the complete session. Thus media
streams delivered from the RTSP server to the client, would be given
the "a=recvonly" attribute.
"a=recvonly" in a SDP provided to the RTSP client indicates that
media delivery will only occur in the direction from the RTSP server
to the client. SDP provided to the RTSP client that lacks any of the
directionality attributes (a=recvonly, a=sendonly, a=sendrecv) would
be interpreated as having a=sendrecv. At the time of writing there
exist no RTSP mode suitable for media traffic in the direction from
the RTSP client to the server. Thus all RTSP SDP SHOULD have
a=recvonly attribute when using the PLAY mode defined in this
document. If future modes are defined for media in client to server
direction, then usage of a=sendonly, or a=sendrecv may become
suitable to indicate intended media directions.
D.1.6. Range of Presentation
The "a=range" attribute defines the total time range of the stored
session or an individual media. Non-seekable live sessions can be
indicated as specified below, while the length of live sessions can
be deduced from the "t=" and "r=" SDP parameters.
The attribute is both a session and a media level attribute. For
presentations that contain media streams of the same duration, the
range attribute SHOULD only be used at session-level. In case of
different lengths the range attribute MUST be given at media level
for all media, and SHOULD NOT be given at session level. If the
attribute is present at both media level and session level the media
level values MUST be used.
Note: Usually one will specify the same length for all media, even if
there isn't media available for the full duration on all media.
However, that requires that the server accepts PLAY requests within
that range.
Servers MUST take care to provide RTSP Range (see Section 18.38)
values that are consistent with what is presented in the SDP for the
content. There is no reason for non dynamic content, like media
clips provided on demand to have inconsistent values. Inconsistent
values between the SDP and the actual values for the content handled
by the server is likely to generate some failure, like 457 "Invalid
Range", in case the client uses PLAY requests with a Range header.
In case the content is dynamic in length and it is infeasible to
provide a correct value in the SDP the server is recommended to
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describe this as non-seekable content (see below). The server MAY
override that property in the response to a PLAY request using the
correct values in the Range header.
The unit is specified first, followed by the value range. The units
and their values are as defined in Section 4.4, Section 4.5 and
Section 4.6 and MAY be extended with further formats. Any open ended
range (start-), i.e. without stop range, is of unspecified duration
and MUST be considered as non-seekable content unless this property
is overridden. Multiple instances carrying different clock formats
MAY be included at either session or media level.
ABNF for the attribute is defined in Section 20.3.
Examples:
a=range:npt=0-34.4368
a=range:clock=19971113T211503Z-19971113T220300Z
Non seekable stream of unknown duration:
a=range:npt=0-
D.1.7. Time of Availability
The "t=" field defines when the SDP is valid. For on-demand content
the server SHOULD indicate a stop time value for which it guarantees
the description to be valid, and a start time that is equal to or
before the time at which the DESCRIBE request was received. It MAY
also indicate start and stop times of 0, meaning that the session is
always available.
For sessions that are of live type, i.e. specific start time, unknown
stop time, likely unseekable, the "t=" and "r=" field SHOULD be used
to indicate the start time of the event. The stop time SHOULD be
given so that the live event will have ended at that time, while
still not be unnecessary long into the future.
D.1.8. Connection Information
In SDP used with RTSP, the "c=" field contains the destination
address for the media stream. If a multicast address is specified
the client SHOULD use this address in any SETUP request as
destination address, including any additional parameters, such as
TTL. For on-demand unicast streams and some multicast streams, the
destination address MAY be specified by the client via the SETUP
request, thus overriding any specified address. To identify streams
without a fixed destination address, where the client is required to
specify a destination address, the "c=" field SHOULD be set to a null
value. For addresses of type "IP4", this value MUST be "0.0.0.0",
and for type "IP6", this value MUST be "0:0:0:0:0:0:0:0" (can also be
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written as "::"), i.e. the unspecified address according to RFC 4291
[RFC4291].
D.1.9. Message Body Tag
The optional "a=mtag" attribute identifies a version of the session
description. It is opaque to the client. SETUP requests may include
this identifier in the If-Match field (see Section 18.23) to only
allow session establishment if this attribute value still corresponds
to that of the current description. The attribute value is opaque
and may contain any character allowed within SDP attribute values.
ABNF for the attribute is defined in Section 20.3.
Example:
a=mtag:"158bb3e7c7fd62ce67f12b533f06b83a"
One could argue that the "o=" field provides identical
functionality. However, it does so in a manner that would put
constraints on servers that need to support multiple session
description types other than SDP for the same piece of media
content.
D.2. Aggregate Control Not Available
If a presentation does not support aggregate control no session level
"a=control:" attribute is specified. For a SDP with multiple media
sections specified, each section will have its own control URI
specified via the "a=control:" attribute.
Example:
v=0
o=- 2890844256 2890842807 IN IP4 192.0.2.56
s=I came from a web page
e=adm@example.com
c=IN IP4 0.0.0.0
t=0 0
m=video 8002 RTP/AVP 31
a=control:rtsp://audio.example.com/movie.aud
m=audio 8004 RTP/AVP 3
a=control:rtsp://video.example.com/movie.vid
Note that the position of the control URI in the description implies
that the client establishes separate RTSP control sessions to the
servers audio.example.com and video.example.com.
It is recommended that an SDP file contains the complete media
initialization information even if it is delivered to the media
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client through non-RTSP means. This is necessary as there is no
mechanism to indicate that the client should request more detailed
media stream information via DESCRIBE.
D.3. Aggregate Control Available
In this scenario, the server has multiple streams that can be
controlled as a whole. In this case, there are both a media-level
"a=control:" attributes, which are used to specify the stream URIs,
and a session-level "a=control:" attribute which is used as the
Request-URI for aggregate control. If the media-level URI is
relative, it is resolved to absolute URIs according to Appendix D.1.1
above.
Example:
C->M: DESCRIBE rtsp://example.com/movie RTSP/2.0
CSeq: 1
User-Agent: PhonyClient/1.2
M->C: RTSP/2.0 200 OK
CSeq: 1
Date: Thu, 23 Jan 1997 15:35:06 GMT
Expires: Thu, 23 Jan 1997 16:35:06 GMT
Content-Type: application/sdp
Content-Base: rtsp://example.com/movie/
Content-Length: 227
v=0
o=- 2890844256 2890842807 IN IP4 192.0.2.211
s=I contain
i=<more info>
e=adm@example.com
c=IN IP4 0.0.0.0
a=control:*
t=0 0
m=video 8002 RTP/AVP 31
a=control:trackID=1
m=audio 8004 RTP/AVP 3
a=control:trackID=2
In this example, the client is recommended to establish a single RTSP
session to the server, and uses the URIs
rtsp://example.com/movie/trackID=1 and
rtsp://example.com/movie/trackID=2 to set up the video and audio
streams, respectively. The URI rtsp://example.com/movie/, which is
resolved from the "*", controls the whole presentation (movie).
A client is not required to issue SETUP requests for all streams
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within an aggregate object. Servers should allow the client to ask
for only a subset of the streams.
D.4. Grouping of Media Lines in SDP
For some types of media it is desirable to express a relationship
between various media components, for instance, for lip
synchronization or Scalable Video Codec (SVC) [RFC5583]. This
relationship is expressed on the SDP level by grouping of media
lines, as described in [RFC5888] and can be exposed to RTSP.
For RTSP it is mainly important to know how to handle grouped medias
received by means of SDP, i.e., if the media are under aggregate
control (see Appendix D.3) or if aggregate control is not available
(see Appendix D.2).
It is RECOMMENDED that grouped medias are handled by aggregate
control, to give the client the ability to control either the whole
presentation or single medias.
D.5. RTSP external SDP delivery
There are some considerations that need to be made when the session
description is delivered to the client outside of RTSP, for example
via HTTP or email.
First of all, the SDP needs to contain absolute URIs, since relative
will in most cases not work as the delivery will not correctly
forward the base URI.
The writing of the SDP session availability information, i.e. "t="
and "r=", needs to be carefully considered. When the SDP is fetched
by the DESCRIBE method, the probability that it is valid is very
high. However, the same is much less certain for SDPs distributed
using other methods. Therefore the publisher of the SDP should take
care to follow the recommendations about availability in the SDP
specification [RFC4566] in Section 4.2.
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Appendix E. RTSP Use Cases
This Appendix describes the most important and considered use cases
for RTSP. They are listed in descending order of importance in
regards to ensuring that all necessary functionality is present.
This specification only fully supports usage of the two first. Also
in these first two cases, there are special cases or exceptions that
are not supported without extensions, e.g. the redirection of media
delivery to another address than the controlling agent's (client's).
E.1. On-demand Playback of Stored Content
An RTSP capable server stores content suitable for being streamed to
a client. A client desiring playback of any of the stored content
uses RTSP to set up the media transport required to deliver the
desired content. RTSP is then used to initiate, halt and manipulate
the actual transmission (playout) of the content. RTSP is also
required to provide necessary description and synchronization
information for the content.
The above high level description can be broken down into a number of
functions that RTSP needs to be capable of.
Presentation Description: Provide initialization information about
the presentation (content); for example, which media codecs are
needed for the content. Other information that is important
includes the number of media streams the presentation contains,
the transport protocols used for the media streams, and
identifiers for these media streams. This information is
required before setup of the content is possible and to
determine if the client is even capable of using the content.
This information need not be sent using RTSP; other external
protocols can be used to transmit the transport presentation
descriptions. Two good examples are the use of HTTP [RFC2616]
or email to fetch or receive presentation descriptions like SDP
[RFC4566]
Setup: Set up some or all of the media streams in a presentation.
The setup itself consists of selecting the protocol for media
transport and the necessary parameters for the protocol, like
addresses and ports.
Control of Transmission: After the necessary media streams have been
established the client can request the server to start
transmitting the content. The client must be allowed to start
or stop the transmission of the content at arbitrary times.
The client must also be able to start the transmission at any
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point in the timeline of the presentation.
Synchronization: For media transport protocols like RTP [RFC3550] it
might be beneficial to carry synchronization information within
RTSP. This may be due to either the lack of inter-media
synchronization within the protocol itself, or the potential
delay before the synchronization is established (which is the
case for RTP when using RTCP).
Termination: Terminate the established contexts.
For this use case there are a number of assumptions about how it
works. These are:
On-Demand content: The content is stored at the server and can be
accessed at any time during a time period when it is intended
to be available.
Independent sessions: A server is capable of serving a number of
clients simultaneously, including from the same piece of
content at different points in that presentations time-line.
Unicast Transport: Content for each individual client is transmitted
to them using unicast traffic.
It is also possible to redirect the media traffic to a different
destination than that of the agent controlling the traffic. However,
allowing this without appropriate mechanisms for checking that the
destination approves of this allows for distributed denial of service
attacks (DDoS).
E.2. Unicast Distribution of Live Content
This use case is similar to the above on-demand content case (see
Appendix E.1) the difference is the nature of the content itself.
Live content is continuously distributed as it becomes available from
a source; i.e., the main difference from on-demand is that one starts
distributing content before the end of it has become available to the
server.
In many cases the consumer of live content is only interested in
consuming what actually happens "now"; i.e., very similar to
broadcast TV. However, in this case it is assumed that there exists
no broadcast or multicast channel to the users, and instead the
server functions as a distribution node, sending the same content to
multiple receivers, using unicast traffic between server and client.
This unicast traffic and the transport parameters are individually
negotiated for each receiving client.
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Another aspect of live content is that it often has a very limited
time of availability, as it is only available for the duration of the
event the content covers. An example of such a live content could be
a music concert which lasts 2 hour and starts at a predetermined
time. Thus there is a need to announce when and for how long the
live content is available.
In some cases, the server providing live content may be saving some
or all of the content to allow clients to pause the stream and resume
it from the paused point, or to "rewind" and play continuously from a
point earlier than the live point. Hence, this use case does not
necessarily exclude playing from other than the live point of the
stream, playing with scales other than 1.0, etc.
E.3. On-demand Playback using Multicast
It is possible to use RTSP to request that media be delivered to a
multicast group. The entity setting up the session (the controller)
will then control when and what media is delivered to the group.
This use case has some potential for denial of service attacks by
flooding a multicast group. Therefore, a mechanism is needed to
indicate that the group actually accepts the traffic from the RTSP
server.
An open issue in this use case is how one ensures that all receivers
listening to the multicast or broadcast receives the session
presentation configuring the receivers. This specification has to
rely on an external solution to solve this issue.
E.4. Inviting an RTSP server into a conference
If one has an established conference or group session, it is possible
to have an RTSP server distribute media to the whole group.
Transmission to the group is simplest when controlled by a single
participant or leader of the conference. Shared control might be
possible, but would require further investigation and possibly
extensions.
This use case assumes that there exists either multicast or a
conference focus that redistribute media to all participants.
This use case is intended to be able to handle the following
scenario: A conference leader or participant (hereafter called the
controller) has some pre-stored content on an RTSP server that he
wants to share with the group. The controller sets up an RTSP
session at the streaming server for this content and retrieves the
session description for the content. The destination for the media
content is set to the shared multicast group or conference focus.
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When desired by the controller, he/she can start and stop the
transmission of the media to the conference group.
There are several issues with this use case that are not solved by
this core specification for RTSP:
Denial of service: To avoid an RTSP server from being an unknowing
participant in a denial of service attack the server needs to
be able to verify the destination's acceptance of the media.
Such a mechanism to verify the approval of received media does
not yet exist; instead, only policies can be used, which can be
made to work in controlled environments.
Distributing the presentation description to all participants in the
group: To enable a media receiver to correctly decode the content
the media configuration information needs to be distributed
reliably to all participants. This will most likely require
support from an external protocol.
Passing control of the session: If it is desired to pass control of
the RTSP session between the participants, some support will be
required by an external protocol to exchange state information
and possibly floor control of who is controlling the RTSP
session.
E.5. Live Content using Multicast
This use case in its simplest form does not require any use of RTSP
at all; this is what multicast conferences being announced with SAP
[RFC2974] and SDP are intended to handle. However, in use cases
where more advanced features like access control to the multicast
session are desired, RTSP could be used for session establishment.
A client desiring to join a live multicasted media session with
cryptographic (encryption) access control could use RTSP in the
following way. The source of the session announces the session and
gives all interested an RTSP URI. The client connects to the server
and requests the presentation description, allowing configuration for
reception of the media. In this step it is possible for the client
to use secured transport and any desired level of authentication; for
example, for billing or access control. An RTSP link also allows for
load balancing between multiple servers.
If these were the only goals, they could be achieved by simply using
HTTP. However, for cases where the sender likes to keep track of
each individual receiver of a session, and possibly use the session
as a side channel for distributing key-updates or other information
on a per-receiver basis, and the full set of receivers is not known
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prior to the session start, the state establishment that RTSP
provides can be beneficial. In this case a client would establish an
RTSP session for this multicast group with the RTSP server. The RTSP
server will not transmit any media, but instead will point to the
multicast group. The client and server will be able to keep the
session alive for as long as the receiver participates in the session
thus enabling, for example, the server to push updates to the client.
This use case will most likely not be able to be implemented without
some extensions to the server-to-client push mechanism. Here the
PLAY_NOTIFY method (see Section 13.5) with a suitable extension could
provide clear benefits.
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Appendix F. Text format for Parameters
A resource of type "text/parameters" consists of either 1) a list of
parameters (for a query) or 2) a list of parameters and associated
values (for an response or setting of the parameter). Each entry of
the list is a single line of text. Parameters are separated from
values by a colon. The parameter name MUST only use US-ASCII visible
characters while the values are UTF-8 text strings. The media type
registration form is in Section 22.16.
There is a potential interoperability issue for this format. It was
named in RFC 2326 but never defined, even if used in examples that
hint at the syntax. This format matches the purpose and its syntax
supports the examples provided. However, it goes further by allowing
UTF-8 in the value part, thus usage of UTF-8 strings may not be
supported. However, as individual parameters are not defined, the
using application anyway needs to have out-of-band agreement or using
feature-tag to determine if the end-point supports the parameters.
The ABNF [RFC5234] grammar for "text/parameters" content is:
file = *((parameter / parameter-value) CRLF)
parameter = 1*visible-except-colon
parameter-value = parameter *WSP ":" value
visible-except-colon = %x21-39 / %x3B-7E ; VCHAR - ":"
value = *(TEXT-UTF8char / WSP)
TEXT-UTF8char = %x21-7E / UTF8-NONASCII
UTF8-NONASCII = %xC0-DF 1UTF8-CONT
/ %xE0-EF 2UTF8-CONT
/ %xF0-F7 3UTF8-CONT
/ %xF8-FB 4UTF8-CONT
/ %xFC-FD 5UTF8-CONT
UTF8-CONT = %x80-BF
WSP = <See RFC 5234> ; Space or HTAB
VCHAR = <See RFC 5234>
CRLF = <See RFC 5234>
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Appendix G. Requirements for Unreliable Transport of RTSP
This section provides anyone intending to define how to transport of
RTSP messages over a unreliable transport protocol with some
information learned by the attempt in RFC 2326 [RFC2326]. RFC 2326
defined both an URI scheme and some basic functionality for transport
of RTSP messages over UDP, however, it was not sufficient for
reliable usage and successful interoperability.
The RTSP scheme defined for unreliable transport of RTSP messages was
"rtspu". It has been reserved by this specification as at least one
commercial implementation exists, thus avoiding any collisions in the
name space.
The following considerations should exist for operation of RTSP over
an unreliable transport protocol:
o Request shall be acknowledged by the receiver. If there is no
acknowledgement, the sender may resend the same message after a
timeout of one round-trip time (RTT). Any retransmissions due to
lack of acknowledgement must carry the same sequence number as the
original request.
o The round-trip time can be estimated as in TCP (RFC 6298)
[RFC6298], with an initial round-trip value of 500 ms. An
implementation may cache the last RTT measurement as the initial
value for future connections.
o The Timestamp header (Section 18.51) is used to avoid the
retransmission ambiguity problem [Stevens98].
o The registered default port for RTSP over UDP for the server is
554.
o RTSP messages can be carried over any lower-layer transport
protocol that is 8-bit clean.
o RTSP messages are vulnerable to bit errors and should not be
subjected to them.
o Source authentication, or at least validation that RTSP messages
comes from the same entity becomes extremely important, as session
hijacking may be substantially easier for RTSP message transport
using an unreliable protocol like UDP than for TCP.
There are two RTSP headers that are primarily intended for being used
by the unreliable handling of RTSP messages and which will be
maintained:
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o CSeq: See Section 18.19
o Timestamp: See Section 18.51
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Appendix H. Backwards Compatibility Considerations
This section contains notes on issues about backwards compatibility
with clients or servers being implemented according to RFC 2326
[RFC2326]. Note that there exists no requirement to implement RTSP
1.0; in fact we recommend against it as it is difficult to do in an
interoperable way.
A server implementing RTSP/2.0 MUST include an RTSP-Version of
RTSP/2.0 in all responses to requests containing RTSP-Version
RTSP/2.0. If a server receives an RTSP/1.0 request, it MAY respond
with an RTSP/1.0 response if it chooses to support RFC 2326. If the
server chooses not to support RFC 2326, it MUST respond with a 505
(RTSP Version not supported) status code. A server MUST NOT respond
to an RTSP-Version RTSP/1.0 request with an RTSP-Version RTSP/2.0
response.
Clients implementing RTSP/2.0 MAY use an OPTIONS request with a RTSP-
Version of 2.0 to determine whether a server supports RTSP/2.0. If
the server responds with either an RTSP-Version of 1.0 or a status
code of 505 (RTSP Version not supported), the client will have to use
RTSP/1.0 requests if it chooses to support RFC 2326.
H.1. Play Request in Play State
The behavior in the server when a Play is received in Play state has
changed (Section 13.4). In RFC 2326, the new PLAY request would be
queued until the current Play completed. Any new PLAY request now
takes effect immediately replacing the previous request.
H.2. Using Persistent Connections
Some server implementations of RFC 2326 maintain a one-to-one
relationship between a connection and an RTSP session. Such
implementations require clients to use a persistent connection to
communicate with the server and when a client closes its connection,
the server may remove the RTSP session. This is worth noting if a
RTSP 2.0 client also supporting 1.0 connects to a 1.0 server.
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Appendix I. Changes
This appendix briefly lists the differences between RTSP 1.0
[RFC2326] and RTSP 2.0 for an informational purpose. For
implementers of RTSP 2.0 it is recommended to read carefully through
this memo and not to rely on the list of changes below to adapt from
RTSP 1.0 to RTSP 2.0, as RTSP 2.0 is not intended to be backwards
compatible with RTSP 1.0 [RFC2326] other than the version negotiation
mechanism.
I.1. Brief Overview
The following protocol elements were removed in RTSP 2.0 compared to
RTSP 1.0:
o there is no section on minimal implementation anymore, but more
the definition of RTSP 2.0 core;
o the RECORD and ANNOUNCE methods and all related functionality
(including 201 (Created) and 250 (Low On Storage Space) status
codes);
o the use of UDP for RTSP message transport was removed due to
missing interest and to broken specification;
o the use of PLAY method for keep-alive in Play state.
The following protocol elements were added or changed in RTSP 2.0
compared to RTSP 1.0:
o RTSP session TEARDOWN from the server to the client;
o IPv6 support;
o extended IANA registries (e.g., transport headers parameters,
transport-protocol, profile, lower-transport, and mode);
o request pipelining for quick session start-up;
o fully reworked state-machine;
o RTSP messages now use URIs rather then URLs;
o incorporated much of related HTTP text ([RFC2616]) in this memo,
compared to just referencing the sections in HTTP, to avoid
ambiguities;
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o the REDIRECT method was expanded and diversified for different
situations;
o Includes a new section about how to setup different media
transport alternatives and their profiles, and lower layer
protocols. This caused the appendix on RTP interaction to be
moved there instead of being in the part which describes RTP. The
section also includes guidelines what to consider when writing
usage guidelines for new protocols and profiles;
o Added an asynchronous notification method PLAY_NOTIFY. This
method is used by the RTSP server to asynchronously notify clients
about session changes while in Play state. To a limited extent
this is comparable with some implementations of ANNOUNCE in RTSP
1.0 not intended for Recording.
I.2. Detailed List of Changes
Compared to RTSP 1.0 (RFC 2326), the below changes has been made when
defining RTSP 2.0. Note that this list does not reflect minor
changes in wording or correction of typographical errors.
o The section on minimal implementation was deleted without
substitution.
o The Transport header has been changed in the following way:
* The ABNF has been changed to define that extensions are
possible, and that unknown parameters result in that servers
ignore the transport specification.
* To prevent backwards compatibility issues, any extension or new
parameter requires the usage of a feature-tag combined with the
Require header.
* Syntax unclarities with the Mode parameter have been resolved.
* Syntax error with ";" for multicast and unicast has been
resolved.
* Two new addressing parameters have been defined, src_addr and
dest_addr. These replace the parameters "port", "client_port",
"server_port", "destination", "source".
* Support for IPv6 explicit addresses in all address fields has
been included.
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* To handle URI definitions that contain ";" or "," a quoted URI
format has been introduced and is required.
* Defined IANA registries for the transport headers parameters,
transport-protocol, profile, lower-transport, and mode.
* The transport headers interleaved parameter's text was made
more strict and uses formal requirements levels. It was also
clarified that the interleaved channels are symmetric and that
it is the server that sets the channel numbers.
* It has been clarified that the client can't request of the
server to use a certain RTP SSRC, using a request with the
transport parameter SSRC.
* Syntax definition for SSRC has been clarified to require 8HEX.
It has also been extended to allow multiple values for clients
supporting this version.
* Clarified the text on the transport headers "dest_addr"
parameters regarding what security precautions the server is
required to perform.
o The Range formats has been changed in the following way:
* The NPT format has been given an initial NPT identifier that
must now be used.
* All formats now support initial open ended formats of type
"npt=-10" and also format only "Range: smpte" ranges for usage
with GET_PARAMETER requests.
o RTSP message handling has been changed in the following way:
* RTSP messages now use URIs rather then URLs.
* It has been clarified that a 4xx message due to missing CSeq
header shall be returned without a CSeq header.
* The 300 (Multiple Choices) response code has been removed.
* Rules for how to handle timing out RTSP messages has been
added.
* Extended Pipelining rules allowing for quick session startup.
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o The HTTP references have been updated to RFC 2616 and RFC 2617.
Most of the text has been copied and then altered to fit RTSP into
this specification. Public, and the Content-Base header has also
been imported from RFC 2068 so that they are defined in the RTSP
specification. Known effects on RTSP due to HTTP clarifications:
* Content-Encoding header can include encoding of type
"identity".
o The state machine section has been completely rewritten. It now
includes more details and is also more clear about the model used.
o An IANA section has been included which contains a number of
registries and their rules. This will allow us to use IANA to
keep track of RTSP extensions.
o The transport of RTSP messages has seen the following changes:
* The use of UDP for RTSP message transport has been deprecated
due to missing interest and to broken specification.
* The rules for how TCP connections are to be handled has been
clarified. Now it is made clear that servers should not close
the TCP connection unless they have been unused for significant
time.
* Strong recommendations why server and clients should use
persistent connections have also been added.
* There is now a requirement on the servers to handle non-
persistent connections as this provides fault tolerance.
* Added wording on the usage of Connection:Close for RTSP.
* Specified usage of TLS for RTSP messages, including a scheme to
approve a proxy's TLS connection to the next hop.
o The following header related changes have been made:
* Accept-Ranges response header is added. This header clarifies
which range formats that can be used for a resource.
* Fixed the missing definitions for the Cache-Control header.
Also added to the syntax definition the missing delta-seconds
for max-stale and min-fresh parameters.
* Put requirement on CSeq header that the value is increased by
one for each new RTSP request. A Recommendation to start at 0
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has also been added.
* Added requirement that the Date header must be used for all
messages with message body and the Server should always include
it.
* Removed possibility of using Range header with Scale header to
indicate when it is to be activated, since it can't work as
defined. Also added rule that lack of Scale header in response
indicates lack of support for the header. Feature-tags for
scaled playback has been defined.
* The Speed header must now be responded to indicate support and
the actual speed going to be used. A feature-tag is defined.
Notes on congestion control were also added.
* The Supported header was borrowed from SIP [RFC3261] to help
with the feature negotiation in RTSP.
* Clarified that the Timestamp header can be used to resolve
retransmission ambiguities.
* The Session header text has been expanded with an explanation
on keep alive and which methods to use. SET_PARAMETER is now
recommended to use if only keep-alive within RTSP is desired.
* It has been clarified how the Range header formats are used to
indicate pause points in the PAUSE response.
* Clarified that RTP-Info URIs that are relative, use the
Request-URI as base URI. Also clarified that the used URI must
be the one that was used in the SETUP request. The URIs are
now also required to be quoted. The header also expresses the
SSRC for the provided RTP timestamp and sequence number values.
* Added text that requires the Range to always be present in PLAY
responses. Clarified what should be sent in case of live
streams.
* The headers table has been updated using a structure borrowed
from SIP. Those tables convey much more information and should
provide a good overview of the available headers.
* It has been clarified that any message with a message body is
required to have a Content-Length header. This was the case in
RFC 2326, but could be misinterpreted.
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* ETag has changed name to MTag.
* To resolve functionality around MTag. The MTag and If-None-
Match header have been added from HTTP with necessary
clarification in regards to RTSP operation.
* Imported the Public header from HTTP RFC 2068 [RFC2068] since
it has been removed from HTTP due to lack of use. Public is
used quite frequently in RTSP.
* Clarified rules for populating the Public header so that it is
an intersection of the capabilities of all the RTSP agents in a
chain.
* Added the Media-Range header for listing the current
availability of the media range.
* Added the Notify-Reason header for giving the reason when
sending PLAY_NOTIFY requests.
* A new header Seek-Style has been defined to direct and inform
how any seek operation should/have been performed.
o The Protocol Syntax has been changed in the following way:
* All ABNF definitions are updated according to the rules defined
in RFC 5234 [RFC5234] and have been gathered in a separate
Section 20.
* The ABNF for the User-Agent and Server headers have been
corrected.
* Some definitions in the introduction regarding the RTSP session
have been changed.
* The protocol has been made fully IPv6 capable.
* The CHAR rule has been changed to exclude NULL.
o The Status codes have been changed in the following way:
* The use of status code 303 "See Other" has been deprecated as
it does not make sense to use in RTSP.
* When sending response 451 and 458 the response body should
contain the offending parameters.
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* Clarification on when a 3rr redirect status code can be
received has been added. This includes receiving 3rr as a
result of a request within a established session. This
provides clarification to a previous unspecified behavior.
* Removed the 201 (Created) and 250 (Low On Storage Space) status
codes as they are only relevant to recording, which is
deprecated.
* Several new Status codes have been defined: 464 "Data Transport
Not Ready Yet", 465 "Notification Reason Unknown", 470
"Connection Authorization Required", 471 "Connection
Credentials not accepted", 472 "Failure to establish secure
connection".
o The following functionality has been deprecated from the protocol:
* The use of Queued Play.
* The use of PLAY method for keep-alive in Play state.
* The RECORD and ANNOUNCE methods and all related functionality.
Some of the syntax has been removed.
* The possibility to use timed execution of methods with the time
parameter in the Range header.
* The description on how rtspu works is not part of the core
specification and will require external description. Only that
it exists is defined here and some requirements for the
transport is provided.
o The following changes have been made in relation to methods:
* The OPTIONS method has been clarified with regards to the use
of the Public and Allow headers.
* Added text clarifying the usage of SET_PARAMETER for keep-alive
and usage without any body.
* PLAY method is now allowed to be pipelined with the pipelining
of one or more SETUP requests following the initial that
generates the session for aggregated control.
* REDIRECT has been expanded and diversified for different
situations.
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* Added a new method PLAY_NOTIFY. This method is used by the
RTSP server to asynchronously notify clients about session
changes.
o Wrote a new section about how to setup different media transport
alternatives and their profiles, and lower layer protocols. This
caused the appendix on RTP interaction to be moved there instead
of being in the part which describes RTP. The section also
includes guidelines what to consider when writing usage guidelines
for new protocols and profiles.
o Setup and usage of independent TCP connections for transport of
RTP has been specified.
o Added a new section describing the available mechanisms to
determine if functionality is supported, called "Capability
Handling". Renamed option-tags to feature-tags.
o Added a contributors section with people who have contributed
actual text to the specification.
o Added a section Use Cases that describes the major use cases for
RTSP.
o Clarified the usage of a=range and how to indicate live content
that are not seekable with this header.
o Text specifying the special behavior of PLAY for live content.
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Appendix J. Acknowledgements
This memorandum defines RTSP version 2.0 which is a revision of the
Proposed Standard RTSP version 1.0 which is defined in [RFC2326].
The authors of RFC 2326 are Henning Schulzrinne, Anup Rao, and Robert
Lanphier.
Both RTSP version 1.0 and RTSP version 2.0 borrow format and
descriptions from HTTP/1.1.
This document has benefited greatly from the comments of all those
participating in the MMUSIC-WG. In addition to those already
mentioned, the following individuals have contributed to this
specification:
Rahul Agarwal, Jeff Ayars, Milko Boic, Torsten Braun, Brent Browning,
Bruce Butterfield, Steve Casner, Francisco Cortes, Kelly Djahandari,
Martin Dunsmuir, Eric Fleischman, Jay Geagan, Andy Grignon, V.
Guruprasad, Peter Haight, Mark Handley, Brad Hefta-Gaub, Volker Hilt,
John K. Ho, Go Hori, Philipp Hoschka, Anne Jones, Ingemar Johansson,
Anders Klemets, Ruth Lang, Stephanie Leif, Jonathan Lennox, Eduardo
F. Llach, Thomas Marshall, Rob McCool, David Oran, Joerg Ott, Maria
Papadopouli, Sujal Patel, Ema Patki, Alagu Periyannan, Colin Perkins,
Igor Plotnikov, Jonathan Sergent, Pinaki Shah, David Singer, Lior
Sion, Jeff Smith, Alexander Sokolsky, Dale Stammen, John Francis
Stracke, Maureen Chesire, David Walker, Geetha Srikantan, Stephan
Wenger, Pekka Pessi, Jae-Hwan Kim, Holger Schmidt, Stephen Farrell,
Xavier Marjou, Joe Pallas, Martti Mela, Byungjo Yoon and Patrick
Hoffman, Jinhang Choi, Ross Finlayson, Dale R. Worley, and especially
to Flemming Andreasen.
J.1. Contributors
The following people have made written contributions that were
included in the specification:
o Tom Marshall contributed text on the usage of 3rr status codes.
o Thomas Zheng contributed text on the usage of the Range in PLAY
responses and proposed an earlier version of the PLAY_NOTIFY
method.
o Sean Sheedy contributed text on the timeout behavior of RTSP
messages and connections, the 463 status code, and proposed an
earlier version of the PLAY_NOTIFY method.
o Greg Sherwood proposed an earlier version of the PLAY_NOTIFY
method.
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o Fredrik Lindholm contributed text about the RTSP security
framework.
o John Lazzaro contributed the text for RTP over Independent TCP.
o Aravind Narasimhan contributed by rewriting Media Transport
Alternatives (Appendix C) and editorial improvements on a number
of places in the specification.
o Torbjorn Einarsson has done some editorial improvements of the
text.
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Appendix K. RFC Editor Consideration
Please replace RFC XXXX with the RFC number this specification
receives.
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Authors' Addresses
Henning Schulzrinne
Columbia University
1214 Amsterdam Avenue
New York, NY 10027
USA
Email: schulzrinne@cs.columbia.edu
Anup Rao
Cisco
USA
Email: anrao@cisco.com
Rob Lanphier
Seattle, WA
USA
Email: robla@robla.net
Magnus Westerlund
Ericsson AB
Faeroegatan 6
STOCKHOLM, SE-164 80
SWEDEN
Email: magnus.westerlund@ericsson.com
Martin Stiemerling
NEC Laboratories Europe, NEC Europe Ltd.
Kurfuersten-Anlage 36
Heidelberg 69115
Germany
Phone: +49 (0) 6221 4342 113
Email: martin.stiemerling@neclab.eu
URI: http://ietf.stiemerling.org
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