Javascript Session Establishment Protocol
draft-ietf-rtcweb-jsep-02
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 8829.
|
|
|---|---|---|---|
| Authors | Justin Uberti , Cullen Fluffy Jennings | ||
| Last updated | 2012-10-22 | ||
| Replaces | draft-uberti-rtcweb-jsep | ||
| RFC stream | Internet Engineering Task Force (IETF) | ||
| Formats | |||
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| Additional resources | Mailing list discussion | ||
| Stream | WG state | WG Document | |
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| IESG | IESG state | Became RFC 8829 (Proposed Standard) | |
| Consensus boilerplate | Unknown | ||
| Telechat date | (None) | ||
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| Send notices to | (None) |
draft-ietf-rtcweb-jsep-02
Network Working Group J. Uberti
Internet-Draft Google
Intended status: Standards Track C. Jennings
Expires: April 25, 2013 Cisco
October 22, 2012
Javascript Session Establishment Protocol
draft-ietf-rtcweb-jsep-02
Abstract
This document proposes a mechanism for allowing a Javascript
application to fully control the signaling plane of a multimedia
session, and discusses how this would work with existing signaling
protocols.
Status of this Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
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Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on April 25, 2013.
Copyright Notice
Copyright (c) 2012 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
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described in the Simplified BSD License.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Other Approaches Considered . . . . . . . . . . . . . . . . . 5
3. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6
4. Semantics and Syntax . . . . . . . . . . . . . . . . . . . . . 7
4.1. Signaling Model . . . . . . . . . . . . . . . . . . . . . 7
4.2. Session Descriptions and State Machine . . . . . . . . . . 7
4.3. Session Description Format . . . . . . . . . . . . . . . . 9
4.4. ICE . . . . . . . . . . . . . . . . . . . . . . . . . . . 10
4.4.1. ICE Candidate Trickling . . . . . . . . . . . . . . . 10
4.4.1.1. ICE Candidate Format . . . . . . . . . . . . . . . 10
4.5. Interactions With Forking . . . . . . . . . . . . . . . . 11
4.5.1. Sequential Forking . . . . . . . . . . . . . . . . . . 11
4.5.2. Parallel Forking . . . . . . . . . . . . . . . . . . . 12
4.6. Session Rehydration . . . . . . . . . . . . . . . . . . . 13
5. Interface . . . . . . . . . . . . . . . . . . . . . . . . . . 14
5.1. SDP Requirements . . . . . . . . . . . . . . . . . . . . . 14
5.2. Methods . . . . . . . . . . . . . . . . . . . . . . . . . 15
5.2.1. createOffer . . . . . . . . . . . . . . . . . . . . . 15
5.2.2. createAnswer . . . . . . . . . . . . . . . . . . . . . 15
5.2.3. SessionDescriptionType . . . . . . . . . . . . . . . . 16
5.2.3.1. Creating Answers . . . . . . . . . . . . . . . . . 17
5.2.4. setLocalDescription . . . . . . . . . . . . . . . . . 17
5.2.5. setRemoteDescription . . . . . . . . . . . . . . . . . 18
5.2.6. localDescription . . . . . . . . . . . . . . . . . . . 18
5.2.7. remoteDescription . . . . . . . . . . . . . . . . . . 18
5.2.8. updateIce . . . . . . . . . . . . . . . . . . . . . . 18
5.2.9. addIceCandidate . . . . . . . . . . . . . . . . . . . 19
6. Configurable SDP Parameters . . . . . . . . . . . . . . . . . 20
7. Security Considerations . . . . . . . . . . . . . . . . . . . 21
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 22
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 23
10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 24
10.1. Normative References . . . . . . . . . . . . . . . . . . . 24
10.2. Informative References . . . . . . . . . . . . . . . . . . 24
Appendix A. JSEP Implementation Examples . . . . . . . . . . . . 26
A.1. Example API Flows . . . . . . . . . . . . . . . . . . . . 26
A.1.1. Call using ROAP . . . . . . . . . . . . . . . . . . . 26
A.1.2. Call using XMPP . . . . . . . . . . . . . . . . . . . 27
A.1.3. Adding video to a call, using XMPP . . . . . . . . . . 28
A.1.4. Simultaneous add of video streams, using XMPP . . . . 28
A.1.5. Call using SIP . . . . . . . . . . . . . . . . . . . . 29
A.1.6. Handling early media (e.g. 1-800-GO FEDEX), using
SIP . . . . . . . . . . . . . . . . . . . . . . . . . 30
Appendix B. Change log . . . . . . . . . . . . . . . . . . . . . 32
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 33
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1. Introduction
The thinking behind WebRTC call setup has been to fully specify and
control the media plane, but to leave the signaling plane up to the
application as much as possible. The rationale is that different
applications may prefer to use different protocols, such as the
existing SIP or Jingle call signaling protocols, or something custom
to the particular application, perhaps for a novel use case. In this
approach, the key information that needs to be exchanged is the
multimedia session description, which specifies the necessary
transport and media configuration information necessary to establish
the media plane.
The browser environment also has its own challenges that cause
problems for an embedded signaling state machine. One of these is
that the user may reload the web page at any time. If this happens,
and the state machine is being run at a server, the server can simply
push the current state back down to the page and resume the call
where it left off.
This document describes the Javascript Session Establishment Protocol
(JSEP) that pulls the signaling state machine out of the browser and
into Javascript. This mechanism effectively removes the browser
almost completely from the core signaling flow; the only interface
needed is a way for the application to pass in the local and remote
session descriptions negotiated by whatever signaling mechanism is
used, and a way to interact with the ICE state machine.
JSEP's handling of session descriptions is simple and
straightforward. Whenever an offer/answer exchange is needed, the
initiating side creates an offer by calling a createOffer() API. The
application optionally modifies that offer, and then uses it to set
up its local config via the setLocalDescription() API. The offer is
then sent off to the remote side over its preferred signaling
mechanism (e.g., WebSockets); upon receipt of that offer, the remote
party installs it using the setRemoteDescription() API.
When the call is accepted, the callee uses the createAnswer() API to
generate an appropriate answer, applies it using
setLocalDescription(), and sends the answer back to the initiator
over the signaling channel. When the offerer gets that answer, it
installs it using setRemoteDescription(), and initial setup is
complete. This process can be repeated for additional offer/answer
exchanges.
Regarding ICE, JSEP decouples the ICE state machine from the overall
signaling state machine, as the ICE state machine must remain in the
browser, because only the browser has the necessary knowledge of
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candidates and other transport info. Performing this separation also
provides additional flexibility; in protocols that decouple session
descriptions from transport, such as Jingle, the transport
information can be sent separately; in protocols that don't, such as
SIP, the information can be used in the aggregated form. Sending
transport information separately can allow for faster ICE and DTLS
startup, since the necessary roundtrips can occur while waiting for
the remote side to accept the session.
The JSEP approach does come with a minor downside. As the
application now is responsible for driving the signaling state
machine, slightly more application code is necessary to perform call
setup; the application must call the right APIs at the right times,
and convert the session descriptions and ICE information into the
defined messages of its chosen signaling protocol, instead of simply
forwarding the messages emitted from the browser.
One way to mitigate this is to provide a Javascript library that
hides this complexity from the developer, which would implement the
state machine and serialization of the desired signaling protocol.
For example, this library could convert easily adapt the JSEP API
into the exact ROAP API [I-D.jennings-rtcweb-signaling], thereby
implementing the ROAP signaling protocol. Such a library could of
course also implement other popular signaling protocols, including
SIP or Jingle. In this fashion we can enable greater control for the
experienced developer without forcing any additional complexity on
the novice developer.
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2. Other Approaches Considered
Another approach that was considered for JSEP was to move the
mechanism for generating offers and answers out of the browser as
well. Instead of providing createOffer/createAnswer methods within
the browser, this approach would instead expose a getCapabilities API
which would provide the application with the information it needed in
order to generate its own session descriptions. This increases the
amount of work that the application needs to do; it needs to know how
to generate session descriptions from capabilities, and especially
how to generate the correct answer from an arbitrary offer and the
supported capabilities. While this could certainly be addressed by
using a library like the one mentioned above, it basically forces the
use of said library even for a simple example. Exposing createOffer/
createAnswer avoids that problem, but still allows applications to
generate their own offers/answers if they choose, using the
description generated by createOffer as an indication of the
browser's capabilities.
Note also that while JSEP transfers more control to Javascript, it is
not intended to be an example of a "low-level" API. The general
argument against a low-level API is that there are too many necessary
API points, and they can be called in any order, leading to something
that is hard to specify and test. In the approach proposed here,
control is performed via session descriptions; this requires only a
few APIs to handle these descriptions, and they are evaluated in a
specific fashion, which reduces the number of possible states and
interactions.
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3. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [RFC2119].
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4. Semantics and Syntax
4.1. Signaling Model
JSEP does not specify a particular signaling model or state machine,
other than the generic need to exchange RFC 3264 offers and answers
in order for both sides of the session to know how to conduct the
session. JSEP provides mechanisms to create offers and answers, as
well as to apply them to a session. However, the actual mechanism by
which these offers and answers are communicated to the remote side,
including addressing, retransmission, forking, and glare handling, is
left entirely up to the application.
+-----------+ +-----------+
| Web App |<--- App-Specific Signaling -->| Web App |
+-----------+ +-----------+
^ ^
| SDP | SDP
V V
+-----------+ +-----------+
| Browser |<----------- Media ------------>| Browser |
+-----------+ +-----------+
Figure 1: JSEP Signaling Model
4.2. Session Descriptions and State Machine
In order to establish the media plane, the user agent needs specific
parameters to indicate what to transmit to the remote side, as well
as how to handle the media that is received. These parameters are
determined by the exchange of session descriptions in offers and
answers, and there are certain details to this process that must be
handled in the JSEP APIs.
Whether a session description was sent or received affects the
meaning of that description. For example, the list of codecs sent to
a remote party indicates what the local side is willing to decode,
and what the remote party should send. Not all parameters follow
this rule; for example, the SRTP parameters [RFC4568] sent to a
remote party indicate what the local side will use to encrypt, and
thereby how the remote party should expect to receive.
In addition, various RFCs put different conditions on the format of
offers versus answers. For example, a offer may propose multiple
SRTP configurations, but an answer may only contain a single SRTP
configuration.
Lastly, while the exact media parameters are only known only after a
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offer and an answer have been exchanged, it is possible for the
offerer to receive media after they have sent an offer and before
they have received an answer. To properly process incoming media in
this case, the offerer's media handler must be aware of the details
of the offerer before the answer arrives.
Therefore, in order to handle session descriptions properly, the user
agent needs:
1. To know if a session description pertains to the local or remote
side.
2. To know if a session description is an offer or an answer.
3. To allow the offer to be specified independently of the answer.
JSEP addresses this by adding both a setLocalDescription and a
setRemoteDescription method and having session description objects
contain a type field indicating the type of session description being
supplied. This satisfies the requirements listed above for both the
offerer, who first calls setLocalDescription(sdp [offer]) and then
later setRemoteDescription(sdp [answer]), as well as for the
answerer, who first calls setRemoteDescription(sdp [offer]) and then
later setLocalDescription(sdp [answer]). While it could be possible
to implicitly determine the value of the offer/answer argument,
requiring it to be specified explicitly is more robust, allowing
invalid combinations (i.e. an answer before an offer) to generate an
appropriate error.
JSEP also allows for an answer to be treated as provisional by the
application. Provisional answers provide a way for an answerer to
communicate initial session parameters back to the offerer, in order
to allow the session to begin, while allowing a final answer to be
specified later. This concept of a final answer is important to the
offer/answer model; when such an answer is received, any extra
resources allocated by the caller can be released, now that the exact
session configuration is known. These "resources" can include things
like extra ICE components, TURN candidates, or video decoders.
Provisional answers, on the other hand, do no such deallocation
results; as a result, multiple dissimilar provisional answers can be
received and applied during call setup.
In [RFC3264], the constraints at the signaling level is that only one
offer can be outstanding for a given session but from the media stack
level, a new offer can be generated at any point. For example, when
using SIP for signaling, if one offer is sent, then cancelled using a
SIP CANCEL, another offer can be generated even though no answer was
received for the first offer. To support this, the JSEP media layer
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can provide an offer whenever the Javascript application needs one
for the signaling. The answerer can send back zero or more
provisional answers, and finally end the offer-answer exchange by
sending a final answer. The state machine for this is as follows:
+-----------+
| |
| |
| Stable |<---------------\
| | |
| | |
+-----------+ |
^ | |
| | OFFER |
ANSWER | | | ANSWER
| V |
+-----------+ +-----------+
| | | |
| | PRANSWER | |
| Offer |-------- >| Pranswer |
| | | |
| |----\ | |----\
+-----------+ | +-----------+ |
^ | ^ |
| | | |
\-----/ \-----/
OFFER PRANSWER
Figure 2: JSEP State Machine
Aside from these state transitions, there is no other difference
between the handling of provisional ("pranswer") and final ("answer")
answers.
4.3. Session Description Format
In the WebRTC specification, session descriptions are formatted as
SDP messages. While this format is not optimal for manipulation from
Javascript, it is widely accepted, and frequently updated with new
features. Any alternate encoding of session descriptions would have
to keep pace with the changes to SDP, at least until the time that
this new encoding eclipsed SDP in popularity. As a result, JSEP
continues to use SDP as the internal representation for its session
descriptions.
However, to simplify Javascript processing, and provide for future
flexibility, the SDP syntax is encapsulated within a
SessionDescription object, which can be constructed from SDP, and be
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serialized out to SDP. If future specifications agree on a JSON
format for session descriptions, we could easily enable this object
to generate and consume that JSON.
Other methods may be added to SessionDescription in the future to
simplify handling of SessionDescriptions from Javascript. Though it
is unclear exactly what manipulations developer will commonly want to
do to SDP, it would be simple to write a Javascript library to
perform these manipulations.
4.4. ICE
When a new ICE candidate is available, the ICE Agent will notify the
application via a callback; these candidates will automatically be
added to the local session description. When all candidates have
been gathered, the callback will also be invoked to signal that the
gathering process is complete.
4.4.1. ICE Candidate Trickling
Candidate trickling is a technique through which a caller may
incrementally provide candidates to the callee after the initial
offer has been dispatched; the semantics of "Trickle ICE" are defined
in [I-D.rescorla-mmusic-ice-trickle]. This process allows the callee
to begin acting upon the call and setting up the ICE (and perhaps
DTLS) connections immediately, without having to wait for the caller
to gather all possible candidates. This results in faster call
startup in cases where gathering is not performed prior to initating
the call.
JSEP supports optional candidate trickling by providing APIs that
provide control and feedback on the ICE candidate gathering process.
Applications that support candidate trickling can send the initial
offer immediately and send individual candidates when they get the
notified of a new candidate; applications that do not support this
feature can simply wait for the indication that gathering is
complete, and then create and send their offer, with all the
candidates, at this time.
Upon receipt of trickled candidates, the receiving application will
supply them to its ICE Agent. This triggers the ICE Agent to start
using the new remote candidates for connectivity checks.
4.4.1.1. ICE Candidate Format
As with session descriptions, the syntax of the IceCandidate object
provides some abstraction, but can be easily converted to and from
the SDP a=candidate lines.
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The a=candidate lines are the only SDP information that is contained
within IceCandidate, as they represent the only information needed
that is not present in the initial offer (i.e. for trickle
candidates). This information is carried with the same syntax as the
"a=candidate" line in SDP. For example:
a=candidate:1 1 UDP 1694498815 192.0.2.33 10000 typ host
The IceCandidate object also contains fields to indicate which m=
line it should be associated with. The m line can be identified in
one of two ways; either by a m-line index, or a MID. The m-line
index is a zero-based index, referring to the Nth m-line in the SDP.
The MID uses the "media stream identification", as defined in [RFC
3388], to identify the m-line. WebRTC implementations creating an
ICE Candidate object MUST populate both of these fields.
Implementations receiving an ICE Candidate object SHOULD use the MID
if they implement that functionality, or the m-line index, if not.
4.5. Interactions With Forking
Some call signaling systems allow various types of forking where an
SDP Offer may be provided to more than one device. For example, SIP
RFC 3261 defines both a "Parallel Search" and "Sequential Search".
Although these are primarily signaling level issues that are outside
the scope of JSEP, they do have some impact on the configuration of
the media plane, which is relevant. When forking is happening at the
signaling layer, the Javascript application responsible for the
signaling needs to make the decisions about what media should be sent
or received at any point of time and which remote endpoint it should
communicate with. JSEP is used to make sure the media engine can
make the RTP and media perform as required by the application. The
basic operations that the applications can have the media engine do
are:
Start exchanging media to a given remote peer but keep all the
resources reserved in the offer.
Start exchanging media with a given remote peer and free any
resources in the offer that are not being used.
4.5.1. Sequential Forking
Sequential forking involves a call being dispatched to multiple
remote callees, where each callee can accept the call, but only one
active session ever exists at a time; no mixing of received media is
performed.
JSEP handles serial forking well, allowing the application to easily
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control the policy for selecting the desired remote endpoint. When
an answer arrives from one of the callees, the application can choose
to apply it either as a provisional answer, leaving open the
possibility of using a different answer in the future, or apply it as
a final answer, ending the setup flow.
In a "first-one-wins" situation, the first answer will be applied as
a final answer, and the application will reject any subsequent
answers. In SIP parlance, this would be ACK + BYE.
In a "last-one-wins" situation, all answers would be applied as
provisional answers, and any previous call leg will be terminated.
At some point, the application will end the setup process, perhaps
with a timer; at this point, the application could reapply the
existing remote description as a final answer.
4.5.2. Parallel Forking
Parallel forking involves a call being dispatched to multiple remote
callees, where each callee can accept the call, and multiple
simultaneous active signaling sessions can be established as a
result. If multiple callees send media at the same time, the
possibilities for handling this are described in Section 3.1 of RFC
3960. Most SIP devices today only support exchanging media with a
single device at a time, and do not try to mix multiple early media
audio sources, as that could result in a confusing situation. For
example. consider having a European ringback tone mixed together with
the North American ringback tone - the resulting sound would not be
like either tone, and would confuse the user. If the signaling
application wishes to only exchange media with one of the remote
endpoints at a time, then from a media engine point of view, this is
exactly like the sequential forking case.
In the parallel forking case where the Javascript application wishes
to simultaneously exchange media with multiple peers, the flow is
slightly more complex, but the Javascript application can follow the
strategy that RFC 3960 describes using UPDATE. (It is worth noting
that use cases where this is the desired behavior are very unusual.)
The UPDATE approach allows the signaling to set up a separate media
flow for each peer that it wishes to exchange media with. In JSEP,
this offer used in the UPDATE would be formed by simply creating a
new PeerConnection and making sure that the same local media streams
have been added into this new PeerConnection. Then the new
PeerConnection object would produce a SDP offer that could be used by
the signaling to perform the UPDATE strategy discussed in RFC 3690.
As a result of sharing the media streams, the application will end up
with N parallel PeerConnection sessions, each with a local and remote
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description and their own local and remote addresses. The media flow
from these sessions can be managed by specifying SDP direction
attributes in the descriptions, or the application can choose to play
out the media from all sessions mixed together. Of course, if the
application wants to only keep a single session, it can simply
terminate the sessions that it no longer needs.
4.6. Session Rehydration
In the event that the local application state is reinitialized,
either due to a user reload of the page, or a decision within the
application to reload itself (perhaps to update to a new version), it
is possible to keep an existing session alive via a process called
"rehydration".
With rehydration, the current signaling state is persisted somewhere
outside of the page, perhaps on the application server, or in browser
local storage. The page is then reloaded, and a new session object
is created in Javascript. The saved signaling state is now
retrieved, and a new PeerConnection object is created for the
session. At this point a new offer can be generated by the new
PeerConnection, with new ICE and SDES credentials. This can then be
used to re-initiate the session with the existing remote endpoint,
who simply sees the new offer as an in-call renegotiation, and will
reply with an answer that can be supplied to setRemoteDescription.
ICE processing proceeds as usual, and as soon as connectivity is
established, the session will be back up and running again.
Open Issue: EKR proposed an alternative rehydration approach where
the actual internal PeerConnection object in the browser was kept
alive for some time after the web page was killed and provided some
way for a new page to acquire the old PeerConnection object.
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5. Interface
This section details the basic operations that must be present to
implement JSEP functionality. The actual API exposed in the W3C API
may have somewhat different syntax, but should map easily to these
concepts.
5.1. SDP Requirements
Note: The text in this section may not represent working group
consensus and is put here so that the working group can discuss it
and find out how to change it such that it does have consensus.
When generating SDP blobs, either for offers or answers, the
generated SDP needs to conform to the following specifications.
Similarly, in order to properly process received SDP blobs,
implementations need to implement the functionality described in the
following specifications. This list is derived from
[I-D.ietf-rtcweb-rtp-usage].
RFC4566 is the base SDP specification and MUST be implemented.
RFC5124 MUST be supported for signaling RTP/SAVPF RTP profile.
RFC5104 MUST be implemented to signal RTCP based feedback.
RFC5761 MUST be implemented to signal multiplexing of RTP and
RTCP.
RFC5245 MUST be implemented for signaling the ICE candidate lines
corresponding to each media stream.
RFC3264 MUST be implemented to signal information about media
direction.
The RFC5888 grouping framework MUST be implemented for signaling
the grouping information.
RFC5506 MAY be implemented to signal Reduced-Size RTCP messages.
RFC5576 MAY be implemented to signal RTP SSRC values.
RFC3556 with bandwidth modifiers MAY be supported for specifying
RTCP bandwidth as a fraction of the media bandwidth, RTCP fraction
allocated to the senders and setting maximum media bit-rate
boundaries.
As required by RFC 4566 Section 5.13 JSEP implementations MUST ignore
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unknown attributes (a=) lines.
Example SDP for RTCWeb call flows can be found in
[I-D.nandakumar-rtcweb-sdp].
5.2. Methods
5.2.1. createOffer
The createOffer method generates a blob of SDP that contains a RFC
3264 offer with the supported configurations for the session,
including descriptions of the local MediaStreams attached to this
PeerConnection, the codec/RTP/RTCP options supported by this
implementation, and any candidates that have been gathered by the ICE
Agent. A constraints parameters may be supplied to provide
additional control over the generated offer, e.g. to get a full set
of session capabilities, or to request a new set of ICE credentials.
In the initial offer, the generated SDP will contain all desired
functionality for the session (certain parts that are supported but
not desired by default may be omitted); for each SDP line, the
generation of the SDP must follow the appropriate process for
generating an offer. In the event createOffer is called after the
session is established, createOffer will generate an offer that is
compatible with the current session, incorporating any changes that
have been made to the session since the last complete offer-answer
exchange, such as addition or removal of streams. If no changes have
been made, the offer will be identical to the current local
description.
Session descriptions generated by createOffer must be immediately
usable by setLocalDescription; if a system has limited resources
(e.g. a finite number of decoders), createOffer should return an
offer that reflects the current state of the system, so that
setLocalDescription will succeed when it attempts to acquire those
resources. Because this method may need to inspect the system state
to determine the currently available resources, it may be implemented
as an async operation.
Calling this method may do things such as generate new ICE
credentials, but does not change media state.
5.2.2. createAnswer
The createAnswer method generates a blob of SDP that contains a RFC
3264 SDP answer with the supported configuration for the session that
is compatible with the parameters supplied in the offer. Like
createOffer, the returned blob contains descriptions of the local
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MediaStreams attached to this PeerConnection, the codec/RTP/RTCP
options negotiated for this session, and any candidates that have
been gathered by the ICE Agent. A constraints parameter may be
supplied to provide additional control over the generated answer.
As an answer, the generated SDP will contain a specific configuration
that specifies how the media plane should be established.
Session descriptions generated by createAnswer must be immediately
usable by setLocalDescription; like createOffer, the returned
description should reflect the current state of the system. Because
this method may need to inspect the system state to determine the
currently available resources, it may need to be implemented as an
async operation.
Calling this method may do things such as generate new ICE
credentials, but does not change media state.
5.2.3. SessionDescriptionType
Session description objects (RTCSessionDescription) may be of type
"offer", "pranswer", and "answer". These types provide information
as to how the description parameter should be parsed, and how the
media state should be changed.
"offer" indicates that a description should be parsed as an offer;
said description may include many possible media configurations. A
description used as an "offer" may be applied anytime the
PeerConnection is in a stable state, or as an update to a previously
sent but unanswered "offer".
"pranswer" indicates that a description should be parsed as an
answer, but not a final answer, and so should not result in the
freeing of allocated resources. It may result in the start of media
transmission, if the answer does not specify an inactive media
direction. A description used as a "pranswer" may be applied as a
response to an "offer", or an update to a previously sent "answer".
"answer" indicates that a description should be parsed as an answer,
the offer-answer exchange should be considered complete, and any
resources (decoders, candidates) that are no longer needed can be
released. A description used as an "answer" may be applied as a
response to a "offer", or an update to a previously sent "pranswer".
The application can use some discretion on whether an answer should
be applied as provisional or final. For example, in a serial forking
scenario, an application may receive multiple "final" answers, one
from each remote endpoint. The application could accept the initial
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answers as provisional answers, and only apply an answer as final
when it receives one that meets its criteria (e.g. a live user
instead of voicemail).
5.2.3.1. Creating Answers
Most web applications will not need to create answers using the
"pranswer" type. The general recommendation for a web application
would be to create an answer more or less immediately after receiving
the offer, instead of waiting for a human user to provide input.
Later when the human input is received, the applications can create a
new offer to update the previous offer/answer pair. Some
applications may not be able to do this, particularly ones that Some
application may not be able to do this, particular ones that are
attempting to gateway to other signaling protocols.
Consider a typical web application that will set up a data channel,
an audio channel, and a video channel. When an endpoint receives an
offer with these channels, it could send an answer accepting the data
channel for two-way data, and accepting the audio and video tracks as
receive-only. It could then ask the user if they wanted to transmit
audio and video to the far end, acquire the local media streams, and
send a new offer to the remote side moving the audio and video to be
two-way media. By the time the human has authorized sending media,
it is likely that the ICE and DTLS handshaking with the remote side
will already be set up.
5.2.4. setLocalDescription
The setLocalDescription method instructs the PeerConnection to apply
the supplied SDP blob as its local configuration. The type field
indicates whether the blob should be processed as an offer,
provisional answer, or final answer; offers and answers are checked
differently, using the various rules that exist for each SDP line.
This API changes the local media state; among other things, it sets
up local resources for receiving and decoding media. In order to
successfully handle scenarios where the application wants to offer to
change from one media format to a different, incompatible format, the
PeerConnection must be able to simultaneously support use of both the
old and new local descriptions (e.g. support codecs that exist in
both descriptions) until a final answer is received, at which point
the PeerConnection can fully adopt the new local description, or roll
back to the old description if the remote side denied the change.
If setRemoteDescription was previous called with an offer, and
setLocalDescription is called with an answer (provisional or final),
and the media directions are compatible, this will result in the
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starting of media transmission.
5.2.5. setRemoteDescription
The setRemoteDescription method instructs the PeerConnection to apply
the supplied SDP blob as the desired remote configuration. As in
setLocalDescription, the type field of the indicates how the blob
should be processed.
This API changes the local media state; among other things, it sets
up local resources for sending and encoding media.
If setRemoteDescription was previous called with an offer, and
setLocalDescription is called with an answer (provisional or final),
and the media directions are compatible, this will result in the
starting of media transmission.
5.2.6. localDescription
The localDescription method returns a copy of the current local
configuration, i.e. what was most recently passed to
setLocalDescription, plus any local candidates that have been
generated by the ICE Agent.
A null object will be returned if the local description has not yet
been established.
5.2.7. remoteDescription
The remoteDescription method returns a copy of the current remote
configuration, i.e. what was most recently passed to
setRemoteDescription, plus any remote candidates that have been
supplied via processIceMessage.
A null object will be returned if the remote description has not yet
been established.
5.2.8. updateIce
The updateIce method allows the configuration of the ICE Agent to be
changed during the session, primarily for changing which types of
local candidates are provided to the application and used for
connectivity checks. A callee may initially configure the ICE Agent
to use only relay candidates, to avoid leaking location information,
but update this configuration to use all candidates once the call is
accepted.
Regardless of the configuration, the gathering process collects all
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available candidates, but excluded candidates will not be surfaced in
onicecallback or used for connectivity checks.
This call may result in a change to the state of the ICE Agent, and
may result in a change to media state if it results in connectivity
being established.
5.2.9. addIceCandidate
The addIceCandidate method provides a remote candidate to the ICE
Agent, which will be added to the remote description. Connectivity
checks will be sent to the new candidate.
This call will result in a change to the state of the ICE Agent, and
may result in a change to media state if it results in connectivity
being established.
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6. Configurable SDP Parameters
Note: This section is still very early and is likely to
significantly change as we get a better understanding of the a) the
use cases for this b) the implications at the protocol level c)
feedback from implementors on what they can do.
The following is a partial list of SDP parameters that an application
may want to control, in either local or remote descriptions, using
this API.
o remove or reorder codecs (m=)
o change codec attributes (a=fmtp; ptime)
o enable/disable BUNDLE (a=group)
o enable/disable RTCP mux (a=rtcp-mux)
o change send resolution or framerate (TBD)
o change desired recv resolution or framerate (TBD)
o change total bandwidth (b=)
o remove desired AVPF mechanisms (a=rtcp-fb)
o remove RTP header extensions (a=rtphdr-ext)
o add/change SSRC grouping (e.g. FID, RTX, etc) (a=ssrc-group)
o add SSRC attributes (a=ssrc)
o change media send/recv state (a=sendonly/recvonly/inactive)
For example, an application could implement call hold by adding an
a=inactive attribute to its local description, and then applying and
signaling that description.
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7. Security Considerations
TODO
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8. IANA Considerations
This document requires no actions from IANA.
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9. Acknowledgements
Harald Alvestrand, Dan Burnett, Neil Stratford, Eric Rescorla, Anant
Narayanan, and Adam Bergkvist all provided valuable feedback on this
proposal. Suhas Nandakumar provided text and input for SDP
requirements. Matthew Kaufman provided the observation that keeping
state out of the browser allows a call to continue even if the page
is reloaded.
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10. References
10.1. Normative References
[I-D.rescorla-mmusic-ice-trickle]
Rescorla, E., Uberti, J., and E. Ivov, "Trickle ICE:
Incremental Provisioning of Candidates for the Interactive
Connectivity Establishment (ICE) Protocol",
draft-rescorla-mmusic-ice-trickle-00 (work in progress),
October 2012.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264,
June 2002.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
10.2. Informative References
[I-D.ietf-rtcweb-rtp-usage]
Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
Communication (WebRTC): Media Transport and Use of RTP",
draft-ietf-rtcweb-rtp-usage-04 (work in progress),
July 2012.
[I-D.jennings-rtcweb-signaling]
Jennings, C., Rosenberg, J., and R. Jesup, "RTCWeb Offer/
Answer Protocol (ROAP)",
draft-jennings-rtcweb-signaling-01 (work in progress),
October 2011.
[I-D.nandakumar-rtcweb-sdp]
Nandakumar, S. and C. Jennings, "SDP for the WebRTC",
draft-nandakumar-rtcweb-sdp-00 (work in progress),
October 2012.
[RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session
Description Protocol (SDP) Security Descriptions for Media
Streams", RFC 4568, July 2006.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245,
April 2010.
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[W3C.WD-webrtc-20111027]
Bergkvist, A., Burnett, D., Narayanan, A., and C.
Jennings, "WebRTC 1.0: Real-time Communication Between
Browsers", World Wide Web Consortium WD WD-webrtc-
20111027, October 2011,
<http://www.w3.org/TR/2011/WD-webrtc-20111027>.
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Appendix A. JSEP Implementation Examples
A.1. Example API Flows
Below are several sample flows for the new PeerConnection and library
APIs, demonstrating when the various APIs are called in different
situations and with various transport protocols. For clarity and
simplicity, the createOffer/createAnswer calls are assumed to be
synchronous in these examples, whereas the actual APIs are async.
A.1.1. Call using ROAP
This example demonstrates a ROAP call, without the use of trickle
candidates.
// Call is initiated toward Answerer
OffererJS->OffererUA: pc = new PeerConnection();
OffererJS->OffererUA: pc.addStream(localStream, null);
OffererUA->OffererJS: iceCallback(candidate);
OffererJS->OffererUA: offer = pc.createOffer(null);
OffererJS->OffererUA: pc.setLocalDescription("offer", offer);
OffererJS->AnswererJS: {"type":"OFFER", "sdp":offer }
// OFFER arrives at Answerer
AnswererJS->AnswererUA: pc = new PeerConnection();
AnswererJS->AnswererUA: pc.setRemoteDescription("offer", msg.sdp);
AnswererUA->AnswererJS: onaddstream(remoteStream);
AnswererUA->OffererUA: iceCallback(candidate);
// Answerer accepts call
AnswererJS->AnswererUA: peer.addStream(localStream, null);
AnswererJS->AnswererUA: answer = peer.createAnswer(msg.sdp, null);
AnswererJS->AnswererUA: peer.setLocalDescription("answer", answer);
AnswererJS->OffererJS: {"type":"ANSWER","sdp":answer }
// ANSWER arrives at Offerer
OffererJS->OffererUA: peer.setRemoteDescription("answer", answer);
OffererUA->OffererJS: onaddstream(remoteStream);
// ICE Completes (at Answerer)
AnswererUA->AnswererJS: onopen();
AnswererUA->OffererUA: Media
// ICE Completes (at Offerer)
OffererUA->OffererJS: onopen();
OffererJS->AnswererJS: {"type":"OK" }
OffererUA->AnswererUA: Media
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A.1.2. Call using XMPP
This example demonstrates an XMPP call, making use of trickle
candidates.
// Call is initiated toward Answerer
OffererJS->OffererUA: pc = new PeerConnection();
OffererJS->OffererUA: pc.addStream(localStream, null);
OffererJS->OffererUA: offer = pc.createOffer(null);
OffererJS->OffererUA: pc.setLocalDescription("offer", offer);
OffererJS: xmpp = createSessionInitiate(offer);
OffererJS->AnswererJS: <jingle action="session-initiate"/>
OffererJS->OffererUA: pc.startIce();
OffererUA->OffererJS: onicecandidate(cand);
OffererJS: createTransportInfo(cand);
OffererJS->AnswererJS: <jingle action="transport-info"/>
// session-initiate arrives at Answerer
AnswererJS->AnswererUA: pc = new PeerConnection();
AnswererJS: offer = parseSessionInitiate(xmpp);
AnswererJS->AnswererUA: pc.setRemoteDescription("offer", offer);
AnswererUA->AnswererJS: onaddstream(remoteStream);
// transport-infos arrive at Answerer
AnswererJS->AnswererUA: candidate = parseTransportInfo(xmpp);
AnswererJS->AnswererUA: pc.addIceCandidate(candidate);
AnswererUA->AnswererJS: onicecandidate(cand)
AnswererJS: createTransportInfo(cand);
AnswererJS->OffererJS: <jingle action="transport-info"/>
// transport-infos arrive at Offerer
OffererJS->OffererUA: candidates = parseTransportInfo(xmpp);
OffererJS->OffererUA: pc.addIceCandidate(candidates);
// Answerer accepts call
AnswererJS->AnswererUA: peer.addStream(localStream, null);
AnswererJS->AnswererUA: answer = peer.createAnswer(offer, null);
AnswererJS: xmpp = createSessionAccept(answer);
AnswererJS->AnswererUA: pc.setLocalDescription("answer", answer);
AnswererJS->OffererJS: <jingle action="session-accept"/>
// session-accept arrives at Offerer
OffererJS: answer = parseSessionAccept(xmpp);
OffererJS->OffererUA: peer.setRemoteDescription("answer", answer);
OffererUA->OffererJS: onaddstream(remoteStream);
// ICE Completes (at Answerer)
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AnswererUA->AnswererJS: onopen();
AnswererUA->OffererUA: Media
// ICE Completes (at Offerer)
OffererUA->OffererJS: onopen();
OffererUA->AnswererUA: Media
A.1.3. Adding video to a call, using XMPP
This example demonstrates an XMPP call, where the XMPP content-add
mechanism is used to add video media to an existing session. For
simplicity, candidate exchange is not shown.
Note that the offerer for the change to the session may be different
than the original call offerer.
// Offerer adds video stream
OffererJS->OffererUA: pc.addStream(videoStream)
OffererJS->OffererUA: offer = pc.createOffer(null);
OffererJS: xmpp = createContentAdd(offer);
OffererJS->OffererUA: pc.setLocalDescription("offer", offer);
OffererJS->AnswererJS: <jingle action="content-add"/>
// content-add arrives at Answerer
AnswererJS: offer = parseContentAdd(xmpp);
AnswererJS->AnswererUA: pc.setRemoteDescription("offer", offer);
AnswererJS->AnswererUA: answer = pc.createAnswer(offer, null);
AnswererJS->AnswererUA: pc.setLocalDescription("answer", answer);
AnswererJS: xmpp = createContentAccept(answer);
AnswererJS->OffererJS: <jingle action="content-accept"/>
// content-accept arrives at Offerer
OffererJS: answer = parseContentAccept(xmpp);
OffererJS->OffererUA: pc.setRemoteDescription("answer", answer);
A.1.4. Simultaneous add of video streams, using XMPP
This example demonstrates an XMPP call, where new video sources are
added at the same time to a call that already has video; since adding
these sources only affects one side of the call, there is no
conflict. The XMPP description-info mechanism is used to indicate
the new sources to the remote side.
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// Offerer and "Answerer" add video streams at the same time
OffererJS->OffererUA: pc.addStream(offererVideoStream2)
OffererJS->OffererUA: offer = pc.createOffer(null);
OffererJS: xmpp = createDescriptionInfo(offer);
OffererJS->OffererUA: pc.setLocalDescription("offer", offer);
OffererJS->AnswererJS: <jingle action="description-info"/>
AnswererJS->AnswererUA: pc.addStream(answererVideoStream2)
AnswererJS->AnswererUA: offer = pc.createOffer(null);
AnswererJS: xmpp = createDescriptionInfo(offer);
AnswererJS->AnswererUA: pc.setLocalDescription("offer", offer);
AnswererJS->OffererJS: <jingle action="description-info"/>
// description-info arrives at "Answerer", and is acked
AnswererJS: offer = parseDescriptionInfo(xmpp);
AnswererJS->OffererJS: <iq type="result"/> // ack
// description-info arrives at Offerer, and is acked
OffererJS: offer = parseDescriptionInfo(xmpp);
OffererJS->AnswererJS: <iq type="result"/> // ack
// ack arrives at Offerer; remote offer is used as an answer
OffererJS->OffererUA: pc.setRemoteDescription("answer", offer);
// ack arrives at "Answerer"; remote offer is used as an answer
AnswererJS->AnswererUA: pc.setRemoteDescription("answer", offer);
A.1.5. Call using SIP
This example demonstrates a simple SIP call (e.g. where the client
talks to a SIP proxy over WebSockets).
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// Call is initiated toward Answerer
OffererJS->OffererUA: pc = new PeerConnection();
OffererJS->OffererUA: pc.addStream(localStream, null);
OffererUA->OffererJS: onicecandidate(candidate);
OffererJS->OffererUA: offer = pc.createOffer(null);
OffererJS->OffererUA: pc.setLocalDescription("offer", offer);
OffererJS: sip = createInvite(offer);
OffererJS->AnswererJS: SIP INVITE w/ SDP
// INVITE arrives at Answerer
AnswererJS->AnswererUA: pc = new PeerConnection();
AnswererJS: offer = parseInvite(sip);
AnswererJS->AnswererUA: pc.setRemoteDescription("offer", offer);
AnswererUA->AnswererJS: onaddstream(remoteStream);
AnswererUA->OffererUA: onicecandidate(candidate);
// Answerer accepts call
AnswererJS->AnswererUA: peer.addStream(localStream, null);
AnswererJS->AnswererUA: answer = peer.createAnswer(offer, null);
AnswererJS: sip = createResponse(200, answer);
AnswererJS->AnswererUA: peer.setLocalDescription("answer", answer);
AnswererJS->OffererJS: 200 OK w/ SDP
// 200 OK arrives at Offerer
OffererJS: answer = parseResponse(sip);
OffererJS->OffererUA: peer.setRemoteDescription("answer", answer);
OffererUA->OffererJS: onaddstream(remoteStream);
OffererJS->AnswererJS: ACK
// ICE Completes (at Answerer)
AnswererUA->AnswererJS: onopen();
AnswererUA->OffererUA: Media
// ICE Completes (at Offerer)
OffererUA->OffererJS: onopen();
OffererUA->AnswererUA: Media
A.1.6. Handling early media (e.g. 1-800-GO FEDEX), using SIP
This example demonstrates how early media could be handled; for
simplicity, only the offerer side of the call is shown.
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// Call is initiated toward Answerer
OffererJS->OffererUA: pc = new PeerConnection();
OffererJS->OffererUA: pc.addStream(localStream, null);
OffererUA->OffererJS: onicecandidate(candidate);
OffererJS->OffererUA: offer = pc.createOffer(null);
OffererJS->OffererUA: pc.setLocalDescription("offer", offer);
OffererJS: sip = createInvite(offer);
OffererJS->AnswererJS: SIP INVITE w/ SDP
// 180 Ringing is received by offerer, w/ SDP
OffererJS: answer = parseResponse(sip);
OffererJS->OffererUA: pc.setRemoteDescription("pranswer", answer);
OffererUA->OffererJS: onaddstream(remoteStream);
// ICE Completes (at Offerer)
OffererUA->OffererJS: onopen();
OffererUA->AnswererUA: Media
// 200 OK arrives at Offerer
OffererJS: answer = parseResponse(sip);
OffererJS->OffererUA: pc.setRemoteDescription("answer", answer);
OffererJS->AnswererJS: ACK
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Appendix B. Change log
Changes in draft -02:
o Converted from nroff
o Removed comparisons to old approaches abandoned by the working
group
o Removed stuff that has moved to W3C specificaiton
o Align SDP handling with W3C draft
o Clarified section on forking.
Changes in draft -01:
o Added diagrams for architecture and state machine.
o Added sections on forking and rehydration.
o Clarified meaning of "pranswer" and "answer".
o Reworked how ICE restarts and media directions are controlled.
o Added list of parameters that can be changed in a description.
o Updated suggested API and examples to match latest thinking.
o Suggested API and examples have been moved to an appendix.
Changes in draft -00:
o Migrated from draft-uberti-rtcweb-jsep-02.
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Authors' Addresses
Justin Uberti
Google
747 6th Ave S
Kirkland, WA 98033
USA
Email: justin@uberti.name
Cullen Jennings
Cisco
170 West Tasman Drive
San Jose, CA 95134
USA
Email: fluffy@iii.ca
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