Network Working Group M. Westerlund
Internet-Draft Ericsson
Intended status: Standards Track C. Perkins
Expires: April 25, 2013 University of Glasgow
October 22, 2012
Multiple RTP Sessions on a Single Lower-Layer Transport
draft-westerlund-avtcore-transport-multiplexing-04
Abstract
This document specifies how multiple RTP sessions are to be
multiplexed on the same lower-layer transport, e.g. a UDP flow. It
discusses various requirements that have been raised and their
feasibility, which results in a solution with a certain
applicability. A solution is recommended and that solution is
provided in more detail, including signalling and examples.
Status of this Memo
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This Internet-Draft will expire on April 25, 2013.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
2. Conventions . . . . . . . . . . . . . . . . . . . . . . . . . 4
2.1. Terminology . . . . . . . . . . . . . . . . . . . . . . . 4
2.2. Requirements Language . . . . . . . . . . . . . . . . . . 5
3. Motivations . . . . . . . . . . . . . . . . . . . . . . . . . 5
3.1. NAT and Firewalls . . . . . . . . . . . . . . . . . . . . 5
3.2. No Transport Level QoS . . . . . . . . . . . . . . . . . . 5
3.3. Multiple RTP sessions . . . . . . . . . . . . . . . . . . 6
3.4. Usage of RTP Extensions . . . . . . . . . . . . . . . . . 6
3.5. Incremental Deployment . . . . . . . . . . . . . . . . . . 7
3.6. Summary . . . . . . . . . . . . . . . . . . . . . . . . . 7
4. Requirements . . . . . . . . . . . . . . . . . . . . . . . . . 7
4.1. Support Use of Multiple RTP Sessions . . . . . . . . . . . 7
4.2. Same SSRC Value in Multiple RTP Sessions . . . . . . . . . 8
4.3. SRTP . . . . . . . . . . . . . . . . . . . . . . . . . . . 8
4.4. Don't Redefine Used Bits . . . . . . . . . . . . . . . . . 9
4.5. Firewall Friendly . . . . . . . . . . . . . . . . . . . . 9
4.6. Monitoring and Reporting . . . . . . . . . . . . . . . . . 9
4.7. Usable Also Over Multicast . . . . . . . . . . . . . . . . 10
4.8. Incremental Deployment . . . . . . . . . . . . . . . . . . 10
5. Design Considerations . . . . . . . . . . . . . . . . . . . . 10
5.1. Location of SHIM . . . . . . . . . . . . . . . . . . . . . 10
5.2. ICE and DTLS-SRTP Integration . . . . . . . . . . . . . . 12
5.3. Signalling Fallback . . . . . . . . . . . . . . . . . . . 12
6. Specification . . . . . . . . . . . . . . . . . . . . . . . . 13
6.1. Shim Layer . . . . . . . . . . . . . . . . . . . . . . . . 14
6.2. Signalling . . . . . . . . . . . . . . . . . . . . . . . . 17
6.3. SRTP Key Management . . . . . . . . . . . . . . . . . . . 18
6.3.1. Security Description . . . . . . . . . . . . . . . . . 19
6.3.2. DTLS-SRTP . . . . . . . . . . . . . . . . . . . . . . 19
6.3.3. MIKEY . . . . . . . . . . . . . . . . . . . . . . . . 19
6.4. Examples . . . . . . . . . . . . . . . . . . . . . . . . . 20
6.4.1. RTP Packet with Transport Header . . . . . . . . . . . 20
6.4.2. SDP Offer/Answer example . . . . . . . . . . . . . . . 21
7. Open Issues . . . . . . . . . . . . . . . . . . . . . . . . . 25
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 26
9. Security Considerations . . . . . . . . . . . . . . . . . . . 26
10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 26
11. References . . . . . . . . . . . . . . . . . . . . . . . . . . 27
11.1. Normative References . . . . . . . . . . . . . . . . . . . 27
11.2. Informational References . . . . . . . . . . . . . . . . . 27
Appendix A. Possible Solutions . . . . . . . . . . . . . . . . . 29
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A.1. Header Extension . . . . . . . . . . . . . . . . . . . . . 29
A.2. Multiplexing Shim . . . . . . . . . . . . . . . . . . . . 30
A.3. Single Session . . . . . . . . . . . . . . . . . . . . . . 31
A.4. Use the SRTP MKI field . . . . . . . . . . . . . . . . . . 32
A.5. Use an Octet in the Padding . . . . . . . . . . . . . . . 33
A.6. Redefine the SSRC field . . . . . . . . . . . . . . . . . 33
Appendix B. Comparison . . . . . . . . . . . . . . . . . . . . . 34
B.1. Support of Multiple RTP Sessions Over Single Transport . . 34
B.2. Enable Same SSRC Value in Multiple RTP Sessions . . . . . 34
B.2.1. Avoid SSRC Translation in Gateways/Translation . . . . 34
B.2.2. Support Existing Extensions . . . . . . . . . . . . . 35
B.3. Ensure SRTP Functions . . . . . . . . . . . . . . . . . . 35
B.4. Don't Redefine Used Bits . . . . . . . . . . . . . . . . . 36
B.5. Firewall Friendly . . . . . . . . . . . . . . . . . . . . 37
B.6. Monitoring and Reporting . . . . . . . . . . . . . . . . . 38
B.7. Usable over Multicast . . . . . . . . . . . . . . . . . . 39
B.8. Incremental Deployment . . . . . . . . . . . . . . . . . . 39
B.9. Summary and Conclusion . . . . . . . . . . . . . . . . . . 40
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 41
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1. Introduction
There has been renewed interest for having a solution that allows
multiple RTP sessions [RFC3550] to use a single lower layer
transport, such as a bi-directional UDP flow. The main reason is the
cost of doing NAT/FW traversal for each individual flow. ICE and
other NAT/FW traversal solutions are clearly capable of attempting to
open multiple flows. However, there is both increased risk for
failure and an increased cost in the creation of multiple flows. The
increased cost comes as slightly higher delay in establishing the
traversal, and the amount of consumed NAT/FW resources. The latter
might be an increasing problem in the IPv4 to IPv6 transition period.
There is ongoing work on specifying how and when one RTP session may
contain multiple media types
[I-D.ietf-avtcore-multi-media-rtp-session]. That addresses certain
use cases, while this proposal addresses a different set of use cases
and motivations. This is further discussed in the section on
Motivations (Section 3). The classical method of having one RTP
session over a specific transport flow is still motivated for a
number of use cases, especially when flow based QoS is to be used for
some media streams.
This document draws up some requirements for consideration on how to
transport multiple RTP sessions over a single lower-layer transport.
These requirements had to be weighted as the combined set of
requirements result in that no known solution exist that can fulfill
them completely.
A number of possible solutions where considered and discussed with
respect to their properties. Based on that, the authors recommended
a shim layer variant as single solution, which specified in detail
including signalling solution and examples. The other considered
proposals and the comparison is available as appendices.
2. Conventions
2.1. Terminology
Some terminology used in this document.
Multiplexing: Unless specifically noted, all mentioning of
multiplexing in this document refer to the multiplexing of
multiple RTP Sessions on the same lower layer transport. It is
important to make this distinction as RTP does contain a number of
multiplexing points for various purposes, such as media formats
(Payload Type), media sources (SSRC), and RTP sessions.
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2.2. Requirements Language
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].
3. Motivations
This section looks at the motivations why an additional solution is
needed assuming that you can do both the classical method of having
one RTP session per transport flow as defined by the RTP
specification [RFC3550] and when you have multiple media types within
one RTP session [I-D.ietf-avtcore-multi-media-rtp-session].
3.1. NAT and Firewalls
The existence of NATs and Firewalls at almost all Internet access has
had implications on protocols like RTP that were designed to use
multiple transport flows. First of all, the NAT/FW traversal
solution one uses needs to ensure that all these transport flows are
established. This has three different impacts:
1. Increased delay to perform the transport flow establishment
2. The more transport flows, the more state and the more resource
consumption in the NAT and Firewalls. When the resource
consumption in NAT/FWs reaches their limits, unexpected behaviors
usually occur. Commonly resulting in service disruptions.
3. More transport flows means a higher risk that some transport flow
fails to be established, thus preventing the application to
communicate.
Using fewer transport flows reduces the risk of communication
failure, improved establishment behavior and less load on NAT and
Firewalls.
3.2. No Transport Level QoS
Many RTP-using applications don't utilize any network level Quality
of Service functions. Nor do they expect or desire any separation in
network treatment of its media packets, independent of whether they
are audio, video or text. When an application has no such desire, it
doesn't need to provide a transport flow structure that simplifies
flow based QoS.
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3.3. Multiple RTP sessions
The usage of multiple RTP sessions allow separation of media streams
that have different usages or purposes in an RTP based application,
for example to separate the video of a presenter or most important
current talker, from those of the listeners that not all end-points
receive. Also separation for different processing based on media
types such as audio and video in end-points and central nodes. Thus
providing the node with the knowledge that any SSRC within the
session is supposed to be processed in a similar or same way.
For simpler cases, where the streams within each media type need the
same processing, it is clearly possible to find other multiplex
solutions, for example based on the Payload Type and the differences
in encoding that the payload type allows to describe. This may
anyhow be insufficient when you get into more advanced usages where
you have multiple sources of the same media type, but for different
usages or as alternatives. For example when you have one set of
video sources that shows session participants and another set of
video sources that shares an application or presentation slides, you
likely want to separate those streams for various reasons such as
control, prioritization, QoS, methods for robustification, etc. In
those cases, using the RTP session for separation of properties is a
powerful tool. A tool with properties that need to be preserved when
providing a solution for how to use only a single lower-layer
transport.
For more discussion of the usage of RTP sessions verses other
multiplexing we recommend RTP Multiplexing Architecture
[I-D.westerlund-avtcore-multiplex-architecture].
3.4. Usage of RTP Extensions
Applications uses different sets of RTP extensions. The solution for
multiple media types in one RTP session
[I-D.ietf-avtcore-multi-media-rtp-session] is known to have
limitations that prevent the usage of the following RTP mechanisms
and extensions:
o XOR FEC (RFC5109)
o RTP Retransmission in session mode (RFC4588)
o Certain Layered Coding
A developed solution should minimize the number of RTP/RTCP extension
and mechanism that can't be used.
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3.5. Incremental Deployment
In various multi-party communication scenarios deployment can become
an issue if all session participants are required to have the
functionality before enabling its usage. This is especially
difficult in communication scenarios where not all possible
participants and their capabilities are know ahead of establishing
the communication session with some sub-set of the participants. At
least for centralized communication sessions it is desirable to have
a solution that enables allows the solution to be used on a single
leg without affecting any other leg, nor require advanced translation
functionality in any central node.
3.6. Summary
The center of the motivation is to ensure that the RTP session is a
available and usable tool also for applications that has no need for
network level separation of its media streams and wants to reduce its
exposure to any NAT or Firewall inconsistencies and minimize the
resource consumption. As a benefit a well designed solution will
enable incremental deployment and minimal limitations in what
existing RTP mechanisms or extensions that can be used by the RTP
using application.
4. Requirements
This section lists and discusses a number of potential requirements.
However, it is not difficult to realize that it is in fact possible
to put requirements that makes the set of feasible solutions an empty
set. It is thus necessary to consider which requirements that are
essential to fulfill and which can be compromised on to arrive at a
solution.
4.1. Support Use of Multiple RTP Sessions
Section 3.3 discusses a number of reasons why an application may like
to have multiple RTP sessions. Considering the motivations for this
work this must be an absolute requirement. We also are of the
opinion that the session provided by the solution must fulfill the
definition in the RTP [RFC3550] specification:
"The distinguishing feature of an RTP session is that each
maintains a full, separate space of SSRC identifiers (defined
next). The set of participants included in one RTP session
consists of those that can receive an SSRC identifier transmitted
by any one of the participants either in RTP as the SSRC or a CSRC
(also defined below) or in RTCP."
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4.2. Same SSRC Value in Multiple RTP Sessions
Two different RTP sessions being multiplexed on the same lower layer
transport need to be able to use the same SSRC value. This is a
absolute requirement, for two reasons:
1. To avoid mandating SSRC assignment rules that are coordinated
between the sessions. If the RTP sessions multiplexed together
must have unique SSRC values, then additional code that works
between RTP Sessions is needed in the implementations. Thus
raising the bar for implementing this solution. In addition, if
one gateways between parts of a system using this multiplexing
and parts that aren't multiplexing, the part that isn't
multiplexing must also fulfill the requirements on how SSRC is
assigned or force the gateway to translate SSRCs. Translating
SSRC is actually hard as it requires one to understand the
semantics of all current and future RTP and RTCP extensions.
Otherwise a barrier for deploying new extensions is created.
2. There are some few RTP extensions that currently rely on being
able to use the same SSRC in different RTP sessions:
* XOR FEC (RFC5109)
* RTP Retransmission in session mode (RFC4588)
* Certain Layered Coding
4.3. SRTP
SRTP [RFC3711] is one of the most commonly used security solutions
for RTP. In addition, it is the only one defined by IETF that is
integrated into RTP. This integration has several aspects that needs
to be considered when designing a solution for multiplexing RTP
sessions on the same lower layer transport.
Determining Crypto Context: SRTP first of all needs to know which
session context a received or to-be-sent packet relates to. It
also normally relies on the lower layer transport to identify the
session. It uses the Master Key Indicatior (MKI), if present, to
determine which key set is to be used. Then the SSRC and sequence
number are used by most crypto suites, including the most common
use of AES Counter Mode, to actually generate the correct cipher
stream.
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Unencrypted Headers: SRTP has chosen to leave the RTP headers and
the first two 32-bit words of the first RTCP header unencrypted,
to allow for both header compression and monitoring to work also
in the presence of encryption. As these fields are in clear text
they are used in most crypto suites for SRTP to determine how to
protect or recover the plain text.
It is here important to contrast SRTP against a set of other possible
protection mechanisms. DTLS, TLS, and IPsec are all protecting and
encapsulating the entire RTP and RTCP packets. They don't perform
any partial operations on the RTP and RTCP packets. Any change that
is considered to be part of the RTP and RTCP packet is transparent to
them, but possibly not to SRTP. Thus the impact on SRTP operations
must be considered when defining a mechanism.
4.4. Don't Redefine Used Bits
As the core of RTP is in use in many systems and has a really large
deployment story and numerous implementations, changing any of the
field definitions is highly problematic. First of all, the
implementations need to change to support this new semantics.
Secondly, you get a large transition issue when you have some session
participants that support the new semantics and some that don't.
Combing the two behaviors in the same session can force the
deployment of costly and less than perfect translation devices.
4.5. Firewall Friendly
It is desirable that current Firewalls will accept the solutions as
normal RTP packets. However, in the authors' opinion we can't let
the firewall stifle invention and evolution of the protocol. It is
also necessary to be aware that a change that will make most deep
inspecting firewall consider the packet as not valid RTP/RTCP will
have a more difficult deployment story.
4.6. Monitoring and Reporting
It is desirable that a third party monitor can still operate on the
multiplexed RTP Sessions. It is however likely that they will
require an update to correctly monitor and report on multiplexed RTP
Sessions.
Another type of function to consider is packet sniffers and their
selector filters. These may be impacted by a change of the fields.
An observation is that many such systems are usually quite rapidly
updated to consider new types of standardized or simply common packet
formats.
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4.7. Usable Also Over Multicast
It is desirable that a solution should be possible to use also when
RTP and RTCP packets are sent over multicast, both Any Source
Multicast (ASM) and Single Source Multicast (SSM). The reason for
this requirement is to allow a system using RTP to use the same
configuration regardless of the transport being done over unicast or
multicast. In addition, multicast can't be claimed to have an issue
with using multiple ports, as each multicast group has a complete
port space scoped by address.
4.8. Incremental Deployment
A good solution has the property that in topologies that contains RTP
mixers or Translators, a single session participant can enable
multiplexing without having any impact on any other session
participants. Thus a node should be able to take a multiplexed
packet and then easily send it out with minimal or no modification on
another leg of the session, where each RTP session is transported
over its own lower-layer transport. It should also be as easy to do
the reverse forwarding operation.
5. Design Considerations
When defining a SHIM solution for identifying RTP sessions over a
single transport layer there has been some special considerations
that is discussed in this section.
5.1. Location of SHIM
A major question affecting the SHIM is the location of the SHIM
header providing the Identifier of the session the packet relate to.
This section will discuss in detail about the impact of making the
different choices.
Identified aspects to consider are:
Possibility to Process: A prefixed shim header, i.e. between the
transport protocol and the RTP/RTCP packet header has the
advantage that any node on the network that likes to include the
header in any per-packet processing can reach it. Reasons for
per-packet processing are:
A. Quality of Service classification
B. SHIM ingress or egress
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C. Monitoring
Many routers or similar devices can only read and process the
first N bytes of the whole packet, where N is commonly on the
order of 64-128 bytes. Any other type of processing means putting
the packet on the slow path. Thus a prefixed solution enables
this processing while a post fixed solution will most likely
forever prevent this type of devices to process it.
Legacy Processing: Packets or at least flows of the type IP/UDP/RTP
can in many cases be identified in Deep Packet Inspection,
Firewalls or other network entities that concern themselves with
trying determine what traffic that flows in a particular packet.
These nodes can clearly be updated but until they have they may
create a hinder against deployment. Thus a post fix gives likely
the least resistance for initial deployment. However, also for
postfix location the deployment can be hindered in cases multiple
RTP sessions using the same SSRC values due to irregular behavior
of the fields for what the third party believes is one media
stream rather than multiple ones. The prefixed will however
maintain the long-term capabilities of such devices assuming they
can be updated to include the SHIM header as part of the
classification.
Header Compression: The different header compression techniques that
has been developed compresses IP/UDP/RTP as complete combination.
If one instead have a IP/UDP/SHIM/RTP then the compression for the
full set may not work or poorly. Instead only IP/UDP header
compression is likely to be applied. Thus a prefix will loose
some compression efficiency until compression profiles for IP/UDP/
SHIM/RTP has been developed, implemented and deployed. Postfix
don't have that issue, but nor can it ever gain anything from
header compression which an prefixed solution could once an
updated profile is deployed. Postfix also will have reduced
efficiency compressing sessions when the same SSRC is used in two
different RTP sessions as the RTP header fields like sequence
number etc will not behave as expected and need frequent explicit
updates.
The question of a prefixed or a postfixed header comes down to a
trade-off between long term usability and deployment issues:
Prefixed: Long term good possibility to adapt any network function
that needs to take the SHIM header into account. At the same time
any function that tries to analyze packets and because of that may
block the packets will be a hinder to deployment.
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Postfixed: This solution will likely short term have the best
possibilities to deploy successfully. However, long term this
choice will likely prevent many network nodes that like to be
capable of separating the RTP sessions being multiplexed together
from successfully doing that.
After discussion in the WG it has been determined that prefixed is
the prefered solution.
5.2. ICE and DTLS-SRTP Integration
When using ICE [RFC5245] or DTLS-SRTP [RFC5764] or both with RTP
there exist the issue that RTP, STUN [RFC5389] and DTLS-SRTP are
simultanously in use over the same lower layer transport flow, like
UDP. This multiplexing is based on the value of the first byte of
the lower layer transport payload as discussed in Section 5.1.2 of
DTLS-SRTP [RFC5764].
The replacement of a single RTP session with the multiple RTP
sessions idenfied by a SHIM must not be missidentified to be either
STUN or DTLS-SRTP or any other protocol intending to take the
available free code-points in the range 193-255 (Decimal). Thus a
prefixed SHIM must have its first byte have the two first bits set to
10 (Binary). Having the SHIM share the identity of RTP is not an
issue as one must have mutual agreement that the SHIM is used instead
of RTP.
Note: This limits a single byte SHIM to only allow a maximum of 64
RTP sessions over a single transport flow.
5.3. Signalling Fallback
There exist an important aspect in how the SDP signalling functions,
especially Offer/Answer [RFC3264]. The initial idea for the
signalling was to build on top of bundle
[I-D.ietf-mmusic-sdp-bundle-negotiation] which in its default
function negotiate multiple media types over one RTP session
[I-D.ietf-avtcore-multi-media-rtp-session]. If the signalling for
the solution that main purpose is to enable multiple RTP sessions
results in those cases the peer doesn't support this specification
the communicating peer can end up in single RTP session if the peer
supports that.
We consider it important that in the signalling design that the
application developer can decide what type of fallback that will
occur. It is also important to consider that one have to signal SHIM
based multiplexing of RTP sessions that are in fact of the type with
multiple media types. Thus the signalling for SHIM must be able to
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describe multiple different scenarios:
1. Multiple RTP sessions multiplexed together using SHIM over one
transport
2. Like 1 but where at least one RTP session is containing multiple
media types
3. Like 1, but where the peer doesn't support SHIM and the initiator
wants to fallback to independent transports
4. Like 2, but where the peer doesn't support SHIM and wants to
fallback to multiple BUNDLED sessions over independent
transports.
In addition it must be possible to have multiple different transports
where each is a SHIM multiplex. This is to support decomposed end-
points or cases where certain media traffic is required to go to a
central processing node while others goes directly to a peer.
To enable all of these scenarios we propose a solution where each
indicates SHIM multiplex is indicated as its own grouping attribute
across all media blocks that are included in some form in the
multiplex. This resulting in that these media blocks fall under a
form of BUNDLE super set. This super set will also have some of
bundles restrictions on the transport layer, but not on higher layer.
Which Session ID pair a particular media block is associated is
signalled using a SDP attribute (a=session-mux-id) in each media
block. When multiple media block are assigned the same session ID
pair, they form a RTP session with multiple media types and have the
full restriction of bundle between them.
The method of fallback is indicated by providing explicit BUNDLE
grouping in addition to the SHIM when the fallback from SHIM is to
BUNDLE.
Note: Signalling solution is awaiting resolution of design path for
bundle and will then consider that solution and issues raised.
6. Specification
This section contains the specification of the RTP session
multiplexing SHIM, using an explicit session identifier of the
encapsulated payload.
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6.1. Shim Layer
This solution is based on a shim layer that is inserted in the stack
between the regular RTP and RTCP packets and the transport layer
being used by the RTP sessions. Thus the layering looks like the
following:
+---------------------+
| RTP / RTCP Packet |
+---------------------+
| Session ID Layer |
+---------------------+
| Transport layer |
+---------------------+
Stack View with Session ID SHIM
The above stack is in fact a layered one as it does allow multiple
RTP Sessions to be multiplexed on top of the Session ID shim layer.
This enables the example presented in Figure 1 where four sessions,
S1-S4 is sent over the same Transport layer and where the Session ID
layer will combine and encapsulate them with the session ID on
transmission and separate and decapsulate them on reception.
+-------------------+
| S1 | S2 | S3 | S4 |
+-------------------+
| Session ID Layer |
+-------------------+
| Transport layer |
+-------------------+
Figure 1: Multiple RTP Session On Top of Session ID Layer
The Session ID layer encapsulates one RTP or RTCP packet from a given
RTP session and prefixes the 2-byte Session ID layer to the packet.
The Session ID layer is depicted below (Figure 2) and consists of
first 2 fixed bit values (10b) followed by a 14 bits unsigned integer
field with the Session ID (SID) value.
0 1
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|1 0| Session ID (SID) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 2: Session ID layer
Each RTP session being multiplexed on top of a given transport layer
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is assigned either a single or a pair of unique SID in the range
0-16383. The reason for assigning a pair of SIDs to a given RTP
session are for RTP Sessions that doesn't support "Multiplexing RTP
Data and Control Packets on a Single Port" [RFC5761] to still be able
to use a single 5-tuple. The reasons for supporting this extra
functionality is that RTP and RTCP multiplexing based on the payload
type/packet type fields enforces certain restrictions on the RTP
sessions. These restrictions may not be acceptable. As this
solution does not have these restrictions, performing RTP and RTCP
multiplexing in this way has benefits.
Each Session ID value space is scoped by the underlying transport
protocol. Common transport protocols like UDP [RFC0768], DCCP
[RFC4340], TCP [RFC0793], and SCTP [RFC4960] can all be scoped by one
or more 5-tuple (Transport protocol, source address and port,
destination address and port). The case of multiple 5-tuples occur
in the case of multi-unicast topologies, also called meshed
multiparty RTP sessions or in case any application would need more
than 8192 RTP sessions.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-------------------------------+
|1 0| Session ID |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<+
|V=2|P|X| CC |M| PT | sequence number | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| timestamp | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| synchronization source (SSRC) identifier | |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ |
| contributing source (CSRC) identifiers | |
| .... | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| RTP extension (OPTIONAL) | |
+>+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| | payload ... | |
| | +-------------------------------+ |
| | | RTP padding | RTP pad count | |
+>+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<+
| ~ SRTP MKI (OPTIONAL) ~ |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| : authentication tag (RECOMMENDED) : |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
+- Encrypted Portion* Authenticated Portion ---+
Figure 3: SRTP Packet encapsulated by Session ID Layer
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0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-------------------------------+
|1 0| Session ID |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<+
|V=2|P| RC | PT=SR or RR | length | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| SSRC of sender | |
+>+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ |
| ~ sender info ~ |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| ~ report block 1 ~ |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| ~ report block 2 ~ |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| ~ ... ~ |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| |V=2|P| SC | PT=SDES=202 | length | |
| +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ |
| | SSRC/CSRC_1 | |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| ~ SDES items ~ |
| +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ |
| ~ ... ~ |
+>+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ |
| |E| SRTCP index | |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<+
| ~ SRTCP MKI (OPTIONAL) ~ |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| : authentication tag : |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
+-- Encrypted Portion Authenticated Portion -----+
Figure 4: SRTCP packet encapsulated by Session ID layer
The processing in a receiver when the Session ID layer is present
will be to
1. Pick up the packet from the lower layer transport
2. Inspect the SID field value
3. Strip the SID field from the packet
4. Forward it to the (S)RTP Session context identified by the SID
value
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6.2. Signalling
Note: This section may need updating as the direction of the solution
for Bundle has settled and the impact of the raised issues has been
analyzed.
The use of the Session ID layer needs to be explicitly agreed on
between the communicating parties. Each RTP Session the application
uses must in addition to the regular configuration such as payload
types, RTCP extension etc, have both the underlying 5-tuple (source
address and port, destination address and port, and transport
protocol) and the Session ID used for the particular RTP session.
The signalling requirement is to assign unique Session ID values to
all RTP Sessions being sent over the same 5-tuple. The same Session
ID shall be used for an RTP session independently of the traffic
direction. Note that nothing prevents a multi-media application from
using multiple 5-tuples if desired for some reason, in which case
each 5-tuple has its own session ID value space.
This section defines how to negotiate the use of the Session ID
layer, using the Session Description Protocol (SDP) Offer/Answer
mechanism [RFC3264]. A new SDP grouping semantics is defined "SHIM"
and a new media-level SDP attribute, 'session-mux-id. The attribute
allows each media description ("m=" line) associated with a 'SHIM'
group to be identified in which RTP session it belongs.
The 'session-mux-id' attribute is included for a media description,
in order to indicate the Session ID for that particular media
description. Every media description that shares a common attribute
value is assumed to be part of a single RTP session. An SDP Offerer
MUST include the 'session-mux-id' attribute for every media
description associated with a 'SHIM' group. If the SDP Answer does
not contain the SHIM group, the SDP Offerer MUST NOT use SHIM based
layering. However, if that is separate RTP sessions or BUNDLE is
determined on what was present in the offer and answer. This will
depend on what the offering party likes to happen. If they want a
failure to negotiate a SHIM, instead may be one or more bundle groups
then also the BUNDLE grouping is included in the offer. If the SDP
Answer still describes a 'BUNDLE' group, the procedures in
[I-D.ietf-mmusic-sdp-bundle-negotiation] apply. If not independent
transports and sessions are used.
An SDP Answerer MUST NOT include the 'SHIM' group and
'session-mux-id' attribute in an SDP Answer, unless they where
included in the SDP Offer.
The attribute has the following ABNF [RFC5234] definition.
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Session-mux-id-attr = "a=session-mux-id:" SID *SID-prop
SID = SID-value / SID-pairs
SID-value = 1*3DIGIT / "NoN"
SID-pairs = SID-value "/" SID-value ; RTP/RTCP SIDs
SID-prop = SP assignment-policy / prop-ext
prop-ext = token "=" value
assignment-policy = "policy=" ("tentative" / "fixed")
The SHIM group SHALL contain all media descriptions that are intended
to be sent over the same transport flow, independent of Session ID.
For all media descriptions part of the same SHIM group the transport
parameters, i.e. ports, ICE-candidates etc MUST be the same and
handled as described by BUNDLE. Note, the parameters related to the
RTP session does not need to be same.
For media descriptions that have the same value of the Session ID
SHALL be treated the same way as if they where part of a BUNDLE
group, independently if that is indicated or not in the SDP.
The SID property "policy" is used in negotiation by an end-point to
indicate if the session ID values are merely a tentative suggestion
or if they must have these values. This is used when negotiating SID
for multi-party RTP sessions to support shared transports such as
multicast or RTP translators that are unable to produce renumbered
SIDs on a per end-point basis. The normal behavior is that the offer
suggest a tentative set of values, indicated by "policy=tentative".
These SHOULD be accepted by the peer unless that peer negotiate
session IDs on behalf of a centralized policy, in which case it MAY
change the value(s) in the answer. If the offer represents a policy
that does not allow changing the session ID values, it can indicate
that to the answerer by setting the policy to "fixed". This enables
the answering peer to either accept the value or indicate that there
is a conflict in who is performing the assignment by setting the SID
value to NoN (Not a Number). Offerer and answerer SHOULD always
include the policy they are operating under. Thus, in case of no
centralized behaviors, both offerer and answerer will indicate the
tentative policy.
6.3. SRTP Key Management
Key management for SRTP do needs discussion as we do cause multiple
SRTP sessions to exist on the same underlying transport flow. Thus
we need to ensure that the key management mechanism still are
properly associated with the SRTP session context it intends to key.
To ensure that we do look at the three SRTP key management mechanism
that IETF has specified, one after another.
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6.3.1. Security Description
Session Description Protocol (SDP) Security Descriptions for Media
Streams [RFC4568] as being based on SDP has no issue with the RTP
session multiplexing on lower layer specified here. The reason is
that the actual keying is done using a media level SDP attribute.
Thus the attribute is already associated with a particular media
description. A media description that also will have an instance of
the "a=session-mux-id" attribute carrying the SID value/pair used
with this particular crypto parameters.
6.3.2. DTLS-SRTP
Datagram Transport Layer Security (DTLS) Extension to Establish Keys
for the Secure Real-time Transport Protocol (SRTP) [RFC5764] is a
keying mechanism that works on the media plane on the same lower
layer transport that SRTP/SRTCP will be transported over.
The most direct solution would be to use the SHIM and the SID context
identifier to be applied also on DTLS packets. Thus using the same
SID that is used with RTP and/or RTCP also for the DTLS message
intended to key that particular SRTP and/or SRTCP flow(s). This of
course requires independent usage of DTLS-SRTP for each RTP session.
In addition it requires changing the layering for DTLS-SRTP as well
as RTP. Thus this behavior doesn't gain you anything in regards to
key-management when using SHIM and have some costs.
Instead we propose that an DTLS-SRTP key-derivation change is
introduced. By including the Session ID value in the derivation of
the keying material a single DTLS-SRTP key-management operation could
apply keys and parameters for all the RTP sessions in the same
transport flow. Thus the keying cost is significantly reduced,
especially in regards to network communication and delay impact and
vunerability to packet loss.
Details to be written up.
6.3.3. MIKEY
MIKEY: Multimedia Internet KEYing [RFC3830] is a key management
protocol that has several transports. In some cases it is used
directly on a transport protocol such as UDP, but there is also a
specification for how MIKEY is used with SDP "Key Management
Extensions for Session Description Protocol (SDP) and Real Time
Streaming Protocol (RTSP)" [RFC4567].
Lets start with the later, i.e. the SDP transport, which shares the
properties with Security Description in that is can be associated
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with a particular media description in a SDP. As long as one avoids
using the session level attribute one can be certain to correctly
associate the key exchange with a given SRTP/SRTCP context.
It does appear that MIKEY directly over a lower layer transport
protocol will have similar issues as DTLS.
6.4. Examples
6.4.1. RTP Packet with Transport Header
The below figure contains an RTP packet with SID field encapsulated
by a UDP packet (added UDP header).
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Source Port | Destination Port |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Length | Checksum |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|1 0| Session ID |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<+
|V=2|P|X| CC |M| PT | sequence number | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| timestamp | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| synchronization source (SSRC) identifier | |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ |
| contributing source (CSRC) identifiers | |
| .... | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| RTP extension (OPTIONAL) | |
+>+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| | payload ... | |
| | +-------------------------------+ |
| | | RTP padding | RTP pad count | |
+>+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<+
| ~ SRTP MKI (OPTIONAL) ~ |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| : authentication tag (RECOMMENDED) : |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
+- Encrypted Portion* Authenticated Portion ---+
SRTP Packet Encapsulated by Session ID Layer
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6.4.2. SDP Offer/Answer example
6.4.2.1. Basic Example
This section contains SDP offer/answer examples. First one example
of successful SHIMing, and then two where fallback occurs. The
fallback option here is to fallback to individual transports, thus no
BUNDLE group.
In the below SDP offer, one audio and one video is being offered.
The audio is using SID 0, and the video is using SID 1 to indicate
that they are different RTP sessions despite being offered over the
same 5-tuple.
v=0
o=alice 2890844526 2890844526 IN IP4 atlanta.example.com
s=
c=IN IP4 atlanta.example.com
t=0 0
a=group:SHIM foo bar
m=audio 10000 RTP/AVP 0 8 97
b=AS:200
a=mid:foo
a=session-mux-id:0 policy=tentative
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
m=video 10000 RTP/AVP 31 32
b=AS:1000
a=mid:bar
a=session-mux-id:1 policy=tentative
a=rtpmap:31 H261/90000
a=rtpmap:32 MPV/90000
The SDP answer from an end-point that supports this BUNDLEing:
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v=0
o=bob 2808844564 2808844564 IN IP4 biloxi.example.com
s=
c=IN IP4 biloxi.example.com
t=0 0
a=group:SHIM foo bar
m=audio 20000 RTP/AVP 0
b=AS:200
a=mid:foo
a=session-mux-id:0 policy=tentative
a=rtpmap:0 PCMU/8000
m=video 20000 RTP/AVP 32
b=AS:1000
a=mid:bar
a=session-mux-id:1 policy=tentative
a=rtpmap:32 MPV/90000
The SDP answer from an end-point that does not support this SHIMing.
v=0
o=bob 2808844564 2808844564 IN IP4 biloxi.example.com
s=
c=IN IP4 biloxi.example.com
t=0 0
m=audio 20000 RTP/AVP 0
b=AS:200
a=rtpmap:0 PCMU/8000
m=video 30000 RTP/AVP 32
b=AS:1000
a=rtpmap:32 MPV/90000
6.4.2.2. Advanced Example
In this example we have two BUNDLED sessions, one with audio and
video and one with XOR based FEC [RFC5109] for the audio and the
video. These two RTP session are then SHIMed into a single transport
flow.
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v=0
o=alice 2890844526 2890844526 IN IP4 atlanta.example.com
s=
c=IN IP4 atlanta.example.com
t=0 0
a=group:SHIM foo bar 1 2
a=group:BUNDLE 1 2
a=group:BUNDLE foo bar
a=group:FEC foo 1
a=group:FEC bar 2
m=audio 10000 RTP/AVP 0 8 97
b=AS:200
a=mid:foo
a=session-mux-id:0 policy=tentative
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
m=video 10000 RTP/AVP 31 32
b=AS:1000
a=mid:bar
a=session-mux-id:0 policy=tentative
a=rtpmap:31 H261/90000
a=rtpmap:32 MPV/90000
m=audio 10000 RTP/AVP 100
b=AS:100
a=rtpmap:100 ulpfec/8000
a=mid:1
a=session-mux-id:1 policy=tentative
m=video 10000 RTP/AVP 101
b=AS:500
a=mid:2
a=session-mux-id:1 policy=tentative
a=rtpmap:101 ulpfec/90000
The SDP answer of a client supporting
[I-D.ietf-mmusic-sdp-bundle-negotiation] but not this SHIMing would
look like this:
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v=0
o=bob 2808844564 2808844564 IN IP4 biloxi.example.com
s=
c=IN IP4 biloxi.example.com
t=0 0
a=group:BUNDLE 1 2
a=group:BUNDLE foo bar
a=group:FEC foo 1
a=group:FEC bar 2
m=audio 20000 RTP/AVP 0 8 97
b=AS:200
a=mid:foo
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
m=video 20000 RTP/AVP 31 32
b=AS:1000
a=mid:bar
a=rtpmap:31 H261/90000
a=rtpmap:32 MPV/90000
m=audio 20002 RTP/AVP 100
b=AS:100
a=rtpmap:100 ulpfec/8000
a=mid:1
m=video 20002 RTP/AVP 101
b=AS:500
a=mid:2
a=rtpmap:101 ulpfec/90000
In the above case two different RTP sessions, both being of a BUNDLE
type with multiple media types in each. The two established flows
will be Alice:10000<->Bob:20000, and Alice:10000<->Bob:20002.
If the peer did support neither of the SHIM or BUNDLE extension the
answer would look like this:
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v=0
o=bob 2808844564 2808844564 IN IP4 biloxi.example.com
s=
c=IN IP4 biloxi.example.com
t=0 0
a=group:FEC foo 1
a=group:FEC bar 2
m=audio 20000 RTP/AVP 0 8 97
b=AS:200
a=mid:foo
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
m=video 20002 RTP/AVP 31 32
b=AS:1000
a=mid:bar
a=rtpmap:31 H261/90000
a=rtpmap:32 MPV/90000
m=audio 20004 RTP/AVP 100
b=AS:100
a=rtpmap:100 ulpfec/8000
a=mid:1
m=video 20006 RTP/AVP 101
b=AS:500
a=mid:2
a=rtpmap:101 ulpfec/90000
In this case four different transport flows would be established for
RTP, each with a different RTP session over them. The answer also
knows the binding between the sessions with FEC and their source data
thanks to the FEC specification.
7. Open Issues
This work is still in the early phase of specification. This section
contains a list of open issues where the author desires some input.
1. In Section 6.2 there is a discussion of which parameters that
must be configured. The scope of these rules and if they do make
sense needs additional discussion.
2. Can we provide better control so that applications that doesn't
desire fallback to single RTP session when Multiplexing shim
fails to be supported but Bundle is supported ends up with a
better alternative?
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3. The details for how to do key-derivation, preferably in such a
way that it can be reused by multiple key-management solutions
like MIKEY and DTLS-SRTP
4. The signalling solution will be revisited when the BUNDLE
solution discussion has yeild some result.
8. IANA Considerations
This document request the registration of one SDP attribute. Details
of the registration to be filled in.
9. Security Considerations
The security properties of the Session ID layer is depending on what
mechanism is used to protect the RTP and RTCP packets of a given RTP
session. If IPsec or transport layer security solutions such as DTLS
or TLS are being used then both the encapsulated RTP/RTCP packets and
the session ID layer will be protected by that security mechanism.
Thus potentially providing both confidentiality, integrity and source
authentication. If SRTP is used, the session ID layer will not be
directly protected by SRTP. However, it will be implicitly integrity
protected (assuming the RTP/RTCP packet is integrity protected) as
the only function of the field is to identify the session context.
Thus any modification of the SID field will attempt to retrieve the
wrong SRTP crypto context. If that retrieval fails, the packet will
be anyway be discarded. If it is successful, the context will not
lead to successful verification of the packet.
10. Acknowledgements
This document is based on the input from various people, especially
in the context of the RTCWEB discussion of how to use only a single
lower layer transport. The RTP and RTCP packet figures are borrowed
from RFC3711. The SDP example is extended from the one present in
[I-D.ietf-mmusic-sdp-bundle-negotiation]. Eric Rescorla contributed
the basic idea of optimizing the DTLS-SRTP key-management by
modifying the key derivation process.
The proposal in Appendix A.5 is original suggested by Colin Perkins.
The idea in Appendix A.6 is from an Internet Draft
[I-D.rosenberg-rtcweb-rtpmux] written by Jonathan Rosenberg et. al.
The proposal in Appendix A.3 is a result of discussion by a group of
people at IETF meeting #81 in Quebec.
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11. References
11.1. Normative References
[I-D.ietf-mmusic-sdp-bundle-negotiation]
Holmberg, C. and H. Alvestrand, "Multiplexing Negotiation
Using Session Description Protocol (SDP) Port Numbers",
draft-ietf-mmusic-sdp-bundle-negotiation-01 (work in
progress), August 2012.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
[RFC5234] Crocker, D. and P. Overell, "Augmented BNF for Syntax
Specifications: ABNF", STD 68, RFC 5234, January 2008.
11.2. Informational References
[I-D.ietf-avtcore-multi-media-rtp-session]
Westerlund, M., Perkins, C., and J. Lennox, "Multiple
Media Types in an RTP Session",
draft-ietf-avtcore-multi-media-rtp-session-00 (work in
progress), October 2012.
[I-D.lennox-rtcweb-rtp-media-type-mux]
Rosenberg, J. and J. Lennox, "Multiplexing Multiple Media
Types In a Single Real-Time Transport Protocol (RTP)
Session", draft-lennox-rtcweb-rtp-media-type-mux-00 (work
in progress), October 2011.
[I-D.rosenberg-rtcweb-rtpmux]
Rosenberg, J., Jennings, C., Peterson, J., Kaufman, M.,
Rescorla, E., and T. Terriberry, "Multiplexing of Real-
Time Transport Protocol (RTP) Traffic for Browser based
Real-Time Communications (RTC)",
draft-rosenberg-rtcweb-rtpmux-00 (work in progress),
July 2011.
[I-D.westerlund-avtcore-multiplex-architecture]
Westerlund, M., Burman, B., Perkins, C., and H.
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Alvestrand, "Guidelines for using the Multiplexing
Features of RTP",
draft-westerlund-avtcore-multiplex-architecture-02 (work
in progress), July 2012.
[RFC0768] Postel, J., "User Datagram Protocol", STD 6, RFC 768,
August 1980.
[RFC0793] Postel, J., "Transmission Control Protocol", STD 7,
RFC 793, September 1981.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264,
June 2002.
[RFC3830] Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K.
Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830,
August 2004.
[RFC4340] Kohler, E., Handley, M., and S. Floyd, "Datagram
Congestion Control Protocol (DCCP)", RFC 4340, March 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.
[RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session
Description Protocol (SDP) Security Descriptions for Media
Streams", RFC 4568, July 2006.
[RFC4960] Stewart, R., "Stream Control Transmission Protocol",
RFC 4960, September 2007.
[RFC5109] Li, A., "RTP Payload Format for Generic Forward Error
Correction", RFC 5109, December 2007.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245,
April 2010.
[RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP
Header Extensions", RFC 5285, July 2008.
[RFC5389] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
"Session Traversal Utilities for NAT (STUN)", RFC 5389,
October 2008.
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[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, April 2009.
[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port", RFC 5761, April 2010.
[RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for the Secure
Real-time Transport Protocol (SRTP)", RFC 5764, May 2010.
Appendix A. Possible Solutions
This section documents the solutions explored when selecting a SHIM
based one and discusses their feasibility.
A.1. Header Extension
One proposal is to define an RTP header extension [RFC5285] that
explicitly enumerates the session identifier in each packet. This
proposal has some merits regarding RTP, since it uses an existing
extension mechanism; it explicitly enumerates the session allowing
for third parties to associate the packet to a given RTP session; and
it works with SRTP as currently defined since a header extension is
by default not encrypted, and is thus readable by the receiving stack
without needing to guess which session it belongs to and attempt to
decrypt it. This approach does, however, conflict with the
requirement from [RFC5285] that "header extensions using this
specification MUST only be used for data that can be safely ignored
by the recipient", since correct processing of the received packet
depends on using the header extension to demultiplex it to the
correct RTP session.
Using a header extension also result in the session ID is in the
integrity protected part of the packet. Thus a translator between
multiplexed and non-multiplexed has the options:
1. to be part of the security context to verify the field
2. to be part of the security context to verify the field and remove
it before forwarding the packet
3. to be outside of the security context and leave the header
extension in the packet. However, that requires successful
negotiation of the header extension, but not of the
functionality, with the receiving end-points.
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The biggest existing hurdle for this solution is that there exist no
header extension field in the RTCP packets. This requires defining a
solution for RTCP that allows carrying the explicit indicator,
preferably in a position that isn't encrypted by SRTCP. However, the
current SRTCP definition does not offer such a position in the
packet.
Modifying the RR or SR packets is possible using profile specific
extensions. However, that has issues when it comes to deployability
and in addition any information placed there would end up in the
encrypted part.
Another alternative could be to define another RTCP packet type that
only contains the common header, using the 5 bits in the first byte
of the common header to carry a session id. That would allow SRTCP
to work correctly as long it accepts this new packet type being the
first in the packet. Allowing a non-SR/RR packet as the first packet
in a compound RTCP packet is also needed if an implementation is to
support Reduced Size RTCP packets [RFC5506]. The remaining downside
with this is that all stack implementations supporting multiplexing
would need to modify its RTCP compound packet rules to include this
packet type first. Thus a translator box between supporting nodes
and non-supporting nodes needs to be in the crypto context.
This solution's per packet overhead is expected to be 64-bits for
RTCP. For RTP it is 64-bits if no header extension was otherwise
used, and an additional 16 bits (short header), or 24 bits plus (if
needed) padding to next 32-bits boundary if other header extensions
are used.
A.2. Multiplexing Shim
This proposal is to prefix or postfix all RTP and RTCP packets with a
session ID field. This field would be outside of the normal RTP and
RTCP packets, thus having no impact on the RTP and RTCP packets and
their processing. An additional step of demultiplexing processing
would be added prior to RTP stack processing to determine in which
RTP session context the packet shall be included. This has also no
impact on SRTP/SRTCP as the shim layer would be outside of its
protection context. The shim layer's session ID is however
implicitly integrity protected as any error in the field will result
in the packet being placed in the wrong or non-existing context, thus
resulting in a integrity failure if processed by SRTP/SRTCP.
This proposal is quite simple to implement in any gateway or
translating device that goes from a multiplexed to a non-multiplexed
domain or vice versa, as only an additional field needs to be added
to or removed from the packet.
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The main downside of this proposal is that it is very likely to
trigger a firewall response from any deep packet inspection device.
If the field is prefixed, the RTP fields are not matching the
heuristics field (unless the shim is designed to look like an RTP
header, in which case the payload length is unlikely to match the
expected value) and thus are likely preventing classification of the
packet as an RTP packet. If it is postfixed, it is likely classified
as an RTP packet but may not correctly validate if the content
validation is such that the payload length is expected to match
certain values. It is expected that a postfixed shim will be less
problematic than a prefixed shim in this regard, but we are lacking
hard data on this.
This solution's per packet overhead is 1 byte.
A.3. Single Session
Given the difficulty of multiplexing several RTP sessions onto a
single lower-layer transport, it's tempting to send multiple media
streams in a single RTP session. Doing this avoids the need to de-
multiplex several sessions on a single transport, but at the cost of
losing the RTP session as a separator for different type of streams.
Lacking different RTP sessions to demultiplex incoming packets, a
receiver will have to dig deeper into the packet before determining
what to do with it. Care must be taken in that inspection. For
example, you must be careful to ensure that each real media source
uses its own SSRC in the session and that this SSRC doesn't change
media type.
The loss of the RTP session as a separator for different usages or
purpose would be an minor issue if the only difference between the
RTP sessions is the media type. In this case, the application could
use the Payload Type field to identify the media type. The loss of
the RTP Session functionality is however severe, if the application
uses the RTP Session for separating different treatments, contexts
etc. Then you would need additional signalling to bind the different
sources to groups which can help make the necessary distinctions.
However, the loss of the RTP session as separator is not the only
issue with this approach. The RTP Multiplexing Architecture
[I-D.westerlund-avtcore-multiplex-architecture] discusses a number of
issues in Section 6.7. These include RTCP bandwidth differences,
limitations in the number of payload types, media aware RTP mixers
and interactions with Legacy end-points.
Additional attention should be place on this important aspect. In
multi-party situations using central nodes there exist some
difficulties in having a legacy implementation using multiple RTP
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sessions interworking with an end-point having only a single RTP
session across the central node. The main reason is the fact that
the one using single session with multiple media types has only one
SSRC space, while the other end-points have multiple spaces. Thus
translation may have to occur because there is several RTP sessions
using the same SSRC value. This has both limitations, processing
overhead and the possibility of becoming an deployment obstacle for
new RTP/RTCP extensions.
This approach has been proposed in the RTCWeb context in
[I-D.lennox-rtcweb-rtp-media-type-mux] and
[I-D.ietf-mmusic-sdp-bundle-negotiation]. These drafts describe how
to signal multiple media streams multiplexed into a single RTP
session, and address some of the issues raised here and in Section
6.7 of the RTP Multiplexing Architecture
[I-D.westerlund-avtcore-multiplex-architecture] draft.
This method has several limitations that limits its usage as solution
in providing multiple RTP sessions on the same lower layer transport.
However, we acknowledge that there are some uses for which this
method may be sufficient and which can accept the methods limitations
and downsides. The RTCWEB WG has a working assumption to support
this method. For more details of this method, see the relevant
drafts under development. We do include this method in the
comparison to provide a more complete picture of the pro and cons of
this method.
This solution has no per packet overhead. The signalling overhead
will be a different question.
A.4. Use the SRTP MKI field
This proposal is to overload the MKI SRTP/SRTCP identifier to not
only identify a particular crypto context, but also identify the
actual RTP Session. This clearly is a miss use of the MKI field,
however it appears to be with little negative implications. SRTP
already supports handling of multiple crypto contexts.
The two major downsides with this proposal is first the fact that it
requires using SRTP/SRTCP to multiplex multiple sessions on a single
lower layer transport. The second issue is that the session ID
parameter needs to be put into the various key-management schemes and
to make them understand that the reason to establish multiple crypto
contexts is because they are connected to various RTP Sessions.
Considering that SRTP have at least 3 used keying mechanisms, DTLS-
SRTP [RFC5764], Security Descriptions [RFC4568], and MIKEY [RFC3830],
this is not an insignificant amount of work.
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This solution has 32-bit per packet overhead, but only if the MKI was
not already used.
A.5. Use an Octet in the Padding
The basics of this proposal is to have the RTP packet and the last
(required by RFC3550) RTCP packet in a compound to include padding,
at least 2 bytes. One byte for the padding count (last byte) and one
byte just before the padding count containing the session ID.
This proposal uses bytes to carry the session ID that have no defined
value and is intended to be ignored by the receiver. From that
perspective it only causes packet expansion that is supported and
handled by all existing equipment. If an implementation fails to
understand that it is required to interpret this padding byte to
learn the session ID, it will see a mostly coherent RTP session
except where SSRCs overlap or where the payload types overlap.
However, reporting on the individual sources or forwarding the RTCP
RR are not completely without merit.
There is one downside of this proposal and that has to do with SRTP.
To be able to determine the crypto context, it is necessary to access
to the encrypted payload of the packet. Thus, the only mechanism
available for a receiver to solve this issue is to try the existing
crypto contexts for any session on the same lower layer transport and
then use the one where the packet decrypts and verifies correctly.
Thus for transport flows with many crypto contexts, an attacker could
simply generate packets that don't validate to force the receiver to
try all crypto contexts they have rather than immediately discard it
as not matching a context. A receiver can mitigate this somewhat by
using heuristics based on the RTP header fields to determine which
context applies for a received packet, but this is not a complete
solution.
This solution has a 16-bit per packet overhead.
A.6. Redefine the SSRC field
The Rosenberg et. al. Internet draft "Multiplexing of Real-Time
Transport Protocol (RTP) Traffic for Browser based Real-Time
Communications (RTC)" [I-D.rosenberg-rtcweb-rtpmux] proposed to
redefine the SSRC field. This has the advantage of no packet
expansion. It also looks like regular RTP. However, it has a number
of implications. First of all it prevents any RTP functionality that
require the same SSRC in multiple RTP sessions.
Secondly its interoperability with end-point using multiple RTP
sessions are problematic. Such interoperability will requires an
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SSRC translator function in the gatewaying node to ensure that the
SSRCs fulfill the semantic rules of the different domains. That
translator is actually far from easy as it needs to understand the
semantics of all RTP and RTCP extensions that include SSRC/CSRC.
This as it is necessary to know when a particular matching 32-bit
pattern is an SSRC field and when the field is just a combination of
other fields that create the same matching 32-bit pattern. Thus
there is a possibility that such a translator becomes a obstacle in
deploying future RTP/RTCP extensions. In addition the translator
actually have significant overhead when SRTP are in use. This as a
verification that the packet is authentic, decryption, SSRC
translation, encryption and finally generation of authentication tags
are required. In addition the translator must be part of the
security context.
This solution has no per packet overhead.
Appendix B. Comparison
This section compares the above potential solutions with the
requirements. Motivations are provided in addition to a high level
metric of successfully, partially and failing to meet requirement.
In the end a summary table (Figure 5) of the high level value are
provided.
B.1. Support of Multiple RTP Sessions Over Single Transport
This one is easy to determine. Only the single session proposal
fails this requirement as it is not at all designed to meet it. The
rest fully support this requirement. The main question around this
requirement is how important it is to have as discussed in
Section 4.1.
B.2. Enable Same SSRC Value in Multiple RTP Sessions
Based on the discussion in Section 4.2 two sub-requirements have been
derived.
B.2.1. Avoid SSRC Translation in Gateways/Translation
This sub-requirement is derived based on the desire to avoid having
gateways or translators perform full SSRC translation to minimize
complexity, avoid the requirement to have gateways in security
context, and as a hinder to long-term evolution. Two of the
proposals have issues with this, due to their lack of support for
multiple 32-bit SSRC spaces and lacking possibility to have the same
SSRC value in multiple RTP sessions. The proposals that have these
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properties and thus are marked as failing are the Single Session and
Redefine the SSRC field. The other proposals are all successful in
meeting this requirement.
B.2.2. Support Existing Extensions
The second sub-requirement is how well the proposals support using
the existing RTP mechanisms. Here both Single Session and Redefine
the SSRC field will have clear issues as they cannot support the same
full 32-bit SSRC value in two different RTP sessions. This is
clearly an issue for the XOR based FEC. RTP retransmission and
scalable encoding are minor issues as there exist alternatives to
those mechanisms that works with the structure of these two
proposals. Thus we give them a fail. The Header Extension gets a
partial due to unclear interaction between putting in an header
extension and these mechanisms.
B.3. Ensure SRTP Functions
This requirement is about ensuring both secure and efficient usage of
SRTP. The Octet in Padding field proposal gets a fail as the
receiving end-point cannot determine the intended RTP session prior
to de-encryption of the padding field. Thus a catch-22 arises which
can only be resolved by trying all session contexts and see what
decrypts. This causes a security vulnerability as an attacker can
inject a packet which does not meet any of the session contexts. The
receiver will then attempt decryption and authentication of it using
all its session contexts, increasing the amount of wasted resources
by a factor equal to the number of multiplexed sessions. Thus this
proposal gets a fail.
The proposal of Overloading the SRTP MKI field as session identifier
gets a partial due to the fact that it cannot use SRTP's key-
management mechanism out of the box. It forces the key-management
mechanism and the SRTP implementations to maintain the MKI-to-RTP
session bindings to maintain secure and correct function.
The Redefine the SSRC field gets a partial due to its need to modify
the key-management mechanisms to correctly identify the partial SSRC
space the parameters applies to. Similarly, the SRTP implementation
also needs to be updated to correctly support this security context
differentiation.
The header extension based solution gets a less severe partial than
Redefine the SSRC and the MKI. It will however have an issue when
being gatewayed to a domain that does not multiplex multiple RTP
sessions over the same transport. Then the gateway will require to
be in the security context to be able to add or remove the header
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extension as it is in the part of the packet that is integrity
protected by SRTP.
The remaining two proposals do not affect SRTP mechanisms and thus
successfully meet this requirement.
B.4. Don't Redefine Used Bits
This requirement is all about RTP and RTCP header fields having a
given definition should not be changed as it can cause
interoperability problems between modified and non-modified
implementations. This becomes especially problematic in RTP sessions
used for multi-party sessions.
Redefine the SSRC field gets a big fail on this as it redefines the
SSRC field, a core field in RTP. It has been identified that such a
change will have issues since if it gets connected to a non-modified
end-point that randomly assigns the SSRC, as supposed by RFC 3550,
those SSRCs will be distributed over different RTP sessions at the
modified end-point. Also other functions using the SSRC field, not
understanding the additional semantics of the SSRC field, is likely
to have issues.
Using the SRTP MKI field to identify a session is overloading that
field with double semantics. This likely has minimal negative impact
in RTP since it should be possible to have the SRTP stack use the MKI
field to both look up the security context and which output RTP
session the processed packet belongs to. However, this redefinition
clearly creates issues with the key-management scheme. That will
have to be modified to handle both this change and deal with the
interoperability issues when negotiating its usage. This gets a full
fail due to that it makes the problem someone else's, namely the RTP
implementors.
Defining an Octet in the Padding field redefines a field, whose
definition is to have zero value and is expected to be ignored by the
receiver according to the original semantics. Thus this is one of
the more benign modifications one can do, however this can still
cause issues in implementations that unnecessarily check the field
values, or in Firewalls. This is judged to be partially meeting the
requirement.
The Header Extension proposal does in fact not redefine any currently
used bits in RTP. The header extension would be a correctly
identified extension with its own definition. However, it does
redefine a rule on what header extensions are for. The RTCP solution
however would have more severe impact as it would need to redefine
the standard meaning of an RTCP packet header in addition to the
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default compound packet rules. Due to these issues the proposal
fails to meet this requirement.
The multiplexing shim and the single session both successfully meet
this requirement.
B.5. Firewall Friendly
This requirement is clearly difficult to judge as firewall
implementations are highly different in both implementation, scope of
what it investigates in packets, and set policies. A reasonable goal
is to minimize the likeliness that rules and policies intended to let
RTP media streams pass, will also let these streams through when
multiplexing RTP sessions over a single transport. The below
analysis shows that no solution is truly firewall friendly and all
are judged as being partially meeting this goal. However, the reason
why it is believed that a firewall might react to the streams are
quite different.
The Single Session and Redefine the SSRC field are likely the least
suspect solutions from a firewall perspective. However, as their
transport flows contain multiple SSRCs with payloads that indicate
likely multiple different media types they are still likely to make a
picky firewall block the transport. This is especially true for
Firewalls that take signalling messages into account where it will
expect a particular media type in a given context. A non upgraded
firewall might in fact produce two different contexts with
overlapping transport parameters where both rules will receive media
streams of the other media type that are outside of the allowed rule.
However, to be clear if these proposals doesn't get through, none of
the other will either as they all will have this behavior.
The header extension proposal is potentially problematic for two
reasons. The first reason, which also other proposals has, is
related to that the same SSRC value can exist in two RTP sessions
over the same underlying flow. Anyone tracking the sequence number
and timestamp will react badly as the second media stream with the
same SSRC causes constant jumps back and forth in these fields
compared to the first stream, if packets are transmitted
simultaneously for both SSRCs. This issue can likely only be solved
by having the Firewalls that like to track flows to also use the
session identifier to create context. This is possible as the header
extension will be in the clear and in the front. The second issue is
that the header extension itself may get the firewall to react.
Especially very picky ones that expect packets with certain media
types to have certain packet lengths. They are not compatible with a
header extension.
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The Multiplexing Shim shares the issue with multiple flows for the
same SSRC. Firewalls and deep packet inspection cause the shim
placement to be in question. If it is a pre-fixed shim, it prevents
the packet from looking like regular IP/UDP/RTP packets and be
correctly classified in Firewalls and DPI engines. However, if one
puts it last, it is unlikely that any firewall or DPI ever will be
able to take the session context into account as it is at the end of
the packet. This as many line rate processing devices only take a
certain amount of the headers into account.
The SRTP MKI field is likely the solution that has least firewall and
DPI issues, after the single RTP session. There is no additional
suspect field. The only difference from a single RTP session in the
transport flow is the fact that multiple MKI are guaranteed to be
used. However, that may occur also in a single RTP session usage.
Thus the only issues are the one shared with single session and the
one that several RTP media streams may use the same SSRC.
The octet in the padding field has, in addition to the issues the
SRTP MKI field has, the single issue that it redefines something that
is supposed to be zero into a value. Thus potentially causing a
deeply inspecting firewall to clamp the flow in fear of covert
channel or non-compliance.
B.6. Monitoring and Reporting
The monitoring and reporting requirement considers several aspects.
How useful monitoring can one get from an existing legacy monitor,
and secondary any issues in upgrading them to handle the selected
solution. Thirdly, packet selector filters and packet sniffers
concerns are considered.
In general one can expect the proposals that have only a single SSRC
space to work better with legacy. Thus both Single Session and
Redefine SSRC space can gather and report data on media flows most
likely. The only potential issue is that due to the different media
types and clock rates, some failure may occur. In particular a third
party monitor may be targeted to a specific media type, like
monitoring VoIP. That monitor will have problems processing any
video packets correctly and generate the VoIP specific metrics for
any video sending SSRC. In general, no legacy solution for
monitoring will be able to correctly create the sub-contexts that
each RTP session has in the solutions, without update to handle the
new semantics. Also when it comes to the packet filtering and
selector filters, fine grained control can only be accomplished
implementing the new semantics. Therefore only the Single Session
meets this requirement fully.
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Redefine the SSRC field is close to fully meeting the requirement,
however due to that there exist a session structure that is hidden to
anyone that is not upgraded to understand the semantics, this only
gets a partial.
The other proposals all can have multiple RTP sessions using the same
SSRC. This will create significant issues for any legacy third party
monitor. Only an updated monitor, or for that matter packet
selector, can pick out the individual media streams and their
associated RTCP traffic. Thus all these proposals gets a failure to
meet the requirement.
B.7. Usable over Multicast
As discussed earlier the goal with having the option usable also over
multicast is to remove the need to produce different media streams
for transport over unicast and multicast. All of the proposals
successfully meet the requirement.
B.8. Incremental Deployment
The possibility to deploy the usage of the multiplexing of multiple
RTP sessions over a single transport, especially in the context of
multi-party sessions, is a great benefit for any of the proposals.
Thus not all end-point implementations needs to be upgraded before
one start enabling it in the central node and any signalling.
Considering a centralized multi-party application where some
participants are using multiple transport flows and you want to
enable one particular participant to use the single transport to the
central node, one criteria stands out. The possibility to have one
RTP session per transport in one leg, and in the next multiplex them
together with minimal complexity and packet changes. Here there are
significant differences.
The Multiplexing Shim has the least overhead for this. As the
central node or gateway between deployments only needs to either add
or remove the shim identifier and then forward the packet over the
corresponding transport, either a joint one on the single transport
side, or over the individual one on the multiple transport side.
The SRTP MKI field proposal is almost as good, as the only main
difference is the need to coordinate the used MKIs on the non-
multiplexed legs so that there is no overlap between the RTP
sessions. And if there is, the MKI can be translated in gateway as
SRTP has no integrity protection over the MKI. Thus both
multiplexing shim and SRTP MKI field does successfully meet this
requirement.
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The Header Extension supports multiple full 32-bit SSRC spaces and
can thus handle all the RTP sessions without need for any SSRC
translation, however this proposal does run into the problem that the
gateway needs to be in the security context to be able to add or
remove the header extension when SRTP is used. In addition to the
security implications of that, there is a complexity overhead due to
the need to redo the authentication tags on all RTP/RTCP packets.
Thus it gets a partial.
The Octet in the Padding field share issues with the header extension
but have even higher complexities for this. The reason is that the
padding field is also encrypted. Thus to add or remove it (although
removing it may be unnecessary) forces the end-point to encrypt at
least that byte also, and for ciphers that are not stream-ciphers,
the whole packet needs to be re-encrypted. Thus this proposal gets a
very weak partially meeting the requirement.
The Single Session and Redefine the SSRC field do not allow several
vanilla RTP sessions to be connected to these proposals. The reason
is the single 32-bit SSRC space they have. Single Session only has
one session and the Redefine the SSRC fields uses some of the bits as
session identifier. This forces the gateway to translate the SSRC
whenever it does not fulfill the rules or semantics of the
multiplexed side. For Redefine SSRC field this becomes almost
constant as the session identifier part of the SSRC must be the same
over all SSRCs from the same session. For Single Session it may only
be needed when there otherwise would be an SSRC collision between the
sessions. This further assumes that the non-multiplexed side would
never use any of the RTP mechanisms that require the same SSRC in
multiple RTP sessions, as they cannot be gatewayed at all. When
translating an SSRC there is first of all an overhead, with SRTP that
includes a complete authenticate, decrypt, encrypt and create a new
authentication tag cycle. In addition, the SSRC translation could
potentially be a deployment obstacle for new RTP/RTCP extensions
required to be understood by the translator to be correctly
translated. Therefore these two proposals gets a fail to meet the
requirements.
B.9. Summary and Conclusion
This section contains a summary table of the high level outcome
against the different requirements.
A table mapping the requirements against the ID numbers used in the
table is the following:
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1: Support multiple RTP sessions over one transport flow
2: Enable same SSRC value in multiple RTP sessions
2.1: Avoid SSRC translation in gateways/translators
2.2: Support existing extensions
3: Ensure SRTP functions
4: Don't Redefine used bits
5: Firewall Friendly
6: Monitoring and Reporting should still function
7: Usable over Multicast
8: Incremental deployment
OH: Overhead in Bytes. + means variable
---------------+---+---+---+---+---+---+---+---+---+----
Solution | 1 |2.1|2.2| 3 | 4 | 5 | 6 | 7 | 8 | OH
---------------+---+---+---+---+---+---+---+---+---+----
Header Ext. | S | S | P | P | F | P | F | S | P | 8+
Multiplex Shim | S | S | S | S | S | P | F | S | S | 1
Single Session | F | F | F | S | S | P | S | S | F | 0
SRTP MKI Field | S | S | S | P | F | P | F | S | S | 4
Padding Field | S | S | S | F | P | P | F | S | P | 2
Redefine SSRC | S | F | F | P | F | P | P | S | S | 0
---------------+---+---+---+---+---+---+---+---+---+----
Figure 5: Summary Table of Evaluation (Successfully (S), Partially
(P) or Fails (F) to meet requirement)
Considering these options, the authors would recommend that AVTCORE
standardize a solution based on a post or prefixed multiplexing
field, i.e. a shim approach combined with the appropriate signalling
as described in Appendix A.2.
Westerlund & Perkins Expires April 25, 2013 [Page 41]
Internet-Draft Multiple RTP Session on Single Transport October 2012
Authors' Addresses
Magnus Westerlund
Ericsson
Farogatan 6
SE-164 80 Kista
Sweden
Phone: +46 10 714 82 87
Email: magnus.westerlund@ericsson.com
Colin Perkins
University of Glasgow
School of Computing Science
Glasgow G12 8QQ
United Kingdom
Email: csp@csperkins.org
Westerlund & Perkins Expires April 25, 2013 [Page 42]