Network Working Group M. Westerlund
Internet-Draft B. Burman
Intended status: Best Current Practice Ericsson
Expires: April 26, 2012 C. . Perkins
University of Glasgow
October 24, 2011

RTP Multiplexing Architecture


RTP has always been a protocol that supports multiple participants each sending their own media streams in an RTP session. Thus relying on the three main multiplexing points in RTP; RTP session, SSRC and Payload Type for their various needs. However, most usages of RTP have been less complex often with a single SSRC in each direction, with a single RTP session per media type. But the more complex usages start to be more common and thus guidance on how to use RTP in various complex cases are needed. This document analyzes a number of cases and discusses the usage of the various multiplexing points and the need for functionality when defining RTP/RTCP extensions that utilize multiple RTP streams and multiple RTP sessions.

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 working documents as Internet-Drafts. The list of current Internet- Drafts is at

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 26, 2012.

Copyright Notice

Copyright (c) 2011 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 ( 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 the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License.

Table of Contents

1. Introduction

This document focuses at issues of non-basic usage of RTP [RFC3550] where multiple media sources of the same media type are sent over RTP. Separation of different media types is another issue that will be discussed in this document. The intended uses include for example multiple sources from the same end-point, multiple streams from a single media source, multiple end-points each having a source, or an application that needs multiple representations (encodings) of a particular source. It will be shown that these uses are inter-related and need a common discussion to ensure consistency. In general, usage of the RTP session and media streams will be discussed in detail.

RTP is already designed for multiple participants in a communication session. This is not restricted to multicast, as many believe, but also provides functionality over unicast, using either multiple transport flows below RTP or a network node that re-distributes the RTP packets. The node can for example be a transport translator (relay) that forwards the packets unchanged, a translator performing media translation in addition to forwarding, or an RTP mixer that creates new conceptual sources from the received streams. In addition, multiple streams may occur when a single end-point have multiple media sources of the same media type, like multiple cameras or microphones that need to be sent simultaneously.

Historically, the most common RTP use cases have been point to point Voice over IP (VoIP) or streaming applications, commonly with no more than one media source per end-point and media type (typically audio and video). Even in conferencing applications, especially voice only, the conference focus or bridge has provided a single stream with a mix of the other participants to each participant. It is also common to have individual RTP sessions between each end-point and the RTP mixer.

SSRC is the RTP media stream identifier that helps to uniquely identify media sources in RTP sessions. Even though available SSRC space can theoretically handle more than 4 billion simultaneous sources, the perceived need for handling multiple SSRCs in implementations has been small. This has resulted in an installed legacy base that isn't fully RTP specification compliant and will have different issues if they receive multiple SSRCs of media, either simultaneously or in sequence. These issues will manifest themselves in various ways, either by software crashes or simply in limited functionality, like only decoding and playing back the first or latest received SSRC and discarding media related to any other SSRCs.

There have also arisen various cases where multiple SSRCs are used to represent different aspects of what is in fact a single underlying media source. A very basic case is RTP retransmission [RFC4588] which have one SSRC for the original stream, and a second SSRC either in the same session or in a different session to represent the retransmitted packets to ensure that the monitoring functions still function. Another use case is scalable encoding, such as the RTP payload format for Scalable Video Coding (SVC) [RFC6190], which has an operation mode named Multiple Session Transmission (MST) that uses one SSRC in each RTP session to send one or more scalability layers. A third example is simulcast where a single media source is encoded in different versions and transmitted to an RTP mixer that picks which version to actually distribute to a given receiver part of the RTP session.

This situation has created a need for a document that discusses the existing possibilities in the RTP protocol and how these can and should be used in applications. A new set of applications needing more advanced functionalities from RTP is also emerging on the market, such as telepresence and advanced video conferencing. Thus furthering the need for a more common understanding of how multiple streams are handled in RTP to ensure media plane interoperability.

The document starts with some definitions and then goes into the existing RTP functionalities around multiplexing. Both the desired behavior and the implications of a particular behavior depend on which topologies are used, which requires some consideration. This is followed by a discussion of some choices in multiplexing behavior and their impacts. Finally, some recommendations and examples are provided.

2. Definitions

2.1. 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].

2.2. Terminology

The following terms and abbreviations are used in this document:

A single entity sending or receiving RTP packets. It may be decomposed into several functional blocks, but as long as it behaves a single RTP stack entity it is classified as a single end-point.
Media Stream:
A sequence of RTP packets using a single SSRC that together carry part or all of the content of a specific Media Type from a specific sender source within a given RTP session.
Media Aggregate:
All Media Streams related to a particular Source.
Media Type:
Audio, video, text or data whose form and meaning are defined by a specific real-time application.
The source of a particular media stream. Either a single media capturing device such as a video camera, or a microphone, or a specific output of a media production function, such as an audio mixer, or some video editing function.

3. RTP Multiplex Points

This section describes the existing RTP tools that enable multiplexing of different media streams and RTP functionalities.

3.1. Session

The RTP Session is the highest semantic level in RTP and contains all of the RTP functionality.

RTP in itself does not contain any Session identifier, but relies on the underlying transport. For example, when running RTP on top of UDP, an RTP endpoint can identify and delimit an RTP Session from other RTP Sessions through the UDP source and destination transport address, consisting of network address and port number(s). Most commonly only the destination address, i.e. all traffic received on a particular port, is defined as belonging to a specific RTP Session. It is worth noting that in practice a more narrow definition of the transport flows that are related to a give RTP session is possible. An RTP session can for example be defined as one or more 5-tuples (Transport Protocol, Source Address, Source Port, Destination Address, Destination Port). Any set of identifiers of RTP and RTCP packet flows are sufficient to determine if the flow belongs to a particular session or not.

Commonly, RTP and RTCP use separate ports and the destination transport address is in fact an address pair, but in the case of RTP/RTCP multiplex [RFC5761] there is only a single port.

A source that changes its source transport address during a session must also choose a new SSRC identifier to avoid being interpreted as a looped source.

The set of participants considered part of the same RTP Session is defined by[RFC3550] as those that share a single SSRC space. That is, those participants that can see an SSRC identifier transmitted by any one of the other participants. A participant can receive an SSRC either as SSRC or CSRC in RTP and RTCP packets. Thus, the RTP Session scope is decided by the participants' network interconnection topology, in combination with RTP and RTCP forwarding strategies deployed by end-points and any interconnecting middle nodes.

3.2. SSRC

The Synchronization Source (SSRC) identifier is used to identify individual sources within an RTP Session. The SSRC number is globally unique within an RTP Session and all RTP implementations must be prepared to use procedures for SSRC collision handling, which results in an SSRC number change. The SSRC number is randomly chosen, carried in every RTP packet header and is not dependent on network address. SSRC is also used as identifier to refer to separate media streams in RTCP.

A media source having an SSRC identifier can be of different types:

Connected to a "physical" media source, for example a camera or microphone.
A source with some attributed property generated by some network node, for example a filtering function in an RTP mixer that provides the most active speaker based on some criteria, or a mix representing a set of other sources.
A source that does not generate any RTP media stream in itself (e.g. an end-point only receiving in an RTP session), but anyway need a sender SSRC for use as source in RTCP reports.

Note that a "multimedia source" that generates more than one media type, e.g. a conference participant sending both audio and video, need not (and commonly should not) use the same SSRC value across RTP sessions. RTCP Compound packets containing the CNAME SDES item is the designated method to bind an SSRC to a CNAME, effectively cross-correlating SSRCs within and between RTP Sessions as coming from the same end-point. The main property attributed to SSRCs associated with the same CNAME is that they are from a particular synchronization context and may be synchronized at playback. There exist a few other methods to relate different SSRC where use of CNAME is inappropriate, such as session-based RTP retransmission [RFC4588].

Note also that RTP sequence number and RTP timestamp are scoped by SSRC and thus independent between different SSRCs.

An RTP receiver receiving a previously unseen SSRC value must interpret it as a new source. It may in fact be a previously existing source that had to change SSRC number due to an SSRC conflict. However, the originator of the previous SSRC should have ended the conflicting source by sending an RTCP BYE for it prior to starting to send with the new SSRC, so the new SSRC is anyway effectively a new source.

Some RTP extension mechanisms already require the RTP stacks to handle additional SSRCs, like SSRC multiplexed RTP retransmission [RFC4588]. However, that still only requires handling a single media decoding chain per pair of SSRCs.

3.3. CSRC

The Contributing Source (CSRC) can arguably be seen as a sub-part of a specific SSRC and thus a multiplexing point. It is optionally included in the RTP header, shares the SSRC number space and specifies which set of SSRCs that has contributed to the RTP payload. However, even though each RTP packet and SSRC can be tagged with the contained CSRCs, the media representation of an individual CSRC is in general not possible to extract from the RTP payload since it is typically the result of a media mixing (merge) operation (by an RTP mixer) on the individual media streams corresponding to the CSRC identifiers. Due to these restrictions, CSRC will not be considered a fully qualified multiplex point and will be disregarded in the rest of this document.

3.4. Payload Type

The Payload Type number is also carried in every RTP packet header and identifies what format the RTP payload has. The term "format" here includes whatever can be described by out-of-band signaling means for dynamic payload types, as well as the statically allocated payload types in [RFC3551]. In SDP the term "format" includes media type, RTP timestamp sampling rate, codec, codec configuration, payload format configurations, and various robustness mechanisms such as redundant encodings [RFC2198].

The meaning of a Payload Type definition (the number) is re-used between all media streams within an RTP session, when the definition is either static or signaled through SDP. There however do exist cases where each end-point have different sets of payload types due to SDP offer/answer.

Although Payload Type definitions are commonly local to an RTP Session, there are some uses where Payload Type numbers need be unique across RTP Sessions. This is for example the case in Media Decoding Dependency [RFC5583] where Payload Types are used to describe media dependency across RTP Sessions.

Given that multiple Payload Types are defined in an RTP Session, a media sender is free to change the Payload Type on a per packet basis. One example of designed per-packet change of Payload Type is a speech codec that makes use of generic Comfort Noise [RFC3389].

The RTP Payload Type in RTP is designed such that only a single Payload Type is valid at any time instant in the SSRC's timestamp time line, effectively time-multiplexing different Payload Types if any switch occurs. Even when this constraint is met, having different rates on the RTP timestamp clock for the RTP Payload Types in use in the same RTP Session have issues such as loss of synchronization. Payload Type clock rate switching requires some special consideration that is described in the multiple clock rates specification [I-D.ietf-avtext-multiple-clock-rates].

If there is a true need to send multiple Payload Types for the same SSRC that are valid for the same RTP Timestamps, then redundant encodings [RFC2198] can be used. Several additional constraints than the ones mentioned above need to be met to enable this use, one of which are that the combined payload sizes of the different Payload Types must not exceed the transport MTU.

Other aspects of RTP payload format use are described in RTP Payload HowTo [I-D.ietf-payload-rtp-howto].

4. Multiple Streams Alternatives

This section reviews the alternatives to enable multi-stream handling. Let's start with describing mechanisms that could enable multiple media streams, independent of the purpose for having multiple streams.

below section [sec-pt-mux]. The RTP payload type alone is not suitable for cases where additional media streams are required. Streams need their own SSRCs, so that they get their own sequence number space. The SSRC itself is also important so that the media stream can be referenced and reported on.

SSRC Multiplexing:
Each additional Media Stream gets its own SSRC within a RTP Session.
Session Multiplexing:
Using additional RTP Sessions to handle additional Media Streams
Payload Type Multiplexing:
Using different RTP payload types for different additional streams.

Independent of the reason to use additional media streams, achieving it using payload type multiplexing is not a good choice as can be seen in the

This leaves us with two choices, either using SSRC multiplexing to have multiple SSRCs from one end-point in one RTP session, or create additional RTP sessions to hold that additional SSRC. As the below discussion will show, in reality we cannot choose a single one of the two solutions. To utilize RTP well and as efficiently as possible, both are needed. The real issue is finding the right guidance on when to create RTP sessions and when additional SSRCs in an RTP session is the right choice.

In the below discussion, please keep in mind that the reasons for having multiple media streams vary and include but are not limited to the following:

  • Multiple Media Sources of the same media type
  • Retransmission streams
  • FEC stream
  • Alternative Encoding
  • Scalability layer

Thus the choice made due to one reason may not be the choice suitable for another reason. In the above list, the different items have different levels of maturity in the discussion on how to solve them. The clearest understanding is associated with multiple media sources of the same media type. However, all warrant discussion and clarification on how to deal with them.

5. RTP Topologies and Issues

The impact of how RTP Multiplex is performed will in general vary with how the RTP Session participants are interconnected; the RTP Topology [RFC5117]. This section describes the topologies and attempts to highlight the important behaviors concerning RTP multiplexing and multi-stream handling. It lists any identified issues regarding RTP and RTCP handling, and introduces additional topologies that are supported by RTP beyond those included in RTP Topologies [RFC5117]. The RTP Topologies that do not follow the RTP specification or do not attempt to utilize the facilities of RTP are ignored in this document.

5.1. Point to Point

This is the most basic use case with an RTP session containing of two end-points. Each end-point has one or more SSRCs.

+---+         +---+
| A |<------->| B |
+---+         +---+

5.1.1. RTCP Reporting

In cases when an end-point uses multiple SSRCs, we have found two closely related issues. The first is if every SSRC shall report on all other SSRC, even the ones originating from the same end-point. The reason for this would be ensure that no monitoring function should suspect a breakage in the RTP session.

The second issue around RTCP reporting arise when an end-point receives one or more media streams, and when the receiving end-point itself sends multiple SSRC in the same RTP session. As transport statistics are gathered per end-point and shared between the nodes, all the end-point's SSRC will report based on the same received data, the only difference will be which SSRCs sends the report. This could be considered unnecessary overhead, but for consistency it might be simplest to always have all sending SSRCs send RTCP reports on all media streams the end-point receives.

The current RTP text is silent about sending RTCP Receiver Reports for an endpoint's own sources, but does not preclude either sending or omitting them. The uncertainty in the expected behavior in those cases have likely caused variations in the implementation strategy. This could cause an interoperability issue where it is not possible to determine if the lack of reports are a true transport issue, or simply a result of implementation.

Although this issue is valid already for the simple point to point case, it needs to be considered in all topologies. From the perspective of an end-point, any solution needs to take into account what a particular end-point can determine without explicit information of the topology. For example, a Transport Translator (Relay) topology will look quite similar as point to point on an RTP level but is different. The main difference between a point to point with two SSRC being sent from the remote end-point and a Transport Translator with two single SSRC remote clients are that the RTT may vary between the SSRCs (but it is not guaranteed), and that the SSRCs may have different CNAMEs.

5.1.2. Compound RTCP Packets

When an end-point has multiple SSRCs and it needs to send RTCP packets on behalf of these SSRCs, the question arises if and how RTCP packets with different source SSRCs can be sent in the same compound packet. If it is allowed, then some consideration of the transmission scheduling is needed.

5.2. Point to Multipoint Using Multicast

This section discusses the Point to Multi-point using Multicast to interconnect the session participants. This needs to consider both Any Source Multicast (ASM) and Source-Specific Multicast (SSM).

+---+     /       \    +---+ 
| A |----/         \---| B |
+---+   /   Multi-  \  +---+
       +    Cast     +      
+---+   \  Network  /  +---+
| C |----\         /---| D |
+---+     \       /    +---+

In Any Source Multicast, any of the participants can send to all the other participants, simply by sending a packet to the multicast group. That is not possible in Source Specific Multicast [RFC4607] where only a single source (Distribution Source) can send to the multicast group, creating a topology that looks like the one below:

+--------+       +-----+          Multicast
|Media   |       |     |     +----------------> R(1)
|Sender 1|<----->| D S |     |                    |
+--------+       | I O |  +--+                    |
                 | S U |  |  |                    |
+--------+       | T R |  |  +-----------> R(2)   |
|Media   |<----->| R C |->+  +---- :         |    |
|Sender 2|       | I E |  |  +------> R(n-1) |    |
+--------+       | B   |  |  |          |    |    |
    :            | U   |  +--+--> R(n)  |    |    |
    :            | T +-|          |     |    |    |
                 | I | |<---------+     |    |    |
+--------+       | O |F|<---------------+    |    |
|Media   |       | N |T|<--------------------+    |
|Sender M|<----->|   | |<-------------------------+
+--------+       +-----+            Unicast

FT = Feedback Target
Transport from the Feedback Target to the Distribution
Source is via unicast or multicast RTCP if they are not

In this topology a number of Media Senders (1 to M) are allowed to send media to the SSM group, sends media to the distribution source which then forwards the media streams to the multicast group. The media streams reach the Receivers (R(1) to R(n)). The Receiver's RTCP cannot be sent to the multicast group. To support RTCP, an RTP extension for SSM [RFC5760] was defined that use unicast transmission to send RTCP from the receivers to one or more Feedback Targets (FT).

As multicast is a one to many distribution system this must be taken into consideration. For example, the only practical method for adapting the bit-rate sent towards a given receiver is to use a set of multicast groups, where each multicast group represents a particular bit-rate. The media encoding is either scalable, where multiple layers can be combined, or simulcast where a single version is selected. By either selecting or combing multicast groups, the receiver can control the bit-rate sent on the path to itself. It is also common that transport robustification is sent in its own multicast group to allow for interworking with legacy or to support different levels of protection.

The result of this is three common behaviors for RTP multicast:

  1. Use of multiple RTP sessions for the same media type.
  2. The need for identifying RTP sessions that are related in one of several ways.
  3. The need for binding related SSRCs in different RTP sessions together.

This indicates that Multicast is an important consideration when working with the RTP multiplexing and multi stream architecture questions. It is also important to note that so far there is no special mode for basic behavior between multicast and unicast usages of RTP. Yes, there are extensions targeted to deal with multicast specific cases but the general applicability does need to be considered.

5.3. Point to Multipoint Using an RTP Translator

Transport Translators (Relays) are a very important consideration for this document as they result in an RTP session situation that is very similar to how an ASM group RTP session would behave.

+---+      +------------+      +---+
| A |<---->|            |<---->| B |
+---+      |            |      +---+
           | Translator |           
+---+      |            |      +---+
| C |<---->|            |<---->| D |
+---+      +------------+      +---+

One of the most important aspects with the simple relay is that it is both easy to implement and require minimal amount of resources as only transport headers are rewritten, no RTP modifications nor media transcoding occur. Thus it is most likely the cheapest and most generally deployable method for multi-point sessions. The most obvious downside of this basic relaying is that the translator has no control over how many streams needs to be delivered to a receiver. Nor can it simply select to deliver only certain streams, at least not without new RTCP extensions to coherently handle the fact that some middlebox temporarily stops a stream, preventing some receivers from reporting on it. This consistency problem in RTCP reporting needs to be handled.

The Transport Translator does not need to have an SSRC of itself, nor need it send any RTCP reports on the flows that passes it, but it may choose to do that.

Use of a transport translator results in that any of the end-points will receive multiple SSRCs over a single unicast transport flow from the translator. That is independent of the other end-points having only a single or several SSRCs. End-points that have multiple SSRCs put further requirements on how SSRCs can be related or bound within and across RTP sessions and how they can be identified on an application level.

A Media Translator can perform a large variety of media functions affecting the media stream passing the translator, coming from one source and destined to a particular end-point. The media stream can be transcoded to a different bit-rate, change to another encoder, change the packetization of the media stream, add FEC streams, or terminate RTP retransmissions. The latter behaviors require the translator to use SSRCs that only exist in a particular sub-domain of the RTP session, and it may also create additional sessions when the mechanism applied on one side so requires.

5.4. Point to Multipoint Using an RTP Mixer

The most commonly used topology in centralized conferencing is based on the RTP Mixer. The main reason for this is that it provides a very consistent view of the RTP session towards each participant. That is accomplished through the mixer having its own SSRCs and any media sent to the participants will be sent using those SSRCs. If the mixer wants to identify the underlying media sources for its conceptual streams, it can identify them using CSRC. The media stream the mixer provides can be an actual media mixing of multiple media sources. It might also be as simple as selecting one of the underlying sources based on some mixer policy or control signalling.

+---+      +------------+      +---+
| A |<---->|            |<---->| B |
+---+      |            |      +---+
           |   Mixer    |           
+---+      |            |      +---+
| C |<---->|            |<---->| D |
+---+      +------------+      +---+

In the case where the mixer does stream selection, an application may in fact desire multiple simultaneous streams but only as many as the mixer can handle. As long as the mixer and an end-point can agree on the maximum number of streams and how the streams that are delivered are selected, this provides very good functionality. As these streams are forwarded using the mixer's SSRCs, there are no inconsistencies within the session.

5.5. Point to Multipoint using Multiple Unicast flows

Based on the RTP session definition, it is clearly possible to have a joint RTP session over multiple transport flows like the below three end-point joint session. In this case, A needs to send its' media streams and RTCP packets to both B and C over their respective transport flows. As long as all participants do the same, everyone will have a joint view of the RTP session.

+---+      +---+
| A |<---->| B |
+---+      +---+
  ^         ^   
   \       /    
    \     /     
     v   v      
     | C |      

This doesn't create any additional requirements beyond the need to have multiple transport flows associated with a single RTP session. Note that an end-point may use a single local port to receive all these transport flows, or it might have separate local reception ports for each of the end-points.

5.6. Decomposited End-Point

There is some possibility that an RTP end-point implementation in fact reside on multiple devices, each with their own network address. A very basic use case for this would be to separate audio and video processing for a particular end-point, like a conference room, into one device handling the audio and another handling the video being interconnected by some control functions allowing them to behave as a single end-point.

| End-point A         |
| Local Area Network  |
|      +------------+ |
|   +->| Audio      |<+----\
|   |  +------------+ |     \    +------+
|   |  +------------+ |      +-->|      |
|   +->| Video      |<+--------->|  B   |
|   |  +------------+ |      +-->|      |
|   |  +------------+ |     /    +------+
|   +->| Control    |<+----/
|      +------------+ |

In the above usage, let us assume that the RTP sessions are different for audio and video. The audio and video parts will use a common CNAME and also have a common clock to ensure that synchronization and clock drift handling works despite the decomposition. However, if the audio and video were in a single RTP session then this use case becomes problematic. This as all transport flow receivers will need to receive all the other media streams that are part of the session. Thus the audio component will receive also all the video media streams, while the video component will receive all the audio ones, thus doubling the site's bandwidth requirements from all other session participants. With a joint RTP session it also becomes evident that a given end-point, as interpreted from a CNAME perspective, has two sets of transport flows for receiving the streams and the decomposition isn't hidden.

The requirements that can derived from the above usage is that the transport flows for each RTP session might be under common control but still go to what looks like different end-points based on addresses and ports. A conclusion can also be reached that decomposition without using separate RTP sessions has downsides and potential for RTP/RTCP issues.

There exist another use case which might be considered as a decomposited end-point. However, as will be shown this should be considered a translator instead. An example of this is when an end-point A sends a media flow to B. On the path there is a device C that on A's behalf does something with the media streams, for example adds an RTP session with FEC information for A's media streams. C will in this case need to bind the new FEC streams to A's media stream by using the same CNAME as A.

+------+        +------+         +------+
|      |        |      |         |      |
|  A   |------->|  C   |-------->|  B   |
|      |        |      |---FEC-->|      |
+------+        +------+         +------+

This type of functionality where C does something with the media stream on behalf of A is clearly covered under the media translator definition [sec-translator].

6. Dismissing Payload Type Multiplexing

Before starting a discussion on when to use what alternative, we will first document a number of reasons why using the payload type as a multiplexing point for anything related to multiple streams is unsuitable and will not be considered further.

If one attempts to use Payload type multiplexing beyond it's defined usage, that has well known negative effects on RTP. To use Payload type as the single discriminator for multiple streams implies that all the different media streams are being sent with the same SSRC, thus using the same timestamp and sequence number space. This has many effects:

  1. Putting restraint on RTP timestamp rate for the multiplexed media. For example, media streams that use different RTP timestamp rates cannot be combined, as the timestamp values need to be consistent across all multiplexed media frames. Thus streams are forced to use the same rate. When this is not possible, Payload Type multiplexing cannot be used.
  2. Many RTP payload formats may fragment a media object over multiple packets, like parts of a video frame. These payload formats need to determine the order of the fragments to correctly decode them. Thus it is important to ensure that all fragments related to a frame or a similar media object are transmitted in sequence and without interruptions within the object. This can relatively simple be solved on the sender side by ensuring that the fragments of each media stream are sent in sequence.
  3. Some media formats require uninterrupted sequence number space between media parts. These are media formats where any missing RTP sequence number will result in decoding failure or invoking of a repair mechanism within a single media context. The text/T140 payload format [RFC4103] is an example of such a format. These formats will need a sequence numbering abstraction function between RTP and the individual media stream before being used with Payload Type multiplexing.
  4. Sending multiple streams in the same sequence number space makes it impossible to determine which Payload Type and thus which stream a packet loss relates to.
  5. If RTP Retransmission [RFC4588] is used and there is a loss, it is possible to ask for the missing packet(s) by SSRC and sequence number, not by Payload Type. If only some of the Payload Type multiplexed streams are of interest, there is no way of telling which missing packet(s) belong to the interesting stream(s) and all lost packets must be requested, wasting bandwidth.
  6. The current RTCP feedback mechanisms are built around providing feedback on media streams based on stream ID (SSRC), packet (sequence numbers) and time interval (RTP Timestamps). There is almost never a field to indicate which Payload Type is reported, so sending feedback for a specific media stream is difficult without extending existing RTCP reporting.
  7. The current RTCP media control messages [RFC5104] specification is oriented around controlling particular media flows, i.e. requests are done addressing a particular SSRC. Such mechanisms would need to be redefined to support Payload Type multiplexing.
  8. The number of payload types are inherently limited. Accordingly, using Payload Type multiplexing limits the number of streams that can be multiplexed and does not scale. This limitation is exacerbated if one uses solutions like RTP and RTCP multiplexing [RFC5761] where a number of payload types are blocked due to the overlap between RTP and RTCP.
  9. At times, there is a need to group multiplexed streams and this is currently possible for RTP Sessions and for SSRC, but there is no defined way to group Payload Types.
  10. It is currently not possible to signal bandwidth requirements per media stream when using Payload Type Multiplexing.
  11. Most existing SDP media level attributes cannot be applied on a per Payload Type level and would require re-definition in that context.
  12. A legacy end-point that doesn't understand the indication that different RTP payload types are different media streams may be slightly confused by the large amount of possibly overlapping or identically defined RTP Payload Types.

7. Multiple Streams Discussion

7.1. Introduction

Using multiple media streams is a well supported feature of RTP. However, what can be unclear for most implementors or people writing RTP/RTCP extensions attempting to apply multiple streams, is when it is most appropriate to add an additional SSRC in an existing RTP session and when it is better to use multiple RTP sessions. This section tries to discuss the various considerations needed. The next section then concludes with some guidelines.

7.2. RTP/RTCP Aspects

This section discusses RTP and RTCP aspects worth considering when selecting between SSRC multiplexing and Session multiplexing.

7.2.1. The RTP Specification

RFC 3550 contains some recommendations and a bullet list with 5 arguments for different aspects of RTP multiplexing. Let's review Section 5.2 of [RFC3550], reproduced below:

"For efficient protocol processing, the number of multiplexing points should be minimized, as described in the integrated layer processing design principle [ALF]. In RTP, multiplexing is provided by the destination transport address (network address and port number) which is different for each RTP session. For example, in a teleconference composed of audio and video media encoded separately, each medium SHOULD be carried in a separate RTP session with its own destination transport address.

Separate audio and video streams SHOULD NOT be carried in a single RTP session and demultiplexed based on the payload type or SSRC fields. Interleaving packets with different RTP media types but using the same SSRC would introduce several problems:

  1. If, say, two audio streams shared the same RTP session and the same SSRC value, and one were to change encodings and thus acquire a different RTP payload type, there would be no general way of identifying which stream had changed encodings.
  2. An SSRC is defined to identify a single timing and sequence number space. Interleaving multiple payload types would require different timing spaces if the media clock rates differ and would require different sequence number spaces to tell which payload type suffered packet loss.
  3. The RTCP sender and receiver reports (see Section 6.4) can only describe one timing and sequence number space per SSRC and do not carry a payload type field.
  4. An RTP mixer would not be able to combine interleaved streams of incompatible media into one stream.
  5. Carrying multiple media in one RTP session precludes: the use of different network paths or network resource allocations if appropriate; reception of a subset of the media if desired, for example just audio if video would exceed the available bandwidth; and receiver implementations that use separate processes for the different media, whereas using separate RTP sessions permits either single- or multiple-process implementations.

Using a different SSRC for each medium but sending them in the same RTP session would avoid the first three problems but not the last two.

On the other hand, multiplexing multiple related sources of the same medium in one RTP session using different SSRC values is the norm for multicast sessions. The problems listed above don't apply: an RTP mixer can combine multiple audio sources, for example, and the same treatment is applicable for all of them. It may also be appropriate to multiplex streams of the same medium using different SSRC values in other scenarios where the last two problems do not apply."

Let's consider one argument at a time. The first is an argument for using different SSRC for each individual media stream, which still is very applicable.

The second argument is advocating against using payload type multiplexing, which still stands as can been seen by the extensive list of issues found in Section 6.

The third argument is yet another argument against payload type multiplexing.

The fourth is an argument against multiplexing media streams that require different handling into the same session. This is to simplify the processing at any receiver of the media stream. If all media streams that exist in an RTP session is of one media type and one particular purpose, there is no need for deeper inspection of the packets before processing them in both end-points and RTP aware middle nodes.

The fifth argument discusses network aspects that we will discuss more below in Section 7.4. It also goes into aspects of implementation, like decomposed end-points where different processes or inter-connected devices handle different aspects of the whole multi-media session.

A summary of RFC 3550's view on multiplexing is to use unique SSRCs for anything that is its' own media/packet stream, and secondly use different RTP sessions for media streams that don't share media type and purpose, to maximize flexibility when it comes to processing and handling of the media streams.

This mostly agrees with the discussion and recommendations in this document. However, there has been an evolution of RTP since that text was written which needs to be reflected in the discussion. Additional clarifications for specific cases are also needed.

7.2.2. Multiple SSRC Legacy Considerations

When establishing RTP sessions that may contain end-points that aren't updated to handle multiple streams following these recommendations, a particular application can have issues with multiple SSRCs within a single session. These issues include:

  1. Need to handle more than one stream simultaneously rather than replacing an already existing stream with a new one.
  2. Be capable of decoding multiple streams simultaneously.
  3. Be capable of rendering multiple streams simultaneously.

RTP Session multiplexing could potentially avoid these issues if there is only a single SSRC at each end-point, and in topologies which appears like point to point as seen the end-point. However, forcing the usage of session multiplexing due to this reason would be a great mistake, as it is likely that a significant set of applications will need a combination of SSRC multiplexing of several media sources and session multiplexing for other aspects such as encoding alternatives, robustification or simply to support legacy. However, this issue does need consideration when deploying multiple media streams within an RTP session where legacy end-points may occur.

7.2.3. RTP Specification Clarifications Needed

The RTP specification contains a few things that are potential interoperability issues when using multiple SSRCs within a session. These issues are described and discussed in Section 9. These should not be considered strong arguments against using SSRC multiplexing when otherwise appropriate, and there are some issues we expect to be solved in the near future.

7.2.4. Handling Varying sets of Senders

Another potential issue that needs to be considered is where a limited set of simultaneously active sources varies within a larger set of session members. As each media decoding chain may contain state, it is important that this type of usage ensures that a receiver can flush a decoding state for an inactive source and if that source becomes active again, it does not assume that this previous state exists.

This behavior might in certain applications be possible to limit to a particular RTP Session and instead use multiple RTP sessions. But in some cases it is likely unavoidable and the most appropriate thing is to SSRC multiplex.

7.2.5. Cross Session RTCP requests

There currently exist no functionality to make truly synchronized and atomic RTCP requests across multiple RTP Sessions. Instead separate RTCP messages will have to be sent in each session. This gives SSRC multiplexed streams a slight advantage as RTCP requests for different streams in the same session can be sent in a compound RTCP packet. Thus providing an atomic operation if different modifications of different streams are requested at the same time.

In Session multiplexed cases, the RTCP timing rules in the sessions and the transport aspects, such as packet loss and jitter, prevents a receiver from relying on atomic operations, instead more robust and forgiving mechanisms need to be used.

7.2.6. Binding Related Sources

A common problem in a number of various RTP extensions has been how to bind together related sources. This issue is common independent of SSRC multiplexing and Session Multiplexing, and any solution and recommendation to the problem should work equally well for both to avoid creating barriers between using session multiplexing and SSRC multiplexing.

The current solutions don't have these properties. There exist one solution for grouping RTP session together in SDP [RFC5888] to know which RTP session contains for example the FEC data for the source data in another session. However, this mechanism does not work on individual media flows and is thus not directly applicable to the problem. The other solution is also SDP based and can group SSRCs within a single RTP session [RFC5576]. Thus this mechanism can bind media streams in SSRC multiplexed cases. Both solutions have the shortcoming of being restricted to SDP based signalling and also do not work in cases where the session's dynamic properties are such that it is difficult or resource consuming to keep the list of related SSRCs up to date.

One possible solution could be to mandate the same SSRC being used in all RTP session in case of session multiplexing. We do note that Section 8.3 of the RTP Specification [RFC3550] recommends using a single SSRC space across all RTP sessions for layered coding. However this recommendation has some downsides and is less applicable beyond the field of layered coding. To use the same sender SSRC in all RTP sessions from a particular end-point can cause issues if an SSRC collision occurs. If the same SSRC is used as the required binding between the streams, then all streams in the related RTP sessions must change their SSRC. This is extra likely to cause problems if the participant populations are different in the different sessions. For example, in case of large number of receivers having selected totally random SSRC values in each RTP session as RFC 3550 specifies, a change due to a SSRC collision in one session can then cause a new collision in another session. This cascading effect is not severe but there is an increased risk that this occurs for well populated sessions. In addition, being forced to change the SSRC affects all the related media streams; instead of having to re-synchronize only the originally conflicting stream, all streams will suddenly need to be re-synchronized with each other. This will prevent also the media streams not having an actual collision from being usable during the re-synchronization and also increases the time until synchronization is finalized. In addition, it requires exception handling in the SSRC generation.

The above collision issue does not occur in case of having only one SSRC space across all sessions and all participants will be part of at least one session, like the base layer in layered encoding. In that case the only downside is the special behavior that needs to be well defined by anyone using this. But, having an exception behavior where the SSRC space is common across all session an that doesn't fit all the RTP extensions or payload formats present in the sessions is a issue. It is possible to create a situation where the different mechanisms can't be combined due to the non standard SSRC allocation behavior.

Existing mechanisms with known issues:

RTP Retransmission (RFC4588):
Has two modes, one for SSRC multiplexing and one for Session multiplexing. The session multiplexing requires the same CNAME and mandates that the same SSRC is used in both sessions. Using the same SSRC does work but will potentially have issues in certain cases. In SSRC multiplexed mode the CNAME is used, and when the first retransmission request is sent, one must not have another retransmission request outstanding for an SSRC which don't have a the binding between the original SSRC and the retransmission stream's SSRC. This works but creates some limitations that can be avoided by a more explicit mechanism. The SDP based ssrc-group mechanism is sufficient in this case as long as the application can rely on the signalling based solution.
Scalable Video Coding (RFC6190):
As an example of scalable coding, SVC [RFC6190] has various modes. The Multi Session Transmission (MST) uses Session multiplexing to separate scalability layers. However, this specification has failed to explicit how these layers are bound together in cases where CNAME isn't sufficient. CNAME is no longer sufficient when more than one media source occur within a session that have the same CNAME, for example due to multiple video cameras capturing the same lecture hall. This likely implies that a single SSRC space as recommend by Section 8.3 of RTP [RFC3550] is to be used.
Forward Error Correction:
If some type of FEC or redundancy stream is being sent, it needs it's own SSRC, with the exception of constructions like redundancy encoding [RFC2198]. Thus in case of transmitting the FEC in the same session as the source data, the inter SSRC relation within a session is needed. In case of sending the redundant data in a separate session from the source, the SSRC in each session needs to be related. This occurs for example in RFC5109 when using session separation of original and FEC data. SSRC multiplexing is not supported, only using redundant encoding is supported.

This issue appears to need action to harmonize and avoid future shortcomings in extension specifications. A proposed solution for handling this issue is [I-D.westerlund-avtext-rtcp-sdes-srcname].

7.2.7. Forward Error Correction

There exist a number of Forward Error Correction (FEC) based schemes for how to reduce the packet loss of the original streams. Most of the FEC schemes will protect a single source flow. The protection is achieved by transmitting a certain amount of redundant information that is encoded such that it can repair one or more packet loss over the set of packets they protect. This sequence of redundant information also needs to be transmitted as its own media stream, or in some cases instead of the original media stream. Thus many of these schemes creates a need for binding the related flows as discussed above. They also create additional flows that need to be transported. Looking at the history of these schemes, there is both SSRC multiplexed and Session multiplexed solutions and some schemes that support both.

Using a Session multiplexed solution provides good support for legacy when deploying FEC or changing the scheme used so that some set of receivers may not be able to utilize the FEC information. By placing it in a separate RTP session, it can easily be ignored.

In usages involving multicast, having the FEC information on its own multicast group and RTP session allows for flexibility, for example when using Rapid Acquisition of Multicast Groups (RAMS) [RFC6285]. During the RAMS burst where data is received over unicast and where it is possible to combine with unicast based retransmission [RFC4588], there is no need to burst the FEC data related to the burst of the source media streams needed to catch up with the multicast group. This saves bandwidth to the receiver during the burst, enabling quicker catch up. When the receiver has catched up and joins the multicast group(s) for the source, it can at the same time join the multicast group with the FEC information. Having the source stream and the FEC in separate groups allow for easy separation in the Burst/Retransmission Source (BRS) without having to individually classify packets.

7.2.8. Transport Translator Sessions

A basic Transport Translator relays any incoming RTP and RTCP packets to the other participants. The main difference between SSRC multiplexing and Session multiplexing resulting from this use case is that for SSRC multiplexing it is not possible for a particular session participant to decide to receive a subset of media streams. When using separate RTP sessions for the different sets of media streams, a single participant can choose to leave one of the sessions but not the other.

7.2.9. Multiple Media Types in one RTP session

Having different media types, like audio and video, in the same RTP sessions is not forbidden, only recommended against as can be seen in Section 7.2.1. When using multiple media types, there are a number of considerations:

Payload Type gives Media Type:
This solution is dependent on getting the media type from the Payload Type. Thus overloading this de-multiplexing point in a receiver for two purposes. First for the main media type and determining the processing chain, then later for the exact configuration of the encoder and packetization.
Payload Type field limiations:
The total number of Payload Types available to use in an RTP session is fairly limited, especially if Multiplexing RTP Data and Control Packets on a Single Port [RFC5761] is used. For certain applications negotiating a large set of codes and configuration may become an issue.
Don't switch media types for an SSRC:
The primary reasons to avoid switching from sending for example audio to sending video using the same SSRC is the implications on a receiver. When this happens, the processing chain in the receiver will have to switch from one media type to another. As the different media type's entire processing chains are different and are connected to different outputs it is difficult to reuse the decoding chain, which a normal codec change likely can. Instead the entire processing chain has to be torn down and replaced. In addition, there is likely a clock rate switching problem, possibly resulting in synchronization loss at the point of switching media type if some packet loss occurs.
RTCP Bit-rate Issues:
If the media types are significantly different in bit-rate, the RTCP bandwidth rates assigned to each source in a session can result in interesting effects, like that the RTCP bit-rate share for an audio stream is larger than the actual audio bit-rate. In itself this doesn't cause any conflicts, only potentially unnecessary overhead. It is possible to avoid this using AVPF or SAVPF and setting trr-int parameter, which can bring down unnecessary regular reporting while still allowing for rapid feedback.
Decomposited end-points:
Decomposited nodes that rely on the regular network to separate audio and video to different devices do not work well with this session setup. If they are forced to work, all media receiver parts of a decomposited end-point will receive all media, thus doubling the bit-rate consumption for the end-point.
RTP Mixers and Translators:
An RTP mixer or Media Translator will also have to support this particular session setup, where it before could rely on the RTP session to determine what processing options should be applied to the incoming packets.

As can be seen, there is nothing in here that prevents using a single RTP session for multiple media types, however it does create a number of limitations and special case implementation requirements. So anyone considering to use this setup should carefully review if the reasons for using a single RTP session is sufficient to motivate this special case.

7.3. Signalling Aspects

There exist various signalling solutions for establishing RTP sessions. Many are SDP [RFC4566] based, however SDP functionality is also dependent on the signalling protocols carrying the SDP. Where RTSP [RFC2326] and SAP [RFC2974] both use SDP in a declarative fashion, SIP [RFC3261] uses SDP with the additional definition of Offer/Answer [RFC3264]. The impact on signalling and especially SDP needs to be considered as it can greatly affect how to deploy a certain multiplexing point choice.

7.3.1. Session Oriented Properties

One aspect of the existing signalling is that it is focused around sessions, or at least in the case of SDP the media description. There are a number of things that are signalled on a session level/media description but that are not necessarily strictly bound to an RTP session and could be of interest to signal specifically for a particular media stream within the session. The following properties have been identified as being potentially useful to signal not only on RTP session level:

  • Bitrate/Bandwidth exist today only at aggregate or a common any media stream limit
  • Which SSRC that will use which RTP Payload Types

Some of these issues are clearly SDP's problem rather than RTP limitations. However, if the aim is to deploy an SSRC multiplexed solution that contains several sets of media streams with different properties (encoding/packetization parameter, bit-rate, etc), putting each set in a different RTP session would directly enable negotiation of the parameters for each set. If insisting on SSRC multiplexing, a number of signalling extensions are needed to clarify that there are multiple sets of media streams with different properties and that they shall in fact be kept different, since a single set will not satisfy the applications requirements.

This does in fact create a strong driver to use RTP session multiplexing for any case where different sets of media streams with different requirements exist.

7.3.2. SDP Prevents Multiple Media Types

SDP encoded in its structure a prevention against using multiple media types in the same RTP session. A media description in SDP can only have a single media type; audio, video, text, image, application. This media type is used as the top-level media type for identifying the actual payload format bound to a particular payload type using the rtpmap attribute. Thus a high fence against using multiple media types in the same session was created.

There is a proposal in the MMUSIC WG for how one could allow multiple media lines describe a single underlying transport [I-D.holmberg-mmusic-sdp-bundle-negotiation] and thus support either one RTP session with multiple media types. There is also a solution for multiplexing multiple RTP sessions onto the same transport [I-D.westerlund-avtcore-single-transport-multiplexing].

7.4. Network Apsects

The multiplexing choice has impact on network level mechanisms that need to be considered by the implementor.

7.4.1. Quality of Service

When it comes to Quality of Service mechanisms, they are either flow based or marking based. RSVP [RFC2205] is an example of a flow based mechanism, while Diff-Serv [RFC2474] is an example of a Marking based one. For a marking based scheme, the method of multiplexing will not affect the possibility to use QoS.

However, for a flow based scheme there is a clear difference between the methods. SSRC multiplexing will result in all media streams being part of the same 5-tuple (protocol, source address, destination address, source port, destination port) which is the most common selector for flow based QoS. Thus, separation of the level of QoS between media streams is not possible. That is however possible for session based multiplexing, where each different version can be in a different RTP session that can be sent over different 5-tuples.

7.4.2. NAT and Firewall Traversal

In today's network there exist a large number of middleboxes. The ones that normally have most impact on RTP are Network Address Translators (NAT) and Firewalls (FW).

Below we analyze and comment on the impact of requiring more underlying transport flows in the presence of NATs and Firewalls:

End-Point Port Consumption:
A given IP address only has 65536 available local ports per transport protocol for all consumers of ports that exist on the machine. This is normally never an issue for an end-user machine. It can become an issue for servers that handle large number of simultaneous streams. However, if the application uses ICE to authenticate STUN requests, a server can serve multiple end-points from the same local port, and use the whole 5-tuple (source and destination address, source and destination port, protocol) as identifier of flows after having securely bound them to the remote end-point address using the STUN request. In theory the minimum number of media server ports needed are the maximum number of simultaneous RTP Sessions a single end-point may use. In practice, implementation will probably benefit from using more server ports to simplify implementation or avoid performance bottlenecks.
NAT State:
If an end-point sits behind a NAT, each flow it generates to an external address will result in a state that has to be kept in the NAT. That state is a limited resource. In home or Small Office/Home Office (SOHO) NATs, memory or processing are usually the most limited resources. For large scale NATs serving many internal end-points, available external ports are typically the scarce resource. Port limitations is primarily a problem for larger centralized NATs where end-point independent mapping requires each flow to use one port for the external IP address. This affects the maximum number of internal users per external IP address. However, it is worth pointing out that a real-time video conference session with audio and video is likely using less than 10 UDP flows, compared to certain web applications that can use 100+ TCP flows to various servers from a single browser instance.
NAT Traversal Excess Time:
Making the NAT/FW traversal takes a certain amount of time for each flow. It also takes time in a phase of communication between accepting to communicate and the media path being established which is fairly critical. The best case scenario for how much extra time it can take following the specified ICE procedures are: 1.5*RTT + Ta*(Additional_Flows-1), where Ta is the pacing timer, which ICE specifies to be no smaller than 20 ms. That assumes a message in one direction, and then an immediate triggered check back. This as ICE first finds one candidate pair that works prior to establish multiple flows. Thus, there are no extra time until one has found a working candidate pair. Based on that working pair the extra time is to in parallel establish the, in most cases 2-3, additional flows.
NAT Traversal Failure Rate:
Due to the need to establish more than a single flow through the NAT, there is some risk that establishing the first flow succeeds but that one or more of the additional flows fail. The risk that this happens is hard to quantify, but it should be fairly low as one flow from the same interfaces has just been successfully established . Thus only rare events such as NAT resource overload, or selecting particular port numbers that are filtered etc, should be reasons for failure.

SSRC multiplexing keeps additional media streams within one RTP Session and does not introduce any additional NAT traversal complexities per media stream. In contrast, the session multiplexing is using one RTP session per media stream. Thus additional lower layer transport flows will be required, unless an explicit de-multiplexing layer is added between RTP and the transport protocol. A proposal for how to multiplex multiple RTP sessions over the same single lower layer transport exist in [I-D.westerlund-avtcore-single-transport-multiplexing].

7.4.3. Multicast

Multicast groups provides a powerful semantics for a number of real-time applications, especially the ones that desire broadcast-like behaviors with one end-point transmitting to a large number of receivers, like in IPTV. But that same semantics do result in a certain number of limitations.

One limitation is that for any group, sender side adaptation to the actual receiver properties causes a degradation for all participants to what is supported by the receiver with the worst conditions among the group participants. In most cases this is not acceptable. Instead various receiver based solutions are employed to ensure that the receivers achieve best possible performance. By using scalable encoding and placing each scalability layer in a different multicast group, the receiver can control the amount of traffic it receives. To have each scalability layer on a different multicast group, one RTP session per multicast group is used.

If instead a single RTP session over multiple transports were to be deployed, i.e. multicast groups with each layer as it's own SSRC, then very different views of the RTP session would exist. That as one receiver may see only a single layer (SSRC), while another may see three SSRCs if it joined three multicast groups. This would cause disjoint RTCP reports where a management system would not be able to determine if a receiver isn't reporting on a particular SSRC due to that it is not a member of that multicast group, or because it doesn't receive it as a result of a transport failure.

Thus it appears easiest and most straightforward to use multiple RTP sessions. In addition, the transport flow considerations in multicast are a bit different from unicast. First of all there is no shortage of port space, as each multicast group has its own port space.

7.4.4. Multiplexing multiple RTP Session on a Single Transport

For applications that doesn't need flow based QoS and like to save ports and NAT/FW traversal costs, there is a proposal for how to achieve multiplexing of multiple RTP sessions over the same lower layer transport [I-D.westerlund-avtcore-single-transport-multiplexing]. Using such a solution would allow session multiplexing without most of the perceived downsides of additional RTP sessions creating a need for additional transport flows.

7.5. Security Aspects

On the basic level there is no significant difference in security when having one RTP session and having multiple. However, there are a few more detailed considerations that might need to be considered in certain usages.

7.5.1. Security Context Scope

When using SRTP [RFC3711] the security context scope is important and can be a necessary differentiation in some applications. As SRTP's crypto suites (so far) is built around symmetric keys, the receiver will need to have the same key as the sender. This results in that none in a multi-party session can be certain that a received packet really was sent by the claimed sender or by another party having access to the key. In most cases this is a sufficient security property, but there are a few cases where this does create situations.

The first case is when someone leaves a multi-party session and one wants to ensure that the party that left can no longer access the media streams. This requires that everyone re-keys without disclosing the keys to the excluded party.

A second case is when using security as an enforcing mechanism for differentiation. Take for example a scalable layer or a high quality simulcast version which only premium users are allowed to access. The mechanism preventing a receiver from getting the high quality stream can be based on the stream being encrypted with a key that user can't access without paying premium, having the key-management limit access to the key.

In the latter case it is likely easiest from signalling, transport (if done over multicast) and security to use a different RTP session. That way the user(s) not intended to receive a particular stream can easily be excluded. There is no need to have SSRC specific keys, which many of the key-management systems cannot handle.

7.5.2. Key-Management for Multi-party session

Performing key-management for Multi-party session can be a challenge. This section considers some of the issues.

Transport translator based session cannot use Security Description [RFC4568] nor DTLS-SRTP [RFC5764] without an extension as each end-point provides it's set of keys. In centralized conference, the signalling counterpart is a conference server and the media plane unicast counterpart (to which DTLS messages would be sent) is the translator. Thus an extension like Encrypted Key Transport [I-D.ietf-avt-srtp-ekt] are needed or a MIKEY [RFC3830] based solution that allows for keying all session participants with the same master key.

Keying of multicast transported SRTP face similar challenges as the transport translator case.

8. Guidelines

This section contains a number of recommendations for implementors or specification writers when it comes to handling multi-stream.

Don't Require the same SSRC across Sessions:
As discussed in Section 7.2.6 there exist drawbacks in using the same SSRC in multiple RTP sessions as a mechanism to bind related media streams together. Instead a mechanism to explicitly signal the relation SHOULD be used, either in RTP/RTCP or in the used signalling mechanism that establish the RTP session(s).
Use SSRC multiplexing for additional Media Sources:
In the cases an RTP end-point needs to transmit additional media source(s) of the same media type and purpose in the application it is RECOMMENDED to send them as additional SSRCs in the same RTP session. For example a telepresence room where there are three cameras, and each camera captures 2 persons sitting at the table, sending each camera as its own SSRC within a single RTP session is recommended.
Use additional RTP sessions for streams with different purposes:
When media streams have different purpose or processing requirements it is RECOMMENDED that the different types of streams are put in different RTP sessions.
When using Session Multiplexing use grouping:
When using Session Multiplexing solutions it is RECOMMENDED to be explicitly group the involved RTP sessions using the signalling mechanism, for example The Session Description Protocol (SDP) Grouping Framework. [RFC5888]
RTP/RTCP Extensions May Support SSRC and Session Multiplexing:
When defining an RTP or RTCP extension, the creator needs to consider if this extension is applicable in both SSRC multiplexed and Session multiplexed usages. If it is, then any generic extensions are RECOMMENDED to support both. Applications that are not as generally applicable will have to consider if interoperability is better served by defining a single solution or providing both options.
Transport Support Extensions:
When defining new RTP/RTCP extensions intended for transport support, like the retransmission or FEC mechanisms, they are RECOMMENDED to include support for both SSRC and Session multiplexing so that application developers can choose freely from the set of mechanisms without concerning themselves with if a particular solution only supports one of the multiplexing choices.

This discussion and guidelines points out that a small set of extension mechanisms could greatly improve the situation when it comes to using multiple streams independently of Session multiplexing or SSRC multiplexing. These extensions are:

Media Source Identification:
A Media source identification that can be used to bind together media streams that are related to the same media source. A proposal [I-D.westerlund-avtext-rtcp-sdes-srcname] exist for a new SDES item SRCNAME that also can be used with the a=ssrc SDP attribute to provide signalling layer binding information.
SSRC limiations within RTP sessions:
By providing a signalling solution that allows the signalling peers to explicitly express both support and limitations on how many simultaneous media streams an end-point can handle within a given RTP Session. That ensures that usage of SSRC multiplexing occurs when supported and without overloading an end-point. This extension is proposed in [I-D.westerlund-avtcore-max-ssrc].

9. RTP Specification Clarifications

This section describes a number of clarifications to the RTP specifications that are likely necessary for aligned behavior when RTP sessions contains more SSRCs than one local and one remote.

9.1. RTCP Reporting from all SSRCs

When one have multiple SSRC in an RTP node, then all these SSRC must send RTCP SR or RR as long as the SSRC exist. It is not sufficient that only one SSRC in the node sends report blocks on the incoming RTP streams. The reason for this is that a third party monitor may not necessarily be able to determine that all these SSRC are in fact co-located and originate from the same stack instance that gather report data.

9.2. RTCP Self-reporting

For any RTP node that sends more than one SSRC, there exist the question if SSRC1 needs to report its reception of SSRC2 and vice versa. The reason that they in fact need to report on all other local streams as being received is report consistency. A third party monitor that considers the full matrix of media streams and all known SSRC reports on these media streams would detect a gap in the reports which could be a transport issue unless identified as in fact being sources from same node.

9.3. Combined RTCP Packets

When a node contains multiple SSRCs, it is questionable if an RTCP compound packet can only contain RTCP packets from a single SSRC or if multiple SSRCs can include their packets in a joint compound packet. The high level question is a matter for any receiver processing on what to expect. In addition to that question there is the issue of how to use the RTCP timer rules in these cases, as the existing rules are focused on determining when a single SSRC can send.

10. IANA Considerations

This document makes no request of IANA.

Note to RFC Editor: this section may be removed on publication as an RFC.

11. Security Considerations

12. Acknowledgements

13. References

13.1. Normative References

[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.

13.2. Informative References

[RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V., Handley, M., Bolot, J., Vega-Garcia, A. and S. Fosse-Parisis, "RTP Payload for Redundant Audio Data", RFC 2198, September 1997.
[RFC2205] Braden, B., Zhang, L., Berson, S., Herzog, S. and S. Jamin, "Resource ReSerVation Protocol (RSVP) -- Version 1 Functional Specification", RFC 2205, September 1997.
[RFC2326] Schulzrinne, H., Rao, A. and R. Lanphier, "Real Time Streaming Protocol (RTSP)", RFC 2326, April 1998.
[RFC2474] Nichols, K., Blake, S., Baker, F. and D.L. Black, "Definition of the Differentiated Services Field (DS Field) in the IPv4 and IPv6 Headers", RFC 2474, December 1998.
[RFC2974] Handley, M., Perkins, C. and E. Whelan, "Session Announcement Protocol", RFC 2974, October 2000.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., Peterson, J., Sparks, R., Handley, M. and E. Schooler, "SIP: Session Initiation Protocol", RFC 3261, June 2002.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with Session Description Protocol (SDP)", RFC 3264, June 2002.
[RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload for Comfort Noise (CN)", RFC 3389, September 2002.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video Conferences with Minimal Control", STD 65, RFC 3551, 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.
[RFC3830] Arkko, J., Carrara, E., Lindholm, F., Naslund, M. and K. Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830, August 2004.
[RFC4103] Hellstrom, G. and P. Jones, "RTP Payload for Text Conversation", RFC 4103, June 2005.
[RFC4566] Handley, M., Jacobson, V. and C. Perkins, "SDP: Session Description Protocol", RFC 4566, July 2006.
[RFC4568] Andreasen, F., Baugher, M. and D. Wing, "Session Description Protocol (SDP) Security Descriptions for Media Streams", RFC 4568, July 2006.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V. and R. Hakenberg, "RTP Retransmission Payload Format", RFC 4588, July 2006.
[RFC4607] Holbrook, H. and B. Cain, "Source-Specific Multicast for IP", RFC 4607, August 2006.
[RFC5104] Wenger, S., Chandra, U., Westerlund, M. and B. Burman, "Codec Control Messages in the RTP Audio-Visual Profile with Feedback (AVPF)", RFC 5104, February 2008.
[RFC5117] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117, January 2008.
[RFC5583] Schierl, T. and S. Wenger, "Signaling Media Decoding Dependency in the Session Description Protocol (SDP)", RFC 5583, July 2009.
[RFC5576] Lennox, J., Ott, J. and T. Schierl, "Source-Specific Media Attributes in the Session Description Protocol (SDP)", RFC 5576, June 2009.
[RFC5760] Ott, J., Chesterfield, J. and E. Schooler, "RTP Control Protocol (RTCP) Extensions for Single-Source Multicast Sessions with Unicast Feedback", RFC 5760, February 2010.
[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.
[RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description Protocol (SDP) Grouping Framework", RFC 5888, June 2010.
[RFC6190] Wenger, S., Wang, Y.-K., Schierl, T. and A. Eleftheriadis, "RTP Payload Format for Scalable Video Coding", RFC 6190, May 2011.
[RFC6285] Ver Steeg, B., Begen, A., Van Caenegem, T. and Z. Vax, "Unicast-Based Rapid Acquisition of Multicast RTP Sessions", RFC 6285, June 2011.
[I-D.ietf-avtext-multiple-clock-rates] Petit-Huguenin, M, "Support for multiple clock rates in an RTP session", Internet-Draft draft-ietf-avtext-multiple-clock-rates-01, July 2011.
[I-D.ietf-payload-rtp-howto] Westerlund, M, "How to Write an RTP Payload Format", Internet-Draft draft-ietf-payload-rtp-howto-01, July 2011.
[I-D.ietf-avt-srtp-ekt] McGrew, D, Wing, D and K Fischer, "Encrypted Key Transport for Secure RTP", Internet-Draft draft-ietf-avt-srtp-ekt-03, October 2011.
[I-D.westerlund-avtext-rtcp-sdes-srcname] Westerlund, M., Burman, B. and P. Sandgren, "RTCP SDES Item SRCNAME to Label Individual Sources", Internet-Draft draft-westerlund-avtext-rtcp-sdes-srcname, October 2011.
[I-D.westerlund-avtcore-max-ssrc] Westerlund, M., Burman, B. and F. Jansson, "Multiple Synchronization sources (SSRC) in RTP Session Signaling", Internet-Draft draft-westerlund-avtcore-max-ssrc, October 2011.
[I-D.westerlund-avtcore-single-transport-multiplexing] Westerlund, M., "Multiple RTP Session on a Single Lower-Layer Transport", Internet-Draft draft-westerlund-avtcore-transport-multiplexing, October 2011.
[ALF] Clark, D. and D. Tennenhouse, "Architectural Considerations for a New Generation of Protocols", SIGCOMM Symposium on Communications Architectures and Protocols (Philadelphia, Pennsylvania), pp. 200--208, IEEE Computer Communications Review, Vol. 20(4), September 1990.
[I-D.holmberg-mmusic-sdp-bundle-negotiation] Holmberg, C and H Alvestrand, "Multiplexing Negotiation Using Session Description Protocol (SDP) Port Numbers", Internet-Draft draft-holmberg-mmusic-sdp-bundle-negotiation-00, October 2011.

Authors' Addresses

Magnus Westerlund Ericsson Farogatan 6 SE-164 80 Kista, Sweden Phone: +46 10 714 82 87 EMail:
Bo Burman Ericsson Farogatan 6 SE-164 80 Kista, Sweden Phone: +46 10 714 13 11 EMail:
Colin Perkins University of Glasgow School of Computing Science Glasgow, G12 8QQ United Kingdom EMail: