AVTCORE WG M. Westerlund
Internet-Draft Ericsson
Updates: 3550, 3551 (if approved) C. Perkins
Intended status: Standards Track University of Glasgow
Expires: August 29, 2013 J. Lennox
February 25, 2013

Multiple Media Types in an RTP Session


This document specifies how an RTP session can contain media streams with media from multiple media types such as audio, video, and text. This has been restricted by the RTP Specification, and thus this document updates RFC 3550 and RFC 3551 to enable this behaviour for applications that satisfy the applicability for using multiple media types in a single RTP session.

Status of This Memo

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This Internet-Draft will expire on August 29, 2013.

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Table of Contents

1. Introduction

When the Real-time Transport Protocol (RTP) [RFC3550] was designed, close to 20 years ago, IP networks were very different compared to the ones in 2013 when this is written. The almost ubiquitous deployment of Network Address Translators (NAT) and Firewalls has increased the cost and likely-hood of communication failure when using many different transport flows. Thus there exists a pressure to reduce the number of concurrent transport flows.

RTP [RFC3550] recommends against sending several different types of media, for example audio and video, in a single RTP session. The RTP profile for Audio and Video Conferences with Minimal Control (RTP/AVP) [RFC3551] mandates a similar restriction. The motivation for these limitations is partly to allow lower layer Quality of Service (QoS) mechanisms to be used, and partly due to limitations of the RTCP timing rules that assumes all media in a session to have similar bandwidth. The Session Description Protocol (SDP) [RFC4566], as one of the dominant signalling method for establishing RTP session, has enforced this rule, simply by not allowing multiple media types for a given receiver destination or set of ICE candidates, which is the most common method to determine which RTP session the packets are intended for.

The fact that these limitations have been in place for so long a time, in addition to RFC 3550 being written without fully considering multiple media types in an RTP session, does result in a number of considerations being needed when allowing this behaviour. This document provides such considerations regarding applicability as well as functionality, including normative specification of behaviour.

First, some basic definitions are provided. This is followed by a background that discusses the motivation in more detail. A overview of the solution of how to provide multiple media types in one RTP session is then presented. Next is the formal applicability this specification have followed by the normative specification. This is followed by a discussion how some RTP/RTCP Extensions is expected to function in the case of multiple media types in one RTP session. A specification of the requirements on signalling from this specification and a look how this is realized in SDP using Bundle [I-D.ietf-mmusic-sdp-bundle-negotiation]. The document ends with the security considerations.

2. Definitions

The following terms are used with supplied definitions:

A single entity sending or receiving RTP packets. It can be decomposed into several functional blocks, but as long as it behaves as a single RTP stack entity it is classified as a single endpoint.
Media Stream:
A sequence of RTP packets using a single SSRC that together carries part or all of the content of a specific Media Type from a specific sender source within a given RTP session.
Media Type:
Audio, video, text or application whose form and meaning are defined by a specific real-time application.
Quality of Service, i.e. network mechanisms that intended to ensure that the packets within a flow or with a specific marking are transported with certain properties.
RTP Session:
As defined by [RFC3550], the endpoints belonging to the same RTP Session are those that share a single SSRC space. That is, those endpoints can see an SSRC identifier transmitted by any one of the other endpoints. An endpoint can receive an SSRC either as SSRC or as CSRC in RTP and RTCP packets. Thus, the RTP Session scope is decided by the endpoints' network interconnection topology, in combination with RTP and RTCP forwarding strategies deployed by endpoints and any interconnecting middle nodes.

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 [RFC2119].

3. Motivation

This section discusses in more detail the main motivations why allowing multiple media types in the same RTP session is suitable.

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 needs to ensure that all these transport flows are established. This has three consequences:

  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 behaviours usually occur.
  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 behaviour 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.

3.3. Architectural Equality

For applications that don't require different lower-layer QoS for different media types, and that have no special requirements for RTP extensions or RTCP reporting, the requirement to separate different media into different RTP sessions might seem unnecessary. Provided the application accepts that all media flows will get similar RTCP reporting, using the same RTP session for several types of media at once appears a reasonable choice. The architecture ought to be agnostic about the type of media being carried in an RTP session to the extent possible given the constraints of the protocol.

4. Overview of Solution

The goal of the solution is to enable each RTP session to contain more than just one media type. This includes having multiple RTP sessions containing a given media type, for example having three sessions containing both video and audio.

The solution is quite straightforward. The first step is to override the SHOULD and SHOULD NOT language of the RTP specification [RFC3550]. Similar change is needed to a sentence in Section 6 of [RFC3551] that states that "different media types SHALL NOT be interleaved or multiplexed within a single RTP Session". This is resolved by appropriate exception clauses given that this specification and its applicability is followed.

Within an RTP session where multiple media types have been configured for use, an SSRC can only send one type of media during its lifetime (i.e., it can switch between different audio codecs, since those are both the same type of media, but cannot switch between audio and video). Different SSRCs MUST be used for the different media sources, the same way multiple media sources of the same media type already have to do. The payload type will inform a receiver which media type the SSRC is being used for. Thus the payload type MUST be unique across all of the payload configurations independent of media type that is used in the RTP session.

Some few extra considerations within the RTP sessions also needs to be considered. RTCP bandwidth and regular reporting suppression (RTP/AVPF and RTP/SAVPF) SHOULD be configured to reduce the impact for bit-rate variations between streams and media types. It is also clarified how timeout calculations are to be done to avoid any issues. Certain payload types like FEC also need additional rules.

The final important part of the solution to this is to use signalling and ensure that agreement on using multiple media types in an RTP session exists, and how that then is configured. This memo describes some existing requirements, while an external reference defines how this is accomplished in SDP.

5. Applicability

This specification has limited applicability, and anyone intending to use it needs to ensure that their application and usage meets the below criteria.

5.1. Usage of the RTP session

Before choosing to use this specification, an application implementer needs to ensure that they don't have a need for different RTP sessions between the media types for some reason. The main rule is that if one expects to have equal treatment of all media packets, then this specification might be suitable. The equal treatment include anything from network level up to RTCP reporting and feedback. The document Guidelines for using the Multiplexing Features of RTP [I-D.westerlund-avtcore-multiplex-architecture] gives more detailed guidance on aspects to consider when choosing how to use RTP and specifically sessions. RTP-using applications that need or would prefer multiple RTP sessions, but do not require the functionalities or behaviours that multiple transport flows give, can consider using Multiple RTP Sessions on a Single Lower-Layer Transport [I-D.westerlund-avtcore-transport-multiplexing]. It needs to be noted that some difference in treatment is still possible to achieve, for example marking based QoS, or RTCP feedback traffic for only some media streams.

The second important consideration is the resulting behaviour when media flows to be sent within a single RTP session does not have similar bandwidth. There are limitations in the RTCP timing rules, and this implies a common RTCP reporting interval across all participants in a session. If an RTP session contains flows with very different bandwidths, for example low-rate audio coupled with high-quality video, this can result in either excessive or insufficient RTCP for some flows, depending how the RTCP session bandwidth, and hence reporting interval, is configured. This is discussed further in Section 6.4.

5.2. Signalled Support

Usage of this specification is not compatible with anyone following RFC 3550 and intending to have different RTP sessions for each media type. Therefore there needs to be mutual agreement to use multiple media types in one RTP session by all participants within that RTP session. This agreement has to be determined using signalling in most cases.

This requirement can be a problem for signalling solutions that can't negotiate with all participants. For declarative signalling solutions, mandating that the session is using multiple media types in one RTP session can be a way of attempting to ensure that all participants in the RTP session follow the requirement. However, for signalling solutions that lack methods for enforcing that a receiver supports a specific feature, this can still cause issues.

5.3. Homogeneous Multi-party

In multiparty communication scenarios it is important to separate two different cases. One case is where the RTP session contains multiple participants in a common RTP session. This occurs for example in Any Source Multicast (ASM) and Transport Translator topologies as defined in RTP Topologies [RFC5117]. It can also occur in some implementations of RTP mixers that share the same SSRC/CSRC space across all participants. The second case is when the RTP session is terminated in a middlebox and the other participants sources are projected or switched into each RTP session and rewritten on RTP header level including SSRC mappings.

For the first case, with a common RTP session or at least shared SSRC/CSRC values, all participants in multiparty communication are REQUIRED to support multiple media types in an RTP session. An participant using two or more RTP sessions towards a multiparty session can't be collapsed into a single session with multiple media types. The reason is that in case of multiple RTP sessions, the same SSRC value can be use in both RTP sessions without any issues, but when collapsed to a single session there is an SSRC collision. In addition some collisions can't be represented in the multiple separate RTP sessions. For example, in a session with audio and video, an SSRC value used for video will not show up in the Audio RTP session at the participant using multiple RTP sessions, and thus not trigger any collision handling. Thus any application using this type of RTP session structure MUST have a homogeneous support for multiple media types in one RTP session, or be forced to insert a translator node between that participant and the rest of the RTP session.

For the second case of separate RTP sessions for each multiparty participant and a central node it is possible to have a mix of single RTP session users and multiple RTP session users as long as one is willing to remap the SSRCs used by a participant with multiple RTP sessions into non-used values in the single RTP session SSRC space for each of the participants using a single RTP session with multiple media types. It can be noted that this type of implementation has to understand all types of RTP/RTCP extension being used in the RTP sessions to correctly be able to translate them between the RTP sessions. It can also negatively impact the possibility for loop detection, as SSRC/CSRC can't be used to detect the loops, instead some other media stream identity name space that is common across all interconnect parts are needed.

5.4. Reduced number of Payload Types

An RTP session with multiple media types in it have only a single 7-bit Payload Type range for all its payload types. Within the 128 available values, only 96 or less if "Multiplexing RTP Data and Control Packets on a Single Port" [RFC5761] is used, all the different RTP payload configurations for all the media types need to fit in the available space. For most applications this will not be a real problem, but the limitation exists and could be encountered.

5.5. Stream Differentiation

If network level differentiation of the media streams of different media types are desired using this specification can cause severe limitations. All media streams in an RTP session, independent of the media type, will be sent over the same underlying transport flow. Any flow-based Quality of Service (QoS) mechanism will be unable to provide differentiated treatment between different media types, e.g. to prioritize audio over video. If differentiated treatment is desired using flow-based QoS, separate RTP sessions over different underlying transport flows needs to be used.

Any marking-based QoS scheme like DiffServ is not affected unless a network ingress marks based on flows, in which case the same considerations as for flow based QoS applies.

5.6. Non-compatible Extensions

There exist some RTP and RTCP extensions that rely on the existence of multiple RTP sessions. If the goal of using an RTP session with multiple media types is to have only a single RTP session, then these extensions can't be used. If one has no need to have different RTP sessions for the media types but is willing to have multiple RTP sessions, one for the main media transmission and one for the extension, they can be used. It is to be noted that this assumes that it is possible to get the extension working when the related RTP session contains multiple media types.

Identified RTP/RTCP extensions that require multiple RTP Sessions are:

RTP Retransmission:
RTP Retransmission [RFC4588] has a session multiplexed mode. It also has a SSRC multiplexed mode that can be used instead. So use the mode that is suitable for the RTP application.
XOR-Based FEC:
The RTP Payload Format for Generic Forward Error Correction [RFC5109] and its predecessor [RFC2733] requires a separate RTP session unless the FEC data is carried in RTP Payload for Redundant Audio Data [RFC2198]. However, using the Generic FEC with the Redundancy payload has another set of restrictions, see Section 7.2.
Note that the Source-Specific Media Attributes [RFC5576] specification defines an SDP syntax (the "FEC" semantic of the "ssrc-group" attribute) to signal FEC relationships between multiple media streams within a single RTP session. However, this can't be used as the FEC repair packets need to have the same SSRC value as the source packets being protected. [RFC5576] does not normatively update and resolve that restriction. There is ongoing work on an ULP extension to allow it be use FEC streams within the same RTP Session as the source stream [I-D.lennox-payload-ulp-ssrc-mux].

6. RTP Session Specification

This section defines what needs to be done or avoided to make an RTP session with multiple media types function without issues.

6.1. RTP Session

Section 5.2 of "RTP: A Transport Protocol for Real-Time Applications" [RFC3550] states:

This specification changes both of these sentences. The first sentence is changed to:

The second sentence is changed to:

Second paragraph of Section 6 in RTP Profile for Audio and Video Conferences with Minimal Control [RFC3551] says:

This specifications purpose is to violate that existing SHALL NOT under certain conditions. Thus also this sentence has to be changed to allow for multiple media type's payload types in the same session. The above sentence is changed to:

RFC-Editor Note: Please replace RFCXXXX with the RFC number of this specification when assigned.

We can now go on and discuss the five bullets that are motivating the previous in Section 5.2 of the RTP Specification [RFC3550]. They are repeated here for the reader's convenience:

  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 of RFC 3550) 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.

Bullets 1 to 3 are all related to that each media source has to use one or more unique SSRCs to avoid these issues as mandated below [sec-source-restrcitctions]. Bullet 4 can be served by two arguments, first of all each SSRC will be associated with a specific media type, communicated through the RTP payload type, allowing a middlebox to do media type specific operations. The second argument is that in many contexts blind combining without additional contexts are anyway not suitable. Regarding bullet 5 this is a understood and explicitly stated applicability limitations for the method described in this document.

6.2. Sender Source Restrictions

A SSRC in the RTP session MUST only send one media type (audio, video, text etc.) during the SSRC's lifetime. The main motivation is that a given SSRC has its own RTP timestamp and sequence number spaces. The same way that you can't send two streams of encoded audio on the same SSRC, you can't send one audio and one video encoding on the same SSRC. Each media encoding when made into an RTP stream needs to have the sole control over the sequence number and timestamp space. If not, one would not be able to detect packet loss for that particular stream. Nor can one easily determine which clock rate a particular SSRCs timestamp will increase with. For additional arguments why RTP payload type based multiplexing of multiple media streams doesn't work see Appendix A in [I-D.westerlund-avtcore-multiplex-architecture].

6.3. Payload Type Applicability

Most Payload Types have a native media type, like an audio codec is natural belonging to the audio media type. However, there exist a number of RTP payload types that don't have a native media type. For example, transport robustness mechanisms like RTP Retransmission [RFC4588] and Generic FEC [RFC5109] inherit their media type from what they protect. RTP Retransmission is explicitly bound to the payload type it is protecting, and thus will inherit it. However Generic FEC is a excellent example of an RTP payload type that has no natural media type. The media type for what it protects is not relevant as it is the recovered RTP packets that have a particular media type, and thus Generic FEC is best categorized as an application media type.

The above discussion is relevant to what limitations exist for RTP payload type usage within an RTP session that has multiple media types. In fact this document [sec-generic-fec] suggest that for usage of Generic FEC (XOR-based) as defined in RFC 5109 can actually use a single media type when used with independent RTP sessions for source and repair data.

Note a particular SSRC carrying Generic FEC will clearly only protect a specific SSRC and thus that instance is bound to the SSRC's media type. For this specific case, it is possible to have one be applicable to both. However, in cases when the signalling is setup to enable fall back to using separate RTP sessions, then using a different media type, e.g. application, than the media being protected can create issues.

6.4. RTCP

An RTP session has a single set of parameters that configure the session bandwidth, the RTCP sender and receiver fractions (e.g., via the SDP "b=RR:" and "b=RS: lines), and the parameters of the RTP/AVPF profile [RFC4585] (e.g., trr-int) if that profile (or its secure extension, RTP/SAVPF [RFC5124]) is used. As a consequence, the RTCP reporting interval will be the same for every SSRC in an RTP session. This uniform RTCP reporting interval can result in RTCP reports being sent more often than is considered desirable for a particular media type. For example, if an audio flow is multiplexed with a high quality video flow where the session bandwidth is configured to match the video bandwidth, this can result in the RTCP packets having a greater bandwidth allocation than the audio data rate. If the reduced minimum RTCP interval described in Section 6.2 of [RFC3550] is used in the session, which might be appropriate for video where rapid feedback is wanted, the audio sources could be expected to send RTCP packets more often than they send audio data packets. This is most likely undesirable, and while the mismatch can be reduced through careful tuning of the RTCP parameters, particularly trr_int in RTP/AVPF sessions, it is inherent in the design of the RTCP timing rules, and affects all RTP sessions containing flows with mismatched bandwidth.

Having multiple media types in one RTP session also results in more SSRCs being present in this RTP session. This increasing the amount of cross reporting between the SSRCs. From an RTCP perspective, two RTP sessions with half the number of SSRCs in each will be slightly more efficient. If someone needs either the higher efficiency due to the lesser number of SSRCs or the fact that one can't tailor RTCP usage per media type, they need to use independent RTP sessions.

When it comes to handling multiple SSRCs in an RTP session there is a clarification under discussion in Real-Time Transport Protocol (RTP) Considerations for Multi-Stream Endpoints [I-D.lennox-avtcore-rtp-multi-stream]. When it comes to configuring RTCP the need for regular periodic reporting needs to be weighted against any feedback or control messages being sent. The applications using RTP/AVPF or RTP/SAVPF are RECOMMENDED to consider setting trr-int parameter to a value suitable for the applications needs, thus potentially reducing the need for regular reporting and thus releasing more bandwidth for use for feedback or control.

Another aspect of an RTP session with multiple media types is that the used RTCP packets, RTCP Feedback Messages, or RTCP XR metrics used might not be applicable to all media types. Instead all RTP/RTCP endpoints need to correlate the media type of the SSRC being referenced in an messages/packet and only use those that apply to that particular SSRC and its media type. Signalling solutions might have shortcomings when it comes to indicate that a particular set of RTCP reports or feedback messages only apply to a particular media type within an RTP session.

6.4.1. Timing out SSRCs

All used SSRCs in the RTP session MUST use the same timeout behaviour to avoid premature timeouts. This will depend on the RTP profile and its configuration. The RTP specification provides several options that can influence the values used when calculating the time-interval, to avoid such issues when using this specification we make clarification on the calculations.

For RTP/AVP, RTP/SAVP, RTP/AVPF, and RTP/SAVPF with T_rr_interval = 0 the timeout interval SHALL be calculated using a multiplier of 5, i.e. the timeout interval becomes 5*Td. The Td calculation SHALL be done using a Tmin value of 5 seconds, not the reduced minimal interval even if used to calculate RTCP packet transmission intervals. If using either the RTP/AVPF or RTP/SAVPF profiles with T_rr_interval != 0 then the calculation as specified in Section 3.5.4 of RFC 4585 SHALL be used with a multiplier of 5, i.e. Tmin in the Td calculation is the T_rr_interval.

Note: If endpoints implementing the RTP/AVP and RTP/AVPF profiles (or their secure variants) are combined in a single RTP session, and the RTP/AVPF endpoints use a non-zero T_rr_interval that is significantly lower than 5 seconds, then there is a risk that the RTP/AVP endpoints will prematurely timeout the RTP/AVPF endpoints due to their different RTCP timeout intervals. Since an RTP session can only use a single RTP profile, this issue ought never occur. If such mixed RTP profiles are used, however, the RTP/AVPF session MUST NOT use a non-zero T_rr_interval that is smaller than 5 seconds.

(tbd: it has been suggested that a minimum non-zero T_rr_interval of 4 seconds is more appropriate, due to the nature of the timing rules).

6.4.2. Tuning RTCP transmissions

This sub-section discusses what tuning can be done to reduce downsides of the shared RTCP packet intervals.

When using the RTP/AVP or RTP/SAVP profile the tuning one can do is very limited. The controls one has are very limited to the RTCP bandwidth values and if one scales the minimum RTCP interval according to the bandwidth. As the scheduling algorithm includes both random factors and reconsideration, one can't simply calculate the expected average transmission interval using formula for Td. But it does indicate the important factors affecting the transmission interval, namely the RTCP bandwidth available for the role (Active Sender or Participant), the average RTCP packet size and the number of SSRCs classified in the relevant role. Note, that if the ratio of senders to total number of session participants are larger than the ratio of RTCP bandwidth for senders in relation to the total RTCP bandwidth, then senders and receivers are treated together.

Lets start with some basic observations:

  1. Unless scaled minimum RTCP interval is used, then Td prior to randomization and reconsideration can never be less than 5 seconds (assuming default Tmin of 5 seconds).
  2. If scaled minimum RTCP interval is used Td can become as low as 360 divided by RTP Session bandwidth in kilobits. In SDP the RTP session bandwidth is signalled using b=AS. A RTP Session bandwidth of 72 kbps results in Tmin being 5 seconds. A RTP session bandwidth of 360 kbps of course gives a Tmin of 1 second, and to achieve a Tmin equal to once every frame for a 25 Hz video stream requires an RTP session bandwidth of 9 Mbps! (The use of the RTP/AVPF or RTP/SAVPF profile allows smaller Tmin, and hence more frequent RTCP report, as discussed below).
  3. Lets calculate the number (n) of SSRCs in the RTP session that 5% of the session bandwidth can support to yield a Td value equal to Tmin with minimal scaling. For this calculation we have to make two assumptions. The first is that we will consider most or all SSRC being senders resulting in everyone sharing the available bandwidth. Secondly we will select an average RTCP packet size. This packet will consist of an SR, containing (n-1) report blocks up to 31 report blocks, a SDES item with at least a CNAME (17 bytes value) in it. Such a basic packet will be 800 bytes for n>=32. With these parameters, and as the bandwidth goes up the time interval is proportionally decreased (due to minimal scaling), thus all the example bandwidths 72 kbps, 360 kbps and 9 Mbps all support 9 SSRCs.
  4. The actual transmission interval for a Td value is [0.5*Td/1.21828,1.5*Td/1.21828], which means that for Td = 5 seconds, the interval is actually [2.052,6.156] and the distribution is not uniform, it is an exponential increasing one. The probability for sending at time X, given it is within the interval, is probability of picking X in the interval times the probability to randomly picking a number that is <=X within the interval with an uniform probability distribution. This results in that the majority of the probability mass is above the Td value.

To conclude, with RTP/AVP and RTP/SAVP the key limitation for small unicast sessions are going to be the Tmin value. Thus the RTP session bandwidth configured in RTCP has to be sufficient high to reach the reporting goals the application has following the rules for scaled minimal RTCP interval.

When using RTP/AVPF or RTP/SAVPF we get a quite powerful additional tool, the setting of the T_rr_interval which has several effects on the RTCP reporting. First of all as Tmin is set to 0 after the initial transmission and regular reporting interval is instead affected of the regular bandwidth based calculation and the T_rr_interval. This has the affect that we are no longer restricted by the minimal interval or even the scaling rule for the minimal rule. Instead the RTCP bandwidth and the T_rr_interval is the governing factors. Now it also becomes important to separate between the applications need for regular reports and RTCP feedback packet types. In both regular RTCP mode, as in Early RTCP Mode, the usage of the T_rr_Interval prevents regular RTCP packets, i.e. packets without any Feedback packets to be sent more often than T_rr_interval. This value is a hard as no regular RTCP packet can be sent less than T_rr_interval after the previous regular packet packet.

So for applications that has a use for feedback packets for some media streams, for example video packets but don't want to frequent regular reporting for audio could configure the T_rr_interval to a value so that the regular reporting for both audio and video is at a level that is considered acceptable for the audio. Then use feedback packets, which will include RTCP SR/RR packets, unless reduced-size RTCP feedback packets [RFC5506] are used, and can include other report information in addition to the feedback packet that needs to be sent. That way the available RTCP bandwidth can be focused for use, which provides the most utility for the application.

Using T_rr_interval still requires one to determine suitable values for the RTCP bandwidth value, in fact it might make it even more important, as one is more likely to affect the RTCP behaviour and performance, than when using RTP/AVP, as their is fewer limitations affecting the RTCP transmission.

When using T_rr_interval, i.e. having it be non zero, there are configurations that have to be avoided. If the resulting Td value is smaller but close to T_rr_interval then the interval in which the actual regular RTCP packet transmission falls into becomes very large, from 0.5 times T_rr_interval up to 2.73 times the T_rr_interval. Therefore for configuration where one intends to have Td smaller than T_rr_interval, then Td is RECOMMENDED to be targeted at values less than 1/4th of T_rr_interval which results in that the range becomes [0.5*T_rr_interval, 1.81*T_rr_interval].

With RTP/AVPF using T_rr_interval of 0 or with another low value, which will be significantly lower than Td still has its utility and different behaviour compared to RTP/AVP. This avoids the Tmin limitations of RTP/AVP, thus allowing more frequent regular RTCP reporting. In fact this will result that the RTCP traffic becomes as high as the configured values.

(tbd: a future version of this memo will include examples of how to choose RTCP parameters for common scenarios)

There exist no method within the specification for using different regular RTCP reporting interval depending on media type or individual media stream.

7. Extension Considerations

This section discusses the impact on some RTP/RTCP extensions due to usage of multiple media types in on RTP session. Only extensions where something worth noting has been included.

7.1. RTP Retransmission

SSRC-multiplexed RTP retransmission [RFC4588] is actually very straightforward. Each retransmission RTP payload type is explicitly connected to an associated payload type. If retransmission is only to be used with a subset of all payload types, this is not a problem, as it will be evident from the retransmission payload types which payload types that have retransmission enabled for them.

Session-multiplexed RTP retransmission is also possible to use where an retransmission session contains the retransmissions of the associated payload types in the source RTP session. The only difference to previously is that the source RTP session is one which contains multiple media types. Thus it is even more likely that only a subset of the source RTP session's payload types and SSRCs are actually retransmitted.

Open Issue: When using SDP to signal retransmission for one RTP session with multiple media types and one RTP session for the retransmission data will cause a situation where one will have multiple m= lines grouped using FID and the ones belonging to respective RTP session being grouped using BUNDLE. This usage might contradict both the FID semantics [RFC5888] and an assumption in the RTP retransmission specification [RFC4588].

7.2. Generic FEC

The RTP Payload Format for Generic Forward Error Correction [RFC5109], and also its predecessor [RFC2733], requires some considerations, and they are different depending on what type of configuration of usage one has.

Independent RTP Sessions, i.e. where source and repair data are sent in different RTP sessions. As this mode of configuration requires different RTP session, there has to be at least one RTP session for source data, this session can be one using multiple media types. The repair session only needs one RTP Payload type indicating repair data, i.e. x/ulpfec or x/parityfec depending if RFC 5109 or RFC 2733 is used. The media type in this session is not relevant and can in theory be any of the defined ones. It is RECOMMENDED that one uses "Application".

In stream, using RTP Payload for Redundant Audio Data [RFC2198] combining repair and source data in the same packets. This is possible to use within a single RTP session. However, the usage and configuration of the payload types can create an issue. First of all it might be necessary to have one payload type per media type for the FEC repair data payload format, i.e. one for audio/ulpfec and one for text/ulpfec if audio and text are combined in an RTP session. Secondly each combination of source payload and its FEC repair data has to be an explicit configured payload type. This has potential for making the limitation of RTP payload types available into a real issue.

8. Signalling

The Signalling requirements

Establishing an RTP session with multiple media types requires signalling. This signalling needs to fulfil the following requirements:

  1. Ensure that any participant in the RTP session is aware that this is an RTP session with multiple media types.
  2. Ensure that the payload types in use in the RTP session are using unique values, with no overlap between the media types.
  3. Configure the RTP session level parameters, such as RTCP RR and RS bandwidth, AVPF trr-int, underlying transport, the RTCP extensions in use, and security parameters, commonly for the RTP session.
  4. RTP and RTCP functions that can be bound to a particular media type SHOULD be reused when possible also for other media types, instead of having to be configured for multiple code-points. Note: In some cases one will not have a choice but to use multiple configurations.

8.1. SDP-Based Signalling

The signalling of multiple media types in one RTP session in SDP is specified in "Multiplexing Negotiation Using Session Description Protocol (SDP) Port Numbers" [I-D.ietf-mmusic-sdp-bundle-negotiation].

9. IANA Considerations

This document makes no request of IANA.

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

10. Security Considerations

Having an RTP session with multiple media types doesn't change the methods for securing a particular RTP session. One possible difference is that the different media have often had different security requirements. When combining multiple media types in one session, their security requirements also have to be combined by selecting the most demanding for each property. Thus having multiple media types can result in increased overhead for security for some media types to ensure that all requirements are meet.

Otherwise, the recommendations for how to configure and RTP session do not add any additional requirements compared to normal RTP, except for the need to be able to ensure that the participants are aware that it is a multiple media type session. If not that is ensured it can cause issues in the RTP session for both the unaware and the aware one. Similar issues can also be produced in an normal RTP session by creating configurations for different end-points that doesn't match each other.

11. Acknowledgements

The authors would like to thank Christer Holmberg, Gunnar Hellström, and Charles Eckel for the feedback on the document.

12. References

12.1. 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.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video Conferences with Minimal Control", STD 65, RFC 3551, July 2003.
[I-D.ietf-mmusic-sdp-bundle-negotiation] Holmberg, C., Alvestrand, H. and C. Jennings, "Multiplexing Negotiation Using Session Description Protocol (SDP) Port Numbers", Internet-Draft draft-ietf-mmusic-sdp-bundle-negotiation-03, February 2013.
[I-D.lennox-avtcore-rtp-multi-stream] Lennox, J. and M. Westerlund, "Real-Time Transport Protocol (RTP) Considerations for Endpoints Sending Multiple Media Streams", Internet-Draft draft-lennox-avtcore-rtp-multi-stream-01, October 2012.

12.2. Informative References

[I-D.westerlund-avtcore-multiplex-architecture] Westerlund, M., Burman, B., Perkins, C. and H. Alvestrand, "Guidelines for using the Multiplexing Features of RTP", Internet-Draft draft-westerlund-avtcore-multiplex-architecture-02, July 2012.
[I-D.westerlund-avtcore-transport-multiplexing] Westerlund, M. and C. Perkins, "Multiple RTP Sessions on a Single Lower-Layer Transport", Internet-Draft draft-westerlund-avtcore-transport-multiplexing-04, October 2012.
[I-D.lennox-payload-ulp-ssrc-mux] Lennox, J., "Supporting Source-Multiplexing of the Real-Time Transport Protocol (RTP) Payload for Generic Forward Error Correction", Internet-Draft draft-lennox-payload-ulp-ssrc-mux-00, February 2013.
[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.
[RFC2733] Rosenberg, J. and H. Schulzrinne, "An RTP Payload Format for Generic Forward Error Correction", RFC 2733, December 1999.
[RFC4566] Handley, M., Jacobson, V. and C. Perkins, "SDP: Session Description Protocol", RFC 4566, July 2006.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C. and J. Rey, "Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July 2006.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V. and R. Hakenberg, "RTP Retransmission Payload Format", RFC 4588, July 2006.
[RFC5109] Li, A., "RTP Payload Format for Generic Forward Error Correction", RFC 5109, December 2007.
[RFC5117] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117, January 2008.
[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF)", RFC 5124, February 2008.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size Real-Time Transport Control Protocol (RTCP): Opportunities and Consequences", RFC 5506, April 2009.
[RFC5576] Lennox, J., Ott, J. and T. Schierl, "Source-Specific Media Attributes in the Session Description Protocol (SDP)", RFC 5576, June 2009.
[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and Control Packets on a Single Port", RFC 5761, April 2010.
[RFC5888] Camarillo, G. and H. Schulzrinne, "The Session Description Protocol (SDP) Grouping Framework", RFC 5888, June 2010.

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
Jonathan Lennox Vidyo, Inc. 433 Hackensack Avenue Seventh Floor Hackensack, NJ 07601 US EMail: jonathan@vidyo.com