| Network Working Group | M. Westerlund |
| Internet-Draft | B. Burman |
| Intended status: Informational | Ericsson |
| Expires: October 24, 2013 | C. Perkins |
| University of Glasgow | |
| H. Alvestrand | |
| April 22, 2013 |
Guidelines for using the Multiplexing Features of RTP
draft-ietf-avtcore-multiplex-guidelines-00
Real-time Transport Protocol (RTP) is a flexible protocol possible to use in a wide range of applications and network and system topologies. This flexibility and the implications of different choices should be understood by any application developer using RTP. To facilitate that understanding, this document contains an in-depth discussion of the usage of RTP's multiplexing points; the RTP session and the Synchronisation Source Identifier (SSRC). The document tries to give guidance and source material for an analysis on the most suitable choices for the application being designed.
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 http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress."
This Internet-Draft will expire on October 24, 2013.
Copyright (c) 2013 IETF Trust and the persons identified as the document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (http://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License.
Real-time Transport Protocol (RTP) [RFC3550] is a commonly used protocol for real-time media transport. It is a protocol that provides great flexibility and can support a large set of different applications. RTP has several multiplexing points designed for different purposes. These enable support of multiple media streams and switching between different encoding or packetization of the media. By using multiple RTP sessions, sets of media streams can be structured for efficient processing or identification. Thus the question for any RTP application designer is how to best use the RTP session, the SSRC and the payload type to meet the application's needs.
The purpose of this document is to provide clear information about the possibilities of RTP when it comes to multiplexing. The RTP application designer should understand the implications that come from a particular usage of the RTP multiplexing points. The document will recommend against some usages as being unsuitable, in general or for particular purposes.
RTP was from the beginning designed for multiple participants in a communication session. This is not restricted to multicast, as some may 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 re-distributing node can for example be a transport translator (relay) that forwards the packets unchanged, a translator performing media or protocol 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 endpoint have multiple media sources, like multiple cameras or microphones that need to be sent simultaneously.
This document has been written due to increased interest in more advanced usage of RTP, resulting in questions regarding the most appropriate RTP usage. The limitations in some implementations, RTP/RTCP extensions, and signalling has also been exposed. It is expected that some limitations will be addressed by updates or new extensions resolving the shortcomings. The authors also hope that clarification on the usefulness of some functionalities in RTP will result in more complete implementations in the future.
The document starts with some definitions and then goes into the existing RTP functionalities around multiplexing. Both the desired behaviour and the implications of a particular behaviour depend on which topologies are used, which requires some consideration. This is followed by a discussion of some choices in multiplexing behaviour and their impacts. Some arch-types of RTP usage are discussed. Finally, some recommendations and examples are provided.
This document is currently an individual contribution, but it is the intention of the authors that this should become a WG document that objectively describes and provides suitable recommendations for which there is WG consensus. Currently this document only represents the views of the authors. The authors gladly accept any feedback on the document and will be happy to discuss suitable recommendations.
The following terms and abbreviations are used in this document:
This document is focused on issues that affect RTP. Thus, issues that involve signalling protocols, such as whether SIP, Jingle or some other protocol is in use for session configuration, the particular syntaxes used to define RTP session properties, or the constraints imposed by particular choices in the signalling protocols, are mentioned only as examples in order to describe the RTP issues more precisely.
This document assumes the applications will use RTCP. While there are such applications that don't send RTCP, they do not conform to the RTP specification, and thus should be regarded as reusing the RTP packet format, not as implementing the RTP protocol.
This section describes the existing RTP tools that are particularly important when discussing multiplexing of different media streams.
The RTP Session is the highest semantic level in RTP and contains all of the RTP functionality. RTP itself has no normative statements about the relationship between different RTP sessions.
A RTP source in an RTP session 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 the RTP specification [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 endpoints and any interconnecting middle nodes.
An SSRC identifies a RTP source or a Media Sink. For end-points that both source and sink media streams its SSRCs are used in both roles. At any given time, a RTP source has one and only one SSRC - although that may change over the lifetime of the RTP source or sink. An RTP Session serves one or more RTP sources.
An SSRC identifier is used by different type of sources as well as sinks:
Note that a endpoint 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 endpoint. The main property attributed to SSRCs associated with the same CNAME is that they are from a particular synchronisation context and may be synchronised at playback.
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.
The Contributing Source (CSRC) is not a separate identifier, but an usage of the SSRC identifier. It is optionally included in the RTP header as list of up to 15 contributing RTP sources. CSRC 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. The exception is the case when only a single CSRC is indicated as this represent forwarding of a media stream, possibly modified. The RTP header extension for Mixer-to-Client Audio Level Indication [RFC6465] expands on the receivers information about a packet with a CSRC list. Due to these restrictions, CSRC will not be considered a fully qualified multiplex point and will be disregarded in the rest of this document.
Each Media Stream utilises one or more encoding formats, identified by the Payload Type.
The Payload Type is not a multiplexing point. Appendix A gives some of the many reasons why attempting to use it as a multiplexing point will have bad results.
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 is 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].
The reasons why an endpoint may choose to send multiple media streams are widespread. 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:
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.
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.
As the below discussion will show, in reality we cannot choose a single one of the two solutions. To utilise 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.
The impact of how RTP Multiplex is performed will in general vary with how the RTP Session participants are interconnected, described by RTP Topology [RFC5117] and its intended successor [I-D.westerlund-avtcore-rtp-topologies-update].
Even the most basic use case, denoted Topo-Point-to-Point in [I-D.westerlund-avtcore-rtp-topologies-update], raises a number of considerations that are discussed in detail below [sec-discussion]. They range over such aspects as:
The application designer will have to make choices here. The point to point topology can contain one to many RTP sessions with one to many media sources per session, resulting in one or more RTP source (SSRC) per media source.
A point to point communication can end up in a situation when the peer it is communicating with is not compatible with the other peer for various reasons:
This is in many situations resolved by the inclusion of a translator in-between the two peers, as described by Topo-PtP-Translator in [I-D.westerlund-avtcore-rtp-topologies-update]. The translator's main purpose is to make the peer look to the other peer like something it is compatible with. There may also be other reasons than compatibility to insert a translator in the form of a middlebox or gateway, for example a need to monitor the media streams. If the stream transport characteristics are changed by the translator, appropriate media handling can require thorough understanding of the application logic, specifically any congestion control or media adaptation.
This section discusses the Point to Multi-point using Multicast to interconnect the session participants. This includes both Topo-ASM and Topo-SSM in [I-D.westerlund-avtcore-rtp-topologies-update].
Special considerations must be made as multicast is a one to many distribution system. For example, the only practical method for adapting the bit-rate sent towards a given receiver for large groups is to use a set of multicast groups, where each multicast group represents a particular bit-rate. Otherwise the whole group gets media adapted to the participant with the worst conditions. 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 streams that improve transport robustness are sent in their own multicast group to allow for interworking with legacy or to support different levels of protection.
The result of this is some common behaviours for RTP multicast:
All multicast configurations share a signalling requirement; all of the participants will need to have the same RTP and payload type configuration. Otherwise, A could for example be using payload type 97 as the video codec H.264 while B thinks it is MPEG-2. It should be noted that SDP offer/answer [RFC3264] has issues with ensuring this property. The signalling aspects of multicast are not explored further in this memo.
Security solutions for this type of group communications are also challenging. First of all the key-management and the security protocol must support group communication. Source authentication becomes more difficult and requires special solutions. For more discussion on this please review Options for Securing RTP Sessions [I-D.ietf-avtcore-rtp-security-options].
This mode is described as Topo-Translator in [I-D.westerlund-avtcore-rtp-topologies-update].
Transport Translators (Relays) result in an RTP session situation that is very similar to how an ASM group RTP session would behave.
One of the most important aspects with the simple relay is that it is only rewriting transport headers, no RTP modifications nor media transcoding occur. The most obvious downside of this basic relaying is that the translator has no control over how many streams need to be delivered to a receiver. Nor can it simply select to deliver only certain streams, as this creates session inconsistencies: If the translator temporarily stops a stream, this prevents some receivers from reporting on it. From the sender's perspective it will look like a transport failure. Applications having needs to stop or switch streams in the central node should consider using an RTP mixer to avoid this issue.
The Transport Translator has the same signalling requirement as multicast: All participants must have the same payload type configuration. Most of the ASM security issues also arise here. Some alternative when it comes to solution do exist as there after all exist a central node to communicate with. One that also can enforce some security policies depending on the level of trust placed in the node.
A mixer, described by Topo-Mixer in [I-D.westerlund-avtcore-rtp-topologies-update], is a centralised node that selects or mixes content in a conference to optimise the RTP session so that each endpoint only needs connect to one entity, the mixer. The media sent from the mixer to the end-point can be optimised in different ways. These optimisations include methods like only choosing media from the currently most active speaker or mixing together audio so that only one audio stream is required.
Mixers have some downsides, the first is that the mixer must be a trusted node as they either perform media operations or at least repacketize the media. When using SRTP, both media operations and repacketization requires that the mixer verifies integrity, decrypts the content, performs the operation and forms new RTP packets, encrypts and integrity-protects them. This applies to all types of mixers. The second downside is that all these operations and optimisations of the session requires processing. How much depends on the implementation, as will become evident below.
A mixer, unlike a pure transport translator, is always application specific: the application logic for stream mixing or stream selection has to be embedded within the mixer, and controlled using application specific signalling. The implementation of a mixer can take several different forms and we will discuss the main themes available that doesn't break RTP.
Please note that a Mixer could also contain translator functionalities, like a media transcoder to adjust the media bit-rate or codec used for a particular RTP media stream.
Using multiple media streams is a well supported feature of RTP. However, it can be unclear for most implementers or people writing RTP/RTCP applications or extensions attempting to apply multiple streams 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.
This section discusses RTP and RTCP aspects worth considering when selecting between using an additional SSRC and Multiple RTP sessions.
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 minimised, 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:
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 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 Appendix A.
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. As we saw in the discussion of RTP mixers, the RTP mixer has to embed application logic in order to handle streams anyway; the separation of streams according to stream type is just another piece of application logic, which may or may not be appropriate for a particular application. A type of application that can mix different media sources "blindly" is the audio only "telephone" bridge; most other type of application needs application-specific logic to perform the mix correctly.
The fifth argument discusses network aspects that we will discuss more below in Section 6.4. It also goes into aspects of implementation, like decomposed endpoints 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 to use different RTP sessions for media streams that don't share media type. The first this document support as very valid. The later is one thing which is further discussed in this document as something the application developer needs to make a conscious choice for.
The above quote from RTP [RFC3550] includes a strong recommendation:
It was identified in "Why RTP Sessions Should Be Content Neutral" [I-D.alvestrand-rtp-sess-neutral] that the above statement is poorly supported by any of the motivations provided in the RTP specification. This has resulted in the creation of a specification Multiple Media Types in an RTP Session specification [I-D.ietf-avtcore-multi-media-rtp-session] which intend to update this recommendation. That document has a detailed analysis of the potential issues in having multiple media types in the same RTP session. This document tries to provide an more over arching consideration regarding the usage of RTP session and considers multiple media types in one RTP session as possible choice for the RTP application designer.
Using multiple SSRCs in an RTP session at one endpoint has some unclarities in the RTP specification. These could potentially lead to some interoperability issues as well as some potential significant inefficencies. These are further discussed in "RTP Considerations for Endpoints Sending Multiple Media Streams" [I-D.lennox-avtcore-rtp-multi-stream]. A application designer may need to consider these issues and the impact availability or lack of the optimization in the endpoints has on their application.
If an application will become affected by the issues described, using Multiple RTP sessions can mitigate these issues.
In some applications, the set of simultaneously active sources varies within a larger set of session members. A receiver can then possibly try to use a set of decoding chains that is smaller than the number of senders, switching the decoding chains between different senders. As each media decoding chain may contain state, either the receiver must either be able to save the state of swapped-out senders, or the sender must be able to send data that permits the receiver to reinitialise when it resumes activity.
This behaviour will cause similar issues independent of Additional SSRC or Multiple RTP session.
There currently exists no functionality to make truly synchronised and atomic RTCP messages with some type of request semantics across multiple RTP Sessions. Instead, separate RTCP messages will have to be sent in each session. This gives streams in the same RTP session a slight advantage as RTCP messages 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.
When using multiple RTP sessions, 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, forcing it to use more robust and forgiving mechanisms.
A common problem in a number of various RTP extensions has been how to bind related RTP sources and their media streams together. This issue is common to both using additional SSRCs and Multiple RTP sessions.
The solutions can be divided into some groups, RTP/RTCP based, Signalling based (SDP), grouping related RTP sessions, and grouping SSRCs within an RTP session. Most solutions are explicit, but some implicit methods have also been applied to the problem.
The SDP-based signalling solutions are:
This supports a lot of use cases. Both solutions have shortcomings 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. As they are two related but still separated solutions it is not well specified to group SSRCs across multiple RTP sessions and SDP media descriptions.
Within RTP/RTCP based solutions when binding to a endpoint or synchronization context, i.e. the CNAME has not be sufficient and one has multiple RTP sessions has been to using the same SSRC value across all the RTP sessions. RTP Retransmission [RFC4588] is multiple RTP session mode, Generic FEC [RFC5109], as well as the RTP payload format for Scalable Video Coding [RFC6190] in Multi Session Transmission (MST) mode uses this method. This method clearly works but might have some downside in RTP sessions with many participating SSRCs. The birthday paradox ensures that if you populate a single session with 9292 SSRCs at random, the chances are approximately 1% that at least one collision will occur. When a collision occur this will force one to change SSRC in all RTP sessions and thus resynchronizing all of them instead of only the single media stream having the collision.
It can be noted that Section 8.3 of the RTP Specification [RFC3550] recommends using a single SSRC space across all RTP sessions for layered coding.
Another solution that has been applied to binding SSRCs have been an implicit method used by RTP Retransmission [RFC4588] when doing retransmissions in the same RTP session as the source RTP media stream. This issues an RTP retransmission request, and then await a new SSRC carrying the RTP retransmission payload and where that SSRC is from the same CNAME. This limits a requestor to having only one outstanding request on any new source SSRCs per endpoint.
There exist no RTP/RTCP based mechanism capable of supporting explicit association accross multiple RTP sessions as well within an RTP session. A proposed solution for handling this issue is [I-D.westerlund-avtext-rtcp-sdes-srcname]. This can potentially be part of an SDP based solution also by reusing the same identifiers and name space.
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 create 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 schemes using multiple SSRCs and multiple RTP sessions, and some schemes that support both modes of operation.
Using multiple RTP sessions supports the case where some set of receivers may not be able to utilise 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 therefore in its own 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 caught 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 allows for easy separation in the Burst/Retransmission Source (BRS) without having to individually classify packets.
A basic Transport Translator relays any incoming RTP and RTCP packets to the other participants. The main difference between Additional SSRCs and Multiple RTP Sessions resulting from this use case is that with Additional SSRCs 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.
There are several different kinds of interworking, and this section discusses two related ones. The interworking between different applications and the implications of potentially different choices of usage of RTP's multiplexing points. The second topic relates to what limitations may have to be considered working with some legacy applications.
It is not uncommon that applications or services of similar usage, especially the ones intended for interactive communication, ends up in a situation where one want to interconnect two or more of these applications.
In these cases one ends up in a situation where one might use a gateway to interconnect applications. This gateway then needs to change the multiplexing structure or adhere to limitations in each application.
There are two fundamental approaches to gatewaying: RTP bridging, where the gateway acts as an RTP Translator, and the two applications are members of the same RTP session, and RTP termination, where there are independent RTP sessions running from each interconnected application to the gateway.
From an RTP perspective the RTP Translator approach could work if all the applications are using the same codecs with the same payload types, have made the same multiplexing choices, have the same capabilities in number of simultaneous media streams combined with the same set of RTP/RTCP extensions being supported. Unfortunately this may not always be true.
When one is gatewaying via an RTP Translator, a natural requirement is that the two applications being interconnected must use the same approach to multiplexing. Furthermore, if one of the applications is capable of working in several modes (such as being able to use Additional SSRCs or Multiple RTP sessions at will), and the other one is not, successful interconnection depends on locking the more flexible application into the operating mode where interconnection can be successful, even if no participants using the less flexible application are present when the RTP sessions are being created.
When one terminates RTP sessions at the gateway, there are certain tasks that the gateway must carry out:
If either of the applications has any security applied, e.g. in the form of SRTP, the gateway must be able to decrypt incoming packets and re-encrypt them in the other application's security context. This is necessary even if all that's required is a simple remapping of SSRC numbers. If this is done, the gateway also needs to be a member of the security contexts of both sides, of course.
Other tasks a gateway may need to apply include transcoding (for incompatible codec types), rescaling (for incompatible video size requirements), suppression of content that is known not to be handled in the destination application, or the addition or removal of redundancy coding or scalability layers to fit the need of the destination domain.
From the above, we can see that the gateway needs to have an intimate knowledge of the application requirements; a gateway is by its nature application specific, not a commodity product.
This fact reveals the potential for these gateways to block evolution of the applications by blocking unknown RTP and RTCP extensions that the regular application has been extended with.
If one uses security functions, like SRTP, they can as seen above incur both additional risk due to the gateway needing to be in security association between the endpoints, unless the gateway is on the transport level, and additional complexities in form of the decrypt-encrypt cycles needed for each forwarded packet. SRTP, due to its keying structure, also requires that each RTP session must have different master keys, as use of the same key in two RTP sessions can result in two-time pads that completely breaks the confidentiality of the packets.
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 endpoint 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 endpoint and the RTP mixer, meaning that the mixer functions as an RTP-terminating gateway.
When establishing RTP sessions that may contain endpoints 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:
This indicates that gateways attempting to interconnect to this class of devices must make sure that only one media stream of each type gets delivered to the endpoint if it's expecting only one, and that the multiplexing format is what the device expects. It is highly unlikely that RTP translator-based interworking can be made to function successfully in such a context.
The multiplexing choice has impact on network level mechanisms that need to be considered by the implementor.
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. Additional SSRC 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 when using multiple RTP sessions, where each media stream for which a separate QoS handling is desired can be in a different RTP session that can be sent over different 5-tuples.
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:
Additional SSRC keeps the additional media streams within one RTP Session and transport flow and does not introduce any additional NAT traversal complexities per media stream. This can be compared with normally one or two additional transport flows per RTP session when using multiple RTP sessions. 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-transport-multiplexing].
Multicast groups provides a powerful semantics for a number of real-time applications, especially the ones that desire broadcast-like behaviours with one endpoint 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 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.
RTP can't function correctly if media streams sent over different multicast groups where considered part of the same RTP session. First of all the different layers needs different SSRCs or the sequence number space seen for a receiver of any sub set of the layers would have sender side holes. Thus triggering packet loss reactions. Also any RTCP reporting of such a session would be non consistent and making it difficult for the sender to determine the sessions actual state.
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.
For applications that doesn't need flow based QoS and like to save ports and NAT/FW traversal costs and where usage of multiple media types in one RTP session is not suitable, there is a proposal for how to achieve multiplexing of multiple RTP sessions over the same lower layer transport [I-D.westerlund-avtcore-transport-multiplexing]. Using such a solution would allow Multiple RTP session without most of the perceived downsides of Multiple RTP sessions creating a need for additional transport flows.
When dealing with point-to-point, 2-member RTP sessions only, there are few security issues that are relevant to the choice of having one RTP session or multiple RTP sessions. However, there are a few aspects of multiparty sessions that might warrant consideration. For general information of possible methods of securing RTP, please review RTP Security Options [I-D.ietf-avtcore-rtp-security-options].
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 no one 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.
SRTP [RFC3711] has not special functions for dealing with different sets of master keys for different SSRCs. The key-management functions has different capabilities to establish different set of keys, normally on a per end-point basis. DTLS-SRTP [RFC5764] and Security Descriptions [RFC4568] for example establish different keys for outgoing and incoming traffic from an end-point. This key usage must be written into the cryptographic context, possibly associated with different SSRCs.
Performing key-management for multi-party session can be a challenge. This section considers some of the issues.
Multi-party sessions, such as transport translator based sessions and multicast sessions, cannot use Security Description [RFC4568] nor DTLS-SRTP [RFC5764] without an extension as each endpoint provides its set of keys. In centralised conference, the signalling counterpart is a conference server and the media plane unicast counterpart (to which DTLS messages would be sent) is the transport translator. Thus an extension like Encrypted Key Transport [I-D.ietf-avt-srtp-ekt] is needed or a MIKEY [RFC3830] based solution that allows for keying all session participants with the same master key.
The usage of security functions can surface complexity implications of the choice of multiplexing and topology. This becomes especially evident in RTP topologies having any type of middlebox that processes or modifies RTP/RTCP packets. Where there is very small overhead for an RTP translator or mixer to rewrite an SSRC value in the RTP packet of an unencrypted session, the cost of doing it when using cryptographic security functions is higher. For example if using SRTP [RFC3711], the actual security context and exact crypto key are determined by the SSRC field value. If one changes it, the encryption and authentication tag must be performed using another key. Thus changing the SSRC value implies a decryption using the old SSRC and its security context followed by an encryption using the new one.
This section discusses some arch-types of how RTP multiplexing can be used in applications to achieve certain goals and a summary of their implications. For each arch-type there is discussion of benefits and downsides.
In this arch-type each endpoint in a point-to-point session has only a single SSRC, thus the RTP session contains only two SSRCs, one local and one remote. This session can be used both unidirectional, i.e. only a single media stream or bi-directional, i.e. both endpoints have one media stream each. If the application needs additional media flows between the endpoints, they will have to establish additional RTP sessions.
The Pros:
The Cons:
RTP applications that need to inter-work with legacy RTP applications, like VoIP and video conferencing, can potentially benefit from this structure. However, a large number of media descriptions in SDP can also run into issues with existing implementations. For any application needing a larger number of media flows, the overhead can become very significant. This structure is also not suitable for multi-party sessions, as any given media stream from each participant, although having same usage in the application, must have its own RTP session. In addition, the dynamic behaviour that can arise in multi-party applications can tax the signalling system and make timely media establishment more difficult.
In this arch-type, each RTP session serves only a single media type. The RTP session can contain multiple media streams, either from a single endpoint or due to multiple endpoints. This commonly creates a low number of RTP sessions, typically only two one for audio and one for video with a corresponding need for two listening ports when using RTP and RTCP multiplexing.
The Pros:
The Cons:
For RTP applications where all media streams of the same media type share same usage, this structure provides efficiency gains in amount of network state used and provides more faith sharing with other media flows of the same type. At the same time, it is still maintaining almost all functionalities when it comes to negotiation in the signalling of the properties for the individual media type and also enabling flow based QoS prioritisation between media types. It handles multi-party session well, independently of multicast or centralised transport distribution, as additional sources can dynamically enter and leave the session.
In this arch-type one goes one step further than in the above [sec-multiple-ssrc-single-session] by using multiple RTP sessions also for a single media type. The main reason for going in this direction is that the RTP application needs separation of the media streams due to their usage. Some typical reasons for going to this arch-type are scalability over multicast, simulcast, need for extended QoS prioritisation of media streams due to their usage in the application, or the need for fine granular signalling using today's tools.
The Pros:
The Cons:
For more complex RTP applications that have several different usages for media streams of the same media type and / or uses scalability or simulcast, this solution can enable those functions at the cost of increased overhead associated with the additional sessions. This type of structure is suitable for more advanced applications as well as multicast based applications requiring differentiation to different participants.
This arch-type is to use a single RTP session for multiple different media types, like audio and video, and possibly also transport robustness mechanisms like FEC or Retransmission. Each media stream will use its own SSRC and a given SSRC value from a particular endpoint will never use the SSRC for more than a single media type.
The Pros:
The Cons:
There are some clear relations between these arch-types. Both the "single SSRC per RTP session" and the "multiple media types in one session" are cases which require full explicit signalling of the media stream relations. However, they operate on two different levels where the first primarily enables session level binding, and the second needs to do it all on SSRC level. From another perspective, the two solutions are the two extreme points when it comes to number of RTP sessions required.
The two other arch-types "Multiple SSRCs of the Same Media Type" and "Multiple Sessions for one Media Type" are examples of two other cases that first of all allows for some implicit mapping of the role or usage of the media streams based on which RTP session they appear in. It thus potentially allows for less signalling and in particular reduced need for real-time signalling in dynamic sessions. They also represent points in between the first two when it comes to amount of RTP sessions established, i.e. representing an attempt to reduce the amount of sessions as much as possible without compromising the functionality the session provides both on network level and on signalling level.
This section contains a number of recommendations for implementors or specification writers when it comes to handling multi-stream.
This document makes no request of IANA.
Note to RFC Editor: this section may be removed on publication as an RFC.
There is discussion of the security implications of choosing SSRC vs Multiple RTP session in Section 6.5.
| [RFC3550] | Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", STD 64, RFC 3550, July 2003. |
This section documents a number of reasons why using the payload type as a multiplexing point for most things related to multiple streams is unsuitable. 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:
The above discussion and guidelines indicates that a small set of extension mechanisms could greatly improve the situation when it comes to using multiple streams independently of Multiple RTP session or Additional SSRC. These extensions are:
This section describes a number of clarifications to the RTP specifications that are likely necessary for aligned behaviour when RTP sessions contain more SSRCs than one local and one remote.
All of the below proposals are under consideration in [I-D.lennox-avtcore-rtp-multi-stream].
When one has multiple SSRC in an RTP node, all these SSRC must send some RTP or RTCP packet 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; any SSRC that intends to remain in the session must send some packets to avoid timing out according to the rules in RFC 3550 section 6.3.5.
It has been hypothesised that a third party monitor may be confused by not necessarily being able to determine that all these SSRC are in fact co-located and originate from the same stack instance; if this hypothesis is true, this may argue for having all the sources send full reception reports, even though they are reporting the same packet delivery.
The contrary argument is that such double reporting may confuse the third party monitor even more by making it seem that utilisation of the last-hop link to the recipient is (number of SSRCs) times higher than what it actually is.
For any RTP node that sends more than one SSRC, there is 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. The hypothetical 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 the same node.
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.
Signalling is not an architectural consideration for RTP itself, so this discussion has been moved to an appendix. However, it is hugely important for anyone building complete applications, so it is deserving of discussion.
The issues raised here need to be addressed in the WGs that deal with signalling; they cannot be addressed by tweaking, extending or profiling RTP.
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, while 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.
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 those are not necessarily strictly bound to an RTP session and could be of interest to signal specifically for a particular media stream (SSRC) within the session. The following properties have been identified as being potentially useful to signal not only on RTP session level:
Some of these issues are clearly SDP's problem rather than RTP limitations. However, if the aim is to deploy an solution using additional SSRCs 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 Additional SSRC only, 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 application's requirements.
For some parameters, such as resolution and framerate, a SSRC-linked mechanism has been proposed: [I-D.lennox-mmusic-sdp-source-selection].
SDP chose to use the m= line both to delineate an RTP session and to specify the top level of the MIME 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. This binding has to be loosened in order to use SDP to describe RTP sessions containing multiple MIME top level types.
There is an accepted WG item in the MMUSIC WG to define how multiple media lines describe a single underlying transport [I-D.ietf-mmusic-sdp-bundle-negotiation] and thus it becomes possible in SDP to define one RTP session with media types having different MIME top level types.
Media streams being transported in RTP has some particular usage in an RTP application. This usage of the media stream is in many applications so far implicitly signalled. For example, an application may choose to take all incoming audio RTP streams, mix them and play them out. However, in more advanced applications that use multiple media streams there will be more than a single usage or purpose among the set of media streams being sent or received. RTP applications will need to signal this usage somehow. The signalling used will have to identify the media streams affected by their RTP-level identifiers, which means that they have to be identified either by their session or by their SSRC + session.
In some applications, the receiver cannot utilise the media stream at all before it has received the signalling message describing the media stream and its usage. In other applications, there exists a default handling that is appropriate.
If all media streams in an RTP session are to be treated in the same way, identifying the session is enough. If SSRCs in a session are to be treated differently, signalling must identify both the session and the SSRC.
If this signalling affects how any RTP central node, like an RTP mixer or translator that selects, mixes or processes streams, treats the streams, the node will also need to receive the same signalling to know how to treat media streams with different usage in the right fashion.