| RTCWEB Working Group | C. S. Perkins |
| Internet-Draft | University of Glasgow |
| Intended status: Standards Track | M. Westerlund |
| Expires: June 19, 2014 | Ericsson |
| J. Ott | |
| Aalto University | |
| December 16, 2013 |
Web Real-Time Communication (WebRTC): Media Transport and Use of RTP
draft-ietf-rtcweb-rtp-usage-11
The Web Real-Time Communication (WebRTC) framework provides support for direct interactive rich communication using audio, video, text, collaboration, games, etc. between two peers' web-browsers. This memo describes the media transport aspects of the WebRTC framework. It specifies how the Real-time Transport Protocol (RTP) is used in the WebRTC context, and gives requirements for which RTP features, profiles, and extensions need to be supported.
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The Real-time Transport Protocol (RTP) [RFC3550] provides a framework for delivery of audio and video teleconferencing data and other real-time media applications. Previous work has defined the RTP protocol, along with numerous profiles, payload formats, and other extensions. When combined with appropriate signalling, these form the basis for many teleconferencing systems.
The Web Real-Time communication (WebRTC) framework provides the protocol building blocks to support direct, interactive, real-time communication using audio, video, collaboration, games, etc., between two peers' web-browsers. This memo describes how the RTP framework is to be used in the WebRTC context. It proposes a baseline set of RTP features that are to be implemented by all WebRTC-aware end-points, along with suggested extensions for enhanced functionality.
This memo specifies a protocol intended for use within the WebRTC framework, but is not restricted to that context. An overview of the WebRTC framework is given in [I-D.ietf-rtcweb-overview].
The structure of this memo is as follows. Section 2 outlines our rationale in preparing this memo and choosing these RTP features. Section 3 defines terminology. Requirements for core RTP protocols are described in Section 4 and suggested RTP extensions are described in Section 5. Section 6 outlines mechanisms that can increase robustness to network problems, while Section 7 describes congestion control and rate adaptation mechanisms. The discussion of mandated RTP mechanisms concludes in Section 8 with a review of performance monitoring and network management tools that can be used in the WebRTC context. Section 9 gives some guidelines for future incorporation of other RTP and RTP Control Protocol (RTCP) extensions into this framework. Section 10 describes requirements placed on the signalling channel. Section 11 discusses the relationship between features of the RTP framework and the WebRTC application programming interface (API), and Section 12 discusses RTP implementation considerations. The memo concludes with security considerations [sec-security] and IANA considerations [sec-iana].
The RTP framework comprises the RTP data transfer protocol, the RTP control protocol, and numerous RTP payload formats, profiles, and extensions. This range of add-ons has allowed RTP to meet various needs that were not envisaged by the original protocol designers, and to support many new media encodings, but raises the question of what extensions are to be supported by new implementations. The development of the WebRTC framework provides an opportunity for us to review the available RTP features and extensions, and to define a common baseline feature set for all WebRTC implementations of RTP. This builds on the past 20 years development of RTP to mandate the use of extensions that have shown widespread utility, while still remaining compatible with the wide installed base of RTP implementations where possible.
Other RTP and RTCP extensions not discussed in this document can be implemented by WebRTC end-points if they are beneficial for new use cases. However, they are not necessary to address the WebRTC use cases and requirements identified to date [I-D.ietf-rtcweb-use-cases-and-requirements].
While the baseline set of RTP features and extensions defined in this memo is targeted at the requirements of the WebRTC framework, it is expected to be broadly useful for other conferencing-related uses of RTP. In particular, it is likely that this set of RTP features and extensions will be appropriate for other desktop or mobile video conferencing systems, or for room-based high-quality telepresence applications.
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]. The RFC 2119 interpretation of these key words applies only when written in ALL CAPS. Lower- or mixed-case uses of these key words are not to be interpreted as carrying special significance in this memo.
We define the following terms:
Other terms are used according to their definitions from the RTP Specification [RFC3550].
The following sections describe the core features of RTP and RTCP that need to be implemented, along with the mandated RTP profiles and payload formats. Also described are the core extensions providing essential features that all WebRTC implementations need to implement to function effectively on today's networks.
The Real-time Transport Protocol (RTP) [RFC3550] is REQUIRED to be implemented as the media transport protocol for WebRTC. RTP itself comprises two parts: the RTP data transfer protocol, and the RTP control protocol (RTCP). RTCP is a fundamental and integral part of RTP, and MUST be implemented in all WebRTC applications.
The following RTP and RTCP features are sometimes omitted in limited functionality implementations of RTP, but are REQUIRED in all WebRTC implementations:
It is known that a significant number of legacy RTP implementations, especially those targeted at VoIP-only systems, do not support all of the above features, and in some cases do not support RTCP at all. Implementers are advised to consider the requirements for graceful degradation when interoperating with legacy implementations.
Other implementation considerations are discussed in Section 12.
The complete specification of RTP for a particular application domain requires the choice of an RTP Profile. For WebRTC use, the Extended Secure RTP Profile for RTCP-Based Feedback (RTP/SAVPF) [RFC5124], as extended by [RFC7007], MUST be implemented. This builds on the basic RTP/AVP profile [RFC3551], the RTP profile for RTCP-based feedback (RTP/AVPF) [RFC4585], and the secure RTP profile (RTP/SAVP) [RFC3711].
The RTCP-based feedback extensions [RFC4585] are needed for the improved RTCP timer model, that allows more flexible transmission of RTCP packets in response to events, rather than strictly according to bandwidth. This is vital for being able to report congestion events. These extensions also save RTCP bandwidth, and will commonly only use the full RTCP bandwidth allocation if there are many events that require feedback. They are also needed to make use of the RTP conferencing extensions discussed in Section 5.1.
The secure RTP profile [RFC3711] is needed to provide media encryption, integrity protection, replay protection and a limited form of source authentication. WebRTC implementations MUST NOT send packets using the basic RTP/AVP profile or the RTP/AVPF profile; they MUST employ the full RTP/SAVPF profile to protect all RTP and RTCP packets that are generated. The default and mandatory to implement transforms listed in Section 5 of [RFC3711] SHALL apply.
The keying mechanism(s) to be used with the RTP/SAVPF profile are defined in Section 5.5 of [I-D.ietf-rtcweb-security-arch] or its replacement.
The set of mandatory to implement codecs and RTP payload formats for WebRTC is not specified in this memo. Implementations can support any codec for which an RTP payload format and associated signalling is defined. Implementation cannot assume that the other participants in an RTP session understand any RTP payload format, no matter how common; the mapping between RTP payload type numbers and specific configurations of particular RTP payload formats MUST be agreed before those payload types/formats can be used. In an SDP context, this can be done using the "a=rtpmap:" and "a=fmtp:" attributes associated with an "m=" line.
Endpoints can signal support for multiple RTP payload formats, or multiple configurations of a single RTP payload format, as long as each unique RTP payload format configuration uses a different RTP payload type number. As outlined in Section 4.8, the RTP payload type number is sometimes used to associate an RTP media stream with a signalling context. This association is possible provided unique RTP payload type numbers are used in each context. For example, an RTP media stream can be associated with an SDP "m=" line by comparing the RTP payload type numbers used by the media stream with payload types signalled in the "a=rtpmap:" lines in the media sections of the SDP. If RTP media streams are being associated with signalling contexts based on the RTP payload type, then the assignment of RTP payload type numbers MUST be unique across signalling contexts; if the same RTP payload format configuration is used in multiple contexts, then a different RTP payload type number has to be assigned in each context to ensure uniqueness. If the RTP payload type number is not being used to associated RTP media streams with a signalling context, then the same RTP payload type number can be used to indicate the exact same RTP payload format configuration in multiple contexts.
An endpoint that has signalled support for multiple RTP payload formats SHOULD accept data in any of those payload formats at any time, unless it has previously signalled limitations on its decoding capability. This requirement is constrained if several types of media (e.g., audio and video) are sent in the same RTP session. In such a case, a source (SSRC) is restricted to switching only between the RTP payload formats signalled for the type of media that is being sent by that source; see Section 4.4. To support rapid rate adaptation by changing codec, RTP does not require advance signalling for changes between RTP payload formats that were signalled during session set-up.
An RTP sender that changes between two RTP payload types that use different RTP clock rates MUST follow the recommendations in Section 4.1 of [I-D.ietf-avtext-multiple-clock-rates]. RTP receivers MUST follow the recommendations in Section 4.3 of [I-D.ietf-avtext-multiple-clock-rates], in order to support sources that switch between clock rates in an RTP session (these recommendations for receivers are backwards compatible with the case where senders use only a single clock rate).
An association amongst a set of participants communicating using RTP is known as an RTP session. A participant can be involved in several RTP sessions at the same time. In a multimedia session, each type of media has typically been carried in a separate RTP session (e.g., using one RTP session for the audio, and a separate RTP session using different transport addresses for the video). WebRTC implementations of RTP are REQUIRED to implement support for multimedia sessions in this way, separating each session using different transport-layer addresses (e.g., different UDP ports) for compatibility with legacy systems.
In modern day networks, however, with the widespread use of network address/port translators (NAT/NAPT) and firewalls, it is desirable to reduce the number of transport-layer flows used by RTP applications. This can be done by sending all the RTP media streams in a single RTP session, which will comprise a single transport-layer flow (this will prevent the use of some quality-of-service mechanisms, as discussed in Section 12.1.3). Implementations are REQUIRED to support transport of all RTP media streams, independent of media type, in a single RTP session according to [I-D.ietf-avtcore-multi-media-rtp-session]. If multiple types of media are to be used in a single RTP session, all participants in that session MUST agree to this usage. In an SDP context, [I-D.ietf-mmusic-sdp-bundle-negotiation] can be used to signal this.
It is also possible to use a shim-based approach to run multiple RTP sessions on a single transport-layer flow. This gives advantages in some gateway scenarios, and makes it easy to distinguish groups of RTP media streams that might need distinct processing. One way of doing this is described in [I-D.westerlund-avtcore-transport-multiplexing]. At the time of this writing, there is no consensus to use a shim-based approach in WebRTC implementations.
Further discussion about when different RTP session structures and multiplexing methods are suitable can be found in [I-D.ietf-avtcore-multiplex-guidelines].
Historically, RTP and RTCP have been run on separate transport layer addresses (e.g., two UDP ports for each RTP session, one port for RTP and one port for RTCP). With the increased use of Network Address/Port Translation (NAPT) this has become problematic, since maintaining multiple NAT bindings can be costly. It also complicates firewall administration, since multiple ports need to be opened to allow RTP traffic. To reduce these costs and session set-up times, support for multiplexing RTP data packets and RTCP control packets on a single port for each RTP session is REQUIRED, as specified in [RFC5761]. For backwards compatibility, implementations are also REQUIRED to support RTP and RTCP sent on separate transport-layer addresses.
Note that the use of RTP and RTCP multiplexed onto a single transport port ensures that there is occasional traffic sent on that port, even if there is no active media traffic. This can be useful to keep NAT bindings alive, and is the recommend method for application level keep-alives of RTP sessions [RFC6263].
RTCP packets are usually sent as compound RTCP packets, and [RFC3550] requires that those compound packets start with an Sender Report (SR) or Receiver Report (RR) packet. When using frequent RTCP feedback messages under the RTP/AVPF Profile [RFC4585] these statistics are not needed in every packet, and unnecessarily increase the mean RTCP packet size. This can limit the frequency at which RTCP packets can be sent within the RTCP bandwidth share.
To avoid this problem, [RFC5506] specifies how to reduce the mean RTCP message size and allow for more frequent feedback. Frequent feedback, in turn, is essential to make real-time applications quickly aware of changing network conditions, and to allow them to adapt their transmission and encoding behaviour. Support for non-compound RTCP feedback packets [RFC5506] is REQUIRED, but MUST be negotiated using the signalling channel before use. For backwards compatibility, implementations are also REQUIRED to support the use of compound RTCP feedback packets if the remote endpoint does not agree to the use of non-compound RTCP in the signalling exchange.
To ease traversal of NAT and firewall devices, implementations are REQUIRED to implement and use Symmetric RTP [RFC4961]. The reasons for using symmetric RTP is primarily to avoid issues with NAT and Firewalls by ensuring that the flow is actually bi-directional and thus kept alive and registered as flow the intended recipient actually wants. In addition, it saves resources, specifically ports at the end-points, but also in the network as NAT mappings or firewall state is not unnecessary bloated. Also the amount of QoS state is reduced.
Implementations are REQUIRED to support signalled RTP synchronisation source (SSRC) identifiers, using the "a=ssrc:" SDP attribute defined in Section 4.1 and Section 5 of [RFC5576]. Implementations MUST also support the "previous-ssrc" source attribute defined in Section 6.2 of [RFC5576]. Other per-SSRC attributes defined in [RFC5576] MAY be supported.
Use of the "a=ssrc:" attribute to signal SSRC identifiers in an RTP session is OPTIONAL. Implementations MUST be prepared to accept RTP and RTCP packets using SSRCs that have not been explicitly signalled ahead of time. Implementations MUST support random SSRC assignment, and MUST support SSRC collision detection and resolution, according to [RFC3550]. When using signalled SSRC values, collision detection MUST be performed as described in Section 5 of [RFC5576].
It is often desirable to associate an RTP media stream with a non-RTP context (e.g., to associate an RTP media stream with an "m=" line in a session description formatted using SDP). If SSRCs are signalled this is straightforward (in SDP the "a=ssrc:" line will be at the media level, allowing a direct association with an "m=" line). If SSRCs are not signalled, the RTP payload type numbers used in an RTP media stream are often sufficient to associate that media stream with a signalling context (e.g., if RTP payload type numbers are assigned as described in Section 4.3 of this memo, the RTP payload types used by an RTP media stream can be compared with values in SDP "a=rtpmap:" lines, which are at the media level in SDP, and so map to an "m=" line).
The RTCP Canonical Name (CNAME) provides a persistent transport-level identifier for an RTP endpoint. While the Synchronisation Source (SSRC) identifier for an RTP endpoint can change if a collision is detected, or when the RTP application is restarted, its RTCP CNAME is meant to stay unchanged, so that RTP endpoints can be uniquely identified and associated with their RTP media streams within a set of related RTP sessions. For proper functionality, each RTP endpoint needs to have at least one unique RTCP CNAME value. An endpoint MAY have multiple CNAMEs, as the CNAME also identifies a particular synchronisation context, i.e. all SSRC associated with a CNAME share a common reference clock, and if an endpoint have SSRCs associated with different reference clocks it will need to use multiple CNAMEs. This ought not be common, and if possible reference clocks ought to be mapped to each other and one chosen to be used with RTP and RTCP.
The RTP specification [RFC3550] includes guidelines for choosing a unique RTP CNAME, but these are not sufficient in the presence of NAT devices. In addition, long-term persistent identifiers can be problematic from a privacy viewpoint. Accordingly, support for generating a short-term persistent RTCP CNAMEs following [RFC7022] is RECOMMENDED.
An WebRTC end-point MUST support reception of any CNAME that matches the syntax limitations specified by the RTP specification [RFC3550] and cannot assume that any CNAME will be chosen according to the form suggested above.
There are a number of RTP extensions that are either needed to obtain full functionality, or extremely useful to improve on the baseline performance, in the WebRTC application context. One set of these extensions is related to conferencing, while others are more generic in nature. The following subsections describe the various RTP extensions mandated or suggested for use within the WebRTC context.
RTP is inherently a group communication protocol. Groups can be implemented using a centralised server, multi-unicast, or using IP multicast. While IP multicast is popular in IPTV systems, overlay-based topologies dominate in interactive conferencing environments. Such overlay-based topologies typically use one or more central servers to connect end-points in a star or flat tree topology. These central servers can be implemented in a number of ways as discussed in the memo on RTP Topologies [I-D.ietf-avtcore-rtp-topologies-update].
Not all of the possible the overlay-based topologies are suitable for use in the WebRTC environment. Specifically:
Accordingly, only Point to Point (Topo-Point-to-Point), Multiple concurrent Point to Point (Mesh) and RTP Mixers (Topo-Mixer) topologies are needed to achieve the use-cases to be supported in WebRTC initially. These RECOMMENDED topologies are expected to be supported by all WebRTC end-points (these topologies require no special RTP-layer support in the end-point if the RTP features mandated in this memo are implemented).
The RTP extensions described in Section 5.1.1 to Section 5.1.6 are designed to be used with centralised conferencing, where an RTP middlebox (e.g., a conference bridge) receives a participant's RTP media streams and distributes them to the other participants. These extensions are not necessary for interoperability; an RTP endpoint that does not implement these extensions will work correctly, but might offer poor performance. Support for the listed extensions will greatly improve the quality of experience and, to provide a reasonable baseline quality, some these extensions are mandatory to be supported by WebRTC end-points.
The RTCP conferencing extensions are defined in Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF) [RFC4585] and the "Codec Control Messages in the RTP Audio-Visual Profile with Feedback (AVPF)" (CCM) [RFC5104] and are fully usable by the Secure variant of this profile (RTP/SAVPF) [RFC5124].
The Full Intra Request is defined in Sections 3.5.1 and 4.3.1 of the Codec Control Messages [RFC5104]. This message is used to make the mixer request a new Intra picture from a participant in the session. This is used when switching between sources to ensure that the receivers can decode the video or other predictive media encoding with long prediction chains. WebRTC senders MUST understand and react to the FIR feedback message since it greatly improves the user experience when using centralised mixer-based conferencing; support for sending the FIR message is OPTIONAL.
The Picture Loss Indication is defined in Section 6.3.1 of the RTP/AVPF profile [RFC4585]. It is used by a receiver to tell the sending encoder that it lost the decoder context and would like to have it repaired somehow. This is semantically different from the Full Intra Request above as there could be multiple ways to fulfil the request. WebRTC senders MUST understand and react to this feedback message as a loss tolerance mechanism; receivers MAY send PLI messages.
The Slice Loss Indicator is defined in Section 6.3.2 of the RTP/AVPF profile [RFC4585]. It is used by a receiver to tell the encoder that it has detected the loss or corruption of one or more consecutive macro blocks, and would like to have these repaired somehow. Support for this feedback message is OPTIONAL as a loss tolerance mechanism.
Reference Picture Selection Indication (RPSI) is defined in Section 6.3.3 of the RTP/AVPF profile [RFC4585]. Some video coding standards allow the use of older reference pictures than the most recent one for predictive coding. If such a codec is in used, and if the encoder has learned about a loss of encoder-decoder synchronisation, a known-as-correct reference picture can be used for future coding. The RPSI message allows this to be signalled. Support for RPSI messages is OPTIONAL.
The temporal-spatial trade-off request and notification are defined in Sections 3.5.2 and 4.3.2 of [RFC5104]. This request can be used to ask the video encoder to change the trade-off it makes between temporal and spatial resolution, for example to prefer high spatial image quality but low frame rate. Support for TSTR requests and notifications is OPTIONAL.
This feedback message is defined in Sections 3.5.4 and 4.2.1 of the Codec Control Messages [RFC5104]. This message and its notification message are used by a media receiver to inform the sending party that there is a current limitation on the amount of bandwidth available to this receiver. This can be various reasons for this: for example, an RTP mixer can use this message to limit the media rate of the sender being forwarded by the mixer (without doing media transcoding) to fit the bottlenecks existing towards the other session participants. WebRTC senders are REQUIRED to implement support for TMMBR messages, and MUST follow bandwidth limitations set by a TMMBR message received for their SSRC. The sending of TMMBR requests is OPTIONAL.
The RTP specification [RFC3550] provides the capability to include RTP header extensions containing in-band data, but the format and semantics of the extensions are poorly specified. The use of header extensions is OPTIONAL in the WebRTC context, but if they are used, they MUST be formatted and signalled following the general mechanism for RTP header extensions defined in [RFC5285], since this gives well-defined semantics to RTP header extensions.
As noted in [RFC5285], the requirement from the RTP specification that header extensions are "designed so that the header extension may be ignored" [RFC3550] stands. To be specific, header extensions MUST only be used for data that can safely be ignored by the recipient without affecting interoperability, and MUST NOT be used when the presence of the extension has changed the form or nature of the rest of the packet in a way that is not compatible with the way the stream is signalled (e.g., as defined by the payload type). Valid examples might include metadata that is additional to the usual RTP information.
Many RTP sessions require synchronisation between audio, video, and other content. This synchronisation is performed by receivers, using information contained in RTCP SR packets, as described in the RTP specification [RFC3550]. This basic mechanism can be slow, however, so it is RECOMMENDED that the rapid RTP synchronisation extensions described in [RFC6051] be implemented in addition to RTCP SR-based synchronisation. The rapid synchronisation extensions use the general RTP header extension mechanism [RFC5285], which requires signalling, but are otherwise backwards compatible.
The Client to Mixer Audio Level extension [RFC6464] is an RTP header extension used by a client to inform a mixer about the level of audio activity in the packet to which the header is attached. This enables a central node to make mixing or selection decisions without decoding or detailed inspection of the payload, reducing the complexity in some types of central RTP nodes. It can also save decoding resources in receivers, which can choose to decode only the most relevant RTP media streams based on audio activity levels.
The Client-to-Mixer Audio Level [RFC6464] extension is RECOMMENDED to be implemented. If it is implemented, it is REQUIRED that the header extensions are encrypted according to [RFC6904] since the information contained in these header extensions can be considered sensitive.
The Mixer to Client Audio Level header extension [RFC6465] provides the client with the audio level of the different sources mixed into a common mix by a RTP mixer. This enables a user interface to indicate the relative activity level of each session participant, rather than just being included or not based on the CSRC field. This is a pure optimisations of non critical functions, and is hence OPTIONAL to implement. If it is implemented, it is REQUIRED that the header extensions are encrypted according to [RFC6904] since the information contained in these header extensions can be considered sensitive.
(tbd: it seems likely that we need a mechanism to associate RTP media streams with signalling contexts. The mechanism by which this is done will likely be some combination of an RTP header extension, periodic transmission of a new RTCP SDES item, and some signalling extension. The semantics of those items are not yet settled; see draft-westerlund-avtext-rtcp-sdes-srcname, draft-ietf-mmusic-msid, and draft-even-mmusic-application-token for discussion).
There are tools that can make RTP media streams robust against packet loss and reduce the impact of loss on media quality. However, they all add extra bits compared to a non-robust stream. The overhead of these extra bits needs to be considered, and the aggregate bit-rate MUST be rate controlled to avoid causing network congestion (see Section 7). As a result, improving robustness might require a lower base encoding quality, but has the potential to deliver that quality with fewer errors. The mechanisms described in the following sub-sections can be used to improve tolerance to packet loss.
As a consequence of supporting the RTP/SAVPF profile, implementations can support negative acknowledgements (NACKs) for RTP data packets [RFC4585]. This feedback can be used to inform a sender of the loss of particular RTP packets, subject to the capacity limitations of the RTCP feedback channel. A sender can use this information to optimise the user experience by adapting the media encoding to compensate for known lost packets, for example.
Senders are REQUIRED to understand the Generic NACK message defined in Section 6.2.1 of [RFC4585], but MAY choose to ignore this feedback (following Section 4.2 of [RFC4585]). Receivers MAY send NACKs for missing RTP packets; [RFC4585] provides some guidelines on when to send NACKs. It is not expected that a receiver will send a NACK for every lost RTP packet, rather it needs to consider the cost of sending NACK feedback, and the importance of the lost packet, to make an informed decision on whether it is worth telling the sender about a packet loss event.
The RTP Retransmission Payload Format [RFC4588] offers the ability to retransmit lost packets based on NACK feedback. Retransmission needs to be used with care in interactive real-time applications to ensure that the retransmitted packet arrives in time to be useful, but can be effective in environments with relatively low network RTT (an RTP sender can estimate the RTT to the receivers using the information in RTCP SR and RR packets, as described at the end of Section 6.4.1 of [RFC3550]). The use of retransmissions can also increase the forward RTP bandwidth, and can potentially worsen the problem if the packet loss was caused by network congestion. We note, however, that retransmission of an important lost packet to repair decoder state can have lower cost than sending a full intra frame. It is not appropriate to blindly retransmit RTP packets in response to a NACK. The importance of lost packets and the likelihood of them arriving in time to be useful needs to be considered before RTP retransmission is used.
Receivers are REQUIRED to implement support for RTP retransmission packets [RFC4588]. Senders MAY send RTP retransmission packets in response to NACKs if the RTP retransmission payload format has been negotiated for the session, and if the sender believes it is useful to send a retransmission of the packet(s) referenced in the NACK. An RTP sender does not need to retransmit every NACKed packet.
The use of Forward Error Correction (FEC) can provide an effective protection against some degree of packet loss, at the cost of steady bandwidth overhead. There are several FEC schemes that are defined for use with RTP. Some of these schemes are specific to a particular RTP payload format, others operate across RTP packets and can be used with any payload format. It needs to be noted that using redundant encoding or FEC will lead to increased play out delay, which needs to be considered when choosing the redundancy or FEC formats and their respective parameters.
If an RTP payload format negotiated for use in a WebRTC session supports redundant transmission or FEC as a standard feature of that payload format, then that support MAY be used in the WebRTC session, subject to any appropriate signalling.
There are several block-based FEC schemes that are designed for use with RTP independent of the chosen RTP payload format. At the time of this writing there is no consensus on which, if any, of these FEC schemes is appropriate for use in the WebRTC context. Accordingly, this memo makes no recommendation on the choice of block-based FEC for WebRTC use.
WebRTC will be used in heterogeneous network environments using a variety set of link technologies, including both wired and wireless links, to interconnect potentially large groups of users around the world. As a result, the network paths between users can have widely varying one-way delays, available bit-rates, load levels, and traffic mixtures. Individual end-points can send one or more RTP media streams to each participant in a WebRTC conference, and there can be several participants. Each of these RTP media streams can contain different types of media, and the type of media, bit rate, and number of flows can be highly asymmetric. Non-RTP traffic can share the network paths with RTP flows. Since the network environment is not predictable or stable, WebRTC endpoints MUST ensure that the RTP traffic they generate can adapt to match changes in the available network capacity.
The quality of experience for users of WebRTC implementation is very dependent on effective adaptation of the media to the limitations of the network. End-points have to be designed so they do not transmit significantly more data than the network path can support, except for very short time periods, otherwise high levels of network packet loss or delay spikes will occur, causing media quality degradation. The limiting factor on the capacity of the network path might be the link bandwidth, or it might be competition with other traffic on the link (this can be non-WebRTC traffic, traffic due to other WebRTC flows, or even competition with other WebRTC flows in the same session).
An effective media congestion control algorithm is therefore an essential part of the WebRTC framework. However, at the time of this writing, there is no standard congestion control algorithm that can be used for interactive media applications such as WebRTC flows. Some requirements for congestion control algorithms for WebRTC sessions are discussed in [I-D.jesup-rtp-congestion-reqs], and it is expected that a future version of this memo will mandate the use of a congestion control algorithm that satisfies these requirements.
In the absence of a concrete congestion control algorithm, all WebRTC implementations MUST implement the RTP circuit breaker algorithm that is in described [I-D.ietf-avtcore-rtp-circuit-breakers]. The RTP circuit breaker is designed to enable applications to recognise and react to situations of extreme network congestion. However, since the RTP circuit breaker might not be triggered until congestion becomes extreme, it cannot be considered a substitute for congestion control, and applications MUST also implement congestion control to allow them to adapt to changes in network capacity. Any future RTP congestion control algorithms are expected to operate within the envelope allowed by the circuit breaker.
The session establishment signalling will also necessarily establish boundaries to which the media bit-rate will conform. The choice of media codecs provides upper- and lower-bounds on the supported bit-rates that the application can utilise to provide useful quality, and the packetization choices that exist. In addition, the signalling channel can establish maximum media bit-rate boundaries using the SDP "b=AS:" or "b=CT:" lines, and the RTP/AVPF Temporary Maximum Media Stream Bit Rate (TMMBR) Requests (see Section 5.1.6 of this memo). The combination of media codec choice and signalled bandwidth limits SHOULD be used to limit traffic based on known bandwidth limitations, for example the capacity of the edge links, to the extent possible.
Experience with the congestion control algorithms of TCP [RFC5681], TFRC [RFC5348], and DCCP [RFC4341], [RFC4342], [RFC4828], has shown that feedback on packet arrivals needs to be sent roughly once per round trip time. We note that the real-time media traffic might not have to adapt to changing path conditions as rapidly as needed for the elastic applications TCP was designed for, but frequent feedback is still needed to allow the congestion control algorithm to track the path dynamics.
The total RTCP bandwidth is limited in its transmission rate to a fraction of the RTP traffic (by default 5%). RTCP packets are larger than, e.g., TCP ACKs (even when non-compound RTCP packets are used). The RTP media stream bit rate thus limits the maximum feedback rate as a function of the mean RTCP packet size.
Interactive communication might not be able to afford waiting for packet losses to occur to indicate congestion, because an increase in play out delay due to queuing (most prominent in wireless networks) can easily lead to packets being dropped due to late arrival at the receiver. Therefore, more sophisticated cues might need to be reported -- to be defined in a suitable congestion control framework as noted above -- which, in turn, increase the report size again. For example, different RTCP XR report blocks (jointly) provide the necessary details to implement a variety of congestion control algorithms, but the (compound) report size grows quickly.
In group communication, the share of RTCP bandwidth needs to be shared by all group members, reducing the capacity and thus the reporting frequency per node.
Example: assuming 512 kbit/s video yields 3200 bytes/s RTCP bandwidth, split across two entities in a point-to-point session. An endpoint could thus send a report of 100 bytes about every 70ms or for every other frame in a 30 fps video.
There are legacy implementations that do not implement RTCP, and hence do not provide any congestion feedback. Congestion control cannot be performed with these end-points. WebRTC implementations that need to interwork with such end-points MUST limit their transmission to a low rate, equivalent to a VoIP call using a low bandwidth codec, that is unlikely to cause any significant congestion.
When interworking with legacy implementations that support RTCP using the RTP/AVP profile [RFC3551], congestion feedback is provided in RTCP RR packets every few seconds. Implementations that have to interwork with such end-points MUST ensure that they keep within the RTP circuit breaker [I-D.ietf-avtcore-rtp-circuit-breakers] constraints to limit the congestion they can cause.
If a legacy end-point supports RTP/AVPF, this enables negotiation of important parameters for frequent reporting, such as the "trr-int" parameter, and the possibility that the end-point supports some useful feedback format for congestion control purpose such as TMMBR [RFC5104]. Implementations that have to interwork with such end-points MUST ensure that they stay within the RTP circuit breaker [I-D.ietf-avtcore-rtp-circuit-breakers] constraints to limit the congestion they can cause, but might find that they can achieve better congestion response depending on the amount of feedback that is available.
With proprietary congestion control algorithms issues can arise when different algorithms and implementations interact in a communication session. If the different implementations have made different choices in regards to the type of adaptation, for example one sender based, and one receiver based, then one could end up in situation where one direction is dual controlled, when the other direction is not controlled. This memo cannot mandate behaviour for proprietary congestion control algorithms, but implementations that use such algorithms ought to be aware of this issue, and try to ensure that both effective congestion control is negotiated for media flowing in both directions. If the IETF were to standardise both sender- and receiver-based congestion control algorithms for WebRTC traffic in the future, the issues of interoperability, control, and ensuring that both directions of media flow are congestion controlled would also need to be considered.
As described in Section 4.1, implementations are REQUIRED to generate RTCP Sender Report (SR) and Reception Report (RR) packets relating to the RTP media streams they send and receive. These RTCP reports can be used for performance monitoring purposes, since they include basic packet loss and jitter statistics.
A large number of additional performance metrics are supported by the RTCP Extended Reports (XR) framework [RFC3611][RFC6792]. It is not yet clear what extended metrics are appropriate for use in the WebRTC context, so there is no requirement that implementations generate RTCP XR packets. However, implementations that can use detailed performance monitoring data MAY generate RTCP XR packets as appropriate; the use of such packets SHOULD be signalled in advance.
All WebRTC implementations MUST be prepared to receive RTP XR report packets, whether or not they were signalled. There is no requirement that the data contained in such reports be used, or exposed to the Javascript application, however.
It is possible that the core set of RTP protocols and RTP extensions specified in this memo will prove insufficient for the future needs of WebRTC applications. In this case, future updates to this memo MUST be made following the Guidelines for Writers of RTP Payload Format Specifications [RFC2736] and Guidelines for Extending the RTP Control Protocol [RFC5968], and SHOULD take into account any future guidelines for extending RTP and related protocols that have been developed.
Authors of future extensions are urged to consider the wide range of environments in which RTP is used when recommending extensions, since extensions that are applicable in some scenarios can be problematic in others. Where possible, the WebRTC framework will adopt RTP extensions that are of general utility, to enable easy implementation of a gateway to other applications using RTP, rather than adopt mechanisms that are narrowly targeted at specific WebRTC use cases.
RTP is built with the assumption that an external signalling channel exists, and can be used to configure RTP sessions and their features. The basic configuration of an RTP session consists of the following parameters:
These parameters are often expressed in SDP messages conveyed within an offer/answer exchange. RTP does not depend on SDP or on the offer/answer model, but does require all the necessary parameters to be agreed upon, and provided to the RTP implementation. We note that in the WebRTC context it will depend on the signalling model and API how these parameters need to be configured but they will be need to either set in the API or explicitly signalled between the peers.
The WebRTC API [W3C.WD-webrtc-20130910] and the Media Capture and Streams API [W3C.WD-mediacapture-streams-20130903] defines and uses the concept of a MediaStream that consists of zero or more MediaStreamTracks. A MediaStreamTrack is an individual stream of media from any type of media source like a microphone or a camera, but also conceptual sources, like a audio mix or a video composition, are possible. The MediaStreamTracks within a MediaStream need to be possible to play out synchronised.
A MediaStreamTrack's realisation in RTP in the context of an RTCPeerConnection consists of a source packet stream identified with an SSRC within an RTP session part of the RTCPeerConnection. The MediaStreamTrack can also result in additional packet streams, and thus SSRCs, in the same RTP session. These can be dependent packet streams from scalable encoding of the source stream associated with the MediaStreamTrack, if such a media encoder is used. They can also be redundancy packet streams, these are created when applying Forward Error Correction [sec-FEC] or RTP retransmission [sec-rtx] to the source packet stream.
It is important to note that the same media source can be feeding multiple MediaStreamTracks. As different sets of constraints or other parameters can be applied to the MediaStreamTrack, each MediaStreamTrack instance added to a RTCPeerConnection SHALL result in an independent source packet stream, with its own set of associated packet streams, and thus different SSRC(s). It will depend on applied constraints and parameters if the source stream and the encoding configuration will be identical between different MediaStreamTracks sharing the same media source. Thus it is possible for multiple source packet streams to share encoded streams (but not packet streams), but this is an implementation choice to try to utilise such optimisations. Note that such optimizations would need to take into account that the constraints for one of the MediaStreamTracks can at any moment change, meaning that the encoding configurations should no longer be identical.
The same MediaStreamTrack can also be included in multiple MediaStreams, thus multiple sets of MediaStreams can implicitly need to use the same synchronisation base. To ensure that this works in all cases, and don't forces a endpoint to change synchronisation base and CNAME in the middle of a ongoing delivery of any packet streams, which would cause media disruption; all MediaStreamTracks and their associated SSRCs originating from the same endpoint MUST be sent using the same CNAME within one RTCPeerConnection as well as across all RTCPeerConnections part of the same communication session context, which for a browser are a single origin.
The above will currently force a WebRTC endpoint that receives an MediaStreamTrack on one RTCPeerConnection and adds it as an outgoing on any RTCPeerConnection to perform resynchronisation of the stream. This, as the sending party needs to change the CNAME, which implies that it has to use a locally available system clock as timebase for the synchronisation. Thus, the relative relation between the timebase of the incoming stream and the system sending out needs to defined. This relation also needs monitoring for clock drift and likely adjustments of the synchronisation. The sending entity is also responsible for congestion control for its the sent streams. In cases of packet loss the loss of incoming data also needs to be handled. This leads to the observation that the method that is least likely to cause issues or interruptions in the outgoing source packet stream is a model of full decoding, including repair etc followed by encoding of the media again into the outgoing packet stream. Optimisations of this method is clearly possible and implementation specific.
A WebRTC endpoint MUST support receiving multiple MediaStreamTracks, where each of different MediaStreamTracks (and their sets of associated packet streams) uses different CNAMEs. However, MediaStreamTracks that are received with different CNAMEs have no defined synchronisation.
The binding between the WebRTC MediaStreams, MediaStreamTracks and the SSRC is done as specified in "Cross Session Stream Identification in the Session Description Protocol" [I-D.ietf-mmusic-msid]. This document [I-D.ietf-mmusic-msid] also defines, in section 4.1, how to map unknown source packet stream SSRCs to MediaStreamTracks and MediaStreams. Commonly the RTP Payload Type of any incoming packets will reveal if the packet stream is a source stream or a redundancy or dependent packet stream. The association to the correct source packet stream depends on the payload format in use for the packet stream.
The following discussion provides some guidance on the implementation of the RTP features described in this memo. The focus is on a WebRTC end-point implementation perspective, and while some mention is made of the behaviour of middleboxes, that is not the focus of this memo.
A WebRTC end-point will be a simultaneous participant in one or more RTP sessions. Each RTP session can convey multiple media flows, and can include media data from multiple end-points. In the following, we outline some ways in which WebRTC end-points can configure and use RTP sessions.
RTP is a group communication protocol, and in a WebRTC context every RTP session can potentially contain multiple media flows. There are several reasons why this might be desirable:
In addition to sending and receiving multiple media streams within a single RTP session, a WebRTC end-point might participate in multiple RTP sessions. There are several reasons why a WebRTC end-point might choose to do this:
+---+ +---+
| A |<--->| B |
+---+ +---+
^ ^
\ /
\ /
v v
+---+
| C |
+---+
Figure 1: Multi-unicast using several RTP sessions
+---+ +-------------+ +---+
| A |<---->| |<---->| B |
+---+ | RTP mixer, | +---+
| translator, |
| or other |
+---+ | middlebox | +---+
| C |<---->| |<---->| D |
+---+ +-------------+ +---+
Figure 2: RTP mixer with only unicast paths
There are use cases for differentiated treatment of RTP media streams. Such differentiation can happen at several places in the system. First of all is the prioritization within the end-point sending the media, which controls, both which RTP media streams that will be sent, and their allocation of bit-rate out of the current available aggregate as determined by the congestion control.
It is expected that the WebRTC API will allow the application to indicate relative priorities for different MediaStreamTracks. These priorities can then be used to influence the local RTP processing, especially when it comes to congestion control response in how to divide the available bandwidth between the RTP flows. Any changes in relative priority will also need to be considered for RTP flows that are associated with the main RTP flows, such as RTP retransmission streams and FEC. The importance of such associated RTP traffic flows is dependent on the media type and codec used, in regards to how robust that codec is to packet loss. However, a default policy might to be to use the same priority for associated RTP flows as for the primary RTP flow.
Secondly, the network can prioritize packet flows, including RTP media streams. Typically, differential treatment includes two steps, the first being identifying whether an IP packet belongs to a class that has to be treated differently, the second the actual mechanism to prioritize packets. This is done according to three methods:
Flow-based differentiation will provide the same treatment to all packets within a flow, i.e., relative prioritization is not possible. Moreover, if the resources are limited it might not be possible to provide differential treatment compared to best-effort for all the flows in a WebRTC application. When flow-based differentiation is available the WebRTC application needs to know about it so that it can provide the separation of the RTP media streams onto different UDP flows to enable a more granular usage of flow based differentiation. That way at least providing different prioritization of audio and video if desired by application.
DiffServ assumes that either the end-point or a classifier can mark the packets with an appropriate DSCP so that the packets are treated according to that marking. If the end-point is to mark the traffic two requirements arise in the WebRTC context: 1) The WebRTC application or browser has to know which DSCP to use and that it can use them on some set of RTP media streams. 2) The information needs to be propagated to the operating system when transmitting the packet. Details of this process are outside the scope of this memo and are further discussed in "DSCP and other packet markings for RTCWeb QoS" [I-D.dhesikan-tsvwg-rtcweb-qos].
For packet based marking schemes it might be possible to mark individual RTP packets differently based on the relative priority of the RTP payload. For example video codecs that have I, P, and B pictures could prioritise any payloads carrying only B frames less, as these are less damaging to loose. As default policy all RTP packets related to a media stream ought to be provided with the same prioritization; per-packet prioritization is outside the scope of this memo, but might be specified elsewhere in future.
It is also important to consider how RTCP packets associated with a particular RTP media flow need to be marked. RTCP compound packets with Sender Reports (SR), ought to be marked with the same priority as the RTP media flow itself, so the RTCP-based round-trip time (RTT) measurements are done using the same flow priority as the media flow experiences. RTCP compound packets containing RR packet ought to be sent with the priority used by the majority of the RTP media flows reported on. RTCP packets containing time-critical feedback packets can use higher priority to improve the timeliness and likelihood of delivery of such feedback.
Each RTP media stream is identified by a unique synchronisation source (SSRC) identifier. The SSRC identifier is carried in the RTP data packets comprising a media stream, and is also used to identify that stream in the corresponding RTCP reports. The SSRC is chosen as discussed in Section 4.8. The first stage in demultiplexing RTP and RTCP packets received at a WebRTC end-point is to separate the media streams based on their SSRC value; once that is done, additional demultiplexing steps can determine how and where to render the media.
RTP allows a mixer, or other RTP-layer middlebox, to combine media flows from multiple sources to form a new media flow. The RTP data packets in that new flow can include a Contributing Source (CSRC) list, indicating which original SSRCs contributed to the combined packet. As described in Section 4.1, implementations need to support reception of RTP data packets containing a CSRC list and RTCP packets that relate to sources present in the CSRC list. The CSRC list can change on a packet-by-packet basis, depending on the mixing operation being performed. Knowledge of what sources contributed to a particular RTP packet can be important if the user interface indicates which participants are active in the session. Changes in the CSRC list included in packets needs to be exposed to the WebRTC application using some API, if the application is to be able to track changes in session participation. It is desirable to map CSRC values back into WebRTC MediaStream identities as they cross this API, to avoid exposing the SSRC/CSRC name space to JavaScript applications.
If the mixer-to-client audio level extension [RFC6465] is being used in the session (see Section 5.2.3), the information in the CSRC list is augmented by audio level information for each contributing source. This information can usefully be exposed in the user interface.
The RTP standard [RFC3550] requires any RTP implementation to have support for detecting and handling SSRC collisions, i.e., resolve the conflict when two different end-points use the same SSRC value. This requirement also applies to WebRTC end-points. There are several scenarios where SSRC collisions can occur.
In a point-to-point session where each SSRC is associated with either of the two end-points and where the main media carrying SSRC identifier will be announced in the signalling channel, a collision is less likely to occur due to the information about used SSRCs provided by Source-Specific SDP Attributes [RFC5576]. Still if both end-points start uses an new SSRC identifier prior to having signalled it to the peer and received acknowledgement on the signalling message, there can be collisions. The Source-Specific SDP Attributes [RFC5576] contains no mechanism to resolve SSRC collisions or reject a end-points usage of an SSRC.
There could also appear SSRC values that are not signalled. This is more likely than it appears as certain RTP functions need extra SSRCs to provide functionality related to another (the "main") SSRC, for example, SSRC multiplexed RTP retransmission [RFC4588]. In those cases, an end-point can create a new SSRC that strictly doesn't need to be announced over the signalling channel to function correctly on both RTP and RTCPeerConnection level.
The more likely case for SSRC collision is that multiple end-points in a multiparty conference create new sources and signals those towards the central server. In cases where the SSRC/CSRC are propagated between the different end-points from the central node collisions can occur.
Another scenario is when the central node manages to connect an end-point's RTCPeerConnection to another RTCPeerConnection the end-point already has, thus forming a loop where the end-point will receive its own traffic. While is is clearly considered a bug, it is important that the end-point is able to recognise and handle the case when it occurs. This case becomes even more problematic when media mixers, and so on, are involved, where the stream received is a different stream but still contains this client's input.
These SSRC/CSRC collisions can only be handled on RTP level as long as the same RTP session is extended across multiple RTCPeerConnections by a RTP middlebox. To resolve the more generic case where multiple RTCPeerConnections are interconnected, then identification of the media source(s) part of a MediaStreamTrack being propagated across multiple interconnected RTCPeerConnection needs to be preserved across these interconnections.
When an end-point sends media from more than one media source, it needs to consider if (and which of) these media sources are to be synchronized. In RTP/RTCP, synchronisation is provided by having a set of RTP media streams be indicated as coming from the same synchronisation context and logical end-point by using the same RTCP CNAME identifier.
The next provision is that the internal clocks of all media sources, i.e., what drives the RTP timestamp, can be correlated to a system clock that is provided in RTCP Sender Reports encoded in an NTP format. By correlating all RTP timestamps to a common system clock for all sources, the timing relation of the different RTP media streams, also across multiple RTP sessions can be derived at the receiver and, if desired, the streams can be synchronized. The requirement is for the media sender to provide the correlation information; it is up to the receiver to use it or not.
The overall security architecture for WebRTC is described in [I-D.ietf-rtcweb-security-arch], and security considerations for the WebRTC framework are described in [I-D.ietf-rtcweb-security]. These considerations apply to this memo also.
The security considerations of the RTP specification, the RTP/SAVPF profile, and the various RTP/RTCP extensions and RTP payload formats that form the complete protocol suite described in this memo apply. We do not believe there are any new security considerations resulting from the combination of these various protocol extensions.
The Extended Secure RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback [RFC5124] (RTP/SAVPF) provides handling of fundamental issues by offering confidentiality, integrity and partial source authentication. A mandatory to implement media security solution is created by combing this secured RTP profile and DTLS-SRTP keying [RFC5764] as defined by Section 5.5 of [I-D.ietf-rtcweb-security-arch].
RTCP packets convey a Canonical Name (CNAME) identifier that is used to associate media flows that need to be synchronised across related RTP sessions. Inappropriate choice of CNAME values can be a privacy concern, since long-term persistent CNAME identifiers can be used to track users across multiple WebRTC calls. Section 4.9 of this memo provides guidelines for generation of untraceable CNAME values that alleviate this risk.
The guidelines in [RFC6562] apply when using variable bit rate (VBR) audio codecs such as Opus (see Section 4.3 for discussion of mandated audio codecs). These guidelines in [RFC6562] also apply, but are of lesser importance, when using the client-to-mixer audio level header extensions (Section 5.2.2) or the mixer-to-client audio level header extensions (Section 5.2.3).
This memo makes no request of IANA.
Note to RFC Editor: this section is to be removed on publication as an RFC.
This section contains a summary of the open issues or to be done things noted in the document:
The authors would like to thank Bernard Aboba, Harald Alvestrand, Cary Bran, Charles Eckel, Cullen Jennings, Dan Romascanu, and the other members of the IETF RTCWEB working group for their valuable feedback.