--- 1/draft-ietf-rtcweb-rtp-usage-03.txt 2012-07-16 19:15:06.409341692 +0200 +++ 2/draft-ietf-rtcweb-rtp-usage-04.txt 2012-07-16 19:15:06.541344345 +0200 @@ -1,21 +1,21 @@ Network Working Group C. Perkins Internet-Draft University of Glasgow Intended status: Standards Track M. Westerlund -Expires: December 6, 2012 Ericsson +Expires: January 17, 2013 Ericsson J. Ott Aalto University - June 4, 2012 + July 16, 2012 Web Real-Time Communication (WebRTC): Media Transport and Use of RTP - draft-ietf-rtcweb-rtp-usage-03 + draft-ietf-rtcweb-rtp-usage-04 Abstract 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. @@ -28,21 +28,21 @@ 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 December 6, 2012. + This Internet-Draft will expire on January 17, 2013. Copyright Notice Copyright (c) 2012 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 @@ -52,147 +52,151 @@ the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License. Table of Contents 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4 2. Rationale . . . . . . . . . . . . . . . . . . . . . . . . . . 4 3. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 5 4. WebRTC Use of RTP: Core Protocols . . . . . . . . . . . . . . 6 4.1. RTP and RTCP . . . . . . . . . . . . . . . . . . . . . . . 6 - 4.2. Choice of RTP Profile . . . . . . . . . . . . . . . . . . 7 - 4.3. Choice of RTP Payload Formats . . . . . . . . . . . . . . 7 - 4.4. RTP Session Multiplexing . . . . . . . . . . . . . . . . . 8 - 4.5. RTP and RTCP Multiplexing . . . . . . . . . . . . . . . . 8 - 4.6. Reduced Size RTCP . . . . . . . . . . . . . . . . . . . . 9 - 4.7. Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . . . 9 - 4.8. Generation of the RTCP Canonical Name (CNAME) . . . . . . 10 - 5. WebRTC Use of RTP: Extensions . . . . . . . . . . . . . . . . 10 - 5.1. Conferencing Extensions . . . . . . . . . . . . . . . . . 10 - 5.1.1. Full Intra Request . . . . . . . . . . . . . . . . . . 11 - 5.1.2. Picture Loss Indication . . . . . . . . . . . . . . . 11 - 5.1.3. Slice Loss Indication . . . . . . . . . . . . . . . . 11 - 5.1.4. Reference Picture Selection Indication . . . . . . . . 12 - 5.1.5. Temporary Maximum Media Stream Bit Rate Request . . . 12 - 5.2. Header Extensions . . . . . . . . . . . . . . . . . . . . 12 - 5.2.1. Rapid Synchronisation . . . . . . . . . . . . . . . . 12 - 5.2.2. Client to Mixer Audio Level . . . . . . . . . . . . . 13 - 5.2.3. Mixer to Client Audio Level . . . . . . . . . . . . . 13 - 6. WebRTC Use of RTP: Improving Transport Robustness . . . . . . 13 - 6.1. Retransmission . . . . . . . . . . . . . . . . . . . . . . 14 - 6.2. Forward Error Correction (FEC) . . . . . . . . . . . . . . 15 - 6.2.1. Basic Redundancy . . . . . . . . . . . . . . . . . . . 15 - 6.2.2. Block Based FEC . . . . . . . . . . . . . . . . . . . 16 - 6.2.3. Recommendations for FEC . . . . . . . . . . . . . . . 17 + 4.2. Choice of the RTP Profile . . . . . . . . . . . . . . . . 7 + 4.3. Choice of RTP Payload Formats . . . . . . . . . . . . . . 8 + 4.4. RTP Session Multiplexing . . . . . . . . . . . . . . . . . 9 + 4.5. RTP and RTCP Multiplexing . . . . . . . . . . . . . . . . 10 + 4.6. Reduced Size RTCP . . . . . . . . . . . . . . . . . . . . 10 + 4.7. Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . . . 11 + 4.8. Choice of RTP Synchronisation Source (SSRC) . . . . . . . 11 + 4.9. Generation of the RTCP Canonical Name (CNAME) . . . . . . 11 + 5. WebRTC Use of RTP: Extensions . . . . . . . . . . . . . . . . 12 + 5.1. Conferencing Extensions . . . . . . . . . . . . . . . . . 12 + 5.1.1. Full Intra Request (FIR) . . . . . . . . . . . . . . . 13 + 5.1.2. Picture Loss Indication (PLI) . . . . . . . . . . . . 13 + 5.1.3. Slice Loss Indication (SLI) . . . . . . . . . . . . . 13 + 5.1.4. Reference Picture Selection Indication (RPSI) . . . . 14 + 5.1.5. Temporal-Spatial Trade-off Request (TSTR) . . . . . . 14 + 5.1.6. Temporary Maximum Media Stream Bit Rate Request . . . 14 + 5.2. Header Extensions . . . . . . . . . . . . . . . . . . . . 14 + 5.2.1. Rapid Synchronisation . . . . . . . . . . . . . . . . 15 + 5.2.2. Client-to-Mixer Audio Level . . . . . . . . . . . . . 15 + 5.2.3. Mixer-to-Client Audio Level . . . . . . . . . . . . . 15 + 6. WebRTC Use of RTP: Improving Transport Robustness . . . . . . 16 + 6.1. Negative Acknowledgements and RTP Retransmission . . . . . 16 + 6.2. Forward Error Correction (FEC) . . . . . . . . . . . . . . 17 7. WebRTC Use of RTP: Rate Control and Media Adaptation . . . . . 17 - 7.1. Congestion Control Requirements . . . . . . . . . . . . . 19 + 7.1. Congestion Control Requirements . . . . . . . . . . . . . 18 7.2. Rate Control Boundary Conditions . . . . . . . . . . . . . 19 - 7.3. RTCP Limiations . . . . . . . . . . . . . . . . . . . . . 19 - 7.4. Legacy Interop Limitations . . . . . . . . . . . . . . . . 20 - 8. WebRTC Use of RTP: Performance Monitoring . . . . . . . . . . 21 + 7.3. RTCP Limitations for Congestion Control . . . . . . . . . 19 + 7.4. Congestion Control Interoperability With Legacy Systems . 20 + 8. WebRTC Use of RTP: Performance Monitoring . . . . . . . . . . 20 9. WebRTC Use of RTP: Future Extensions . . . . . . . . . . . . . 21 10. Signalling Considerations . . . . . . . . . . . . . . . . . . 21 - 11. WebRTC API Considerations . . . . . . . . . . . . . . . . . . 23 - 11.1. API MediaStream to RTP Mapping . . . . . . . . . . . . . . 23 - + 11. WebRTC API Considerations . . . . . . . . . . . . . . . . . . 22 + 11.1. API MediaStream to RTP Mapping . . . . . . . . . . . . . . 22 12. RTP Implementation Considerations . . . . . . . . . . . . . . 23 - 12.1. RTP Sessions and PeerConnection . . . . . . . . . . . . . 24 + 12.1. RTP Sessions and PeerConnection . . . . . . . . . . . . . 23 12.2. Multiple Sources . . . . . . . . . . . . . . . . . . . . . 25 12.3. Multiparty . . . . . . . . . . . . . . . . . . . . . . . . 25 - 12.4. SSRC Collision Detection . . . . . . . . . . . . . . . . . 27 - 12.5. Contributing Sources . . . . . . . . . . . . . . . . . . . 28 - 12.6. Media Synchronization . . . . . . . . . . . . . . . . . . 29 - 12.7. Multiple RTP End-points . . . . . . . . . . . . . . . . . 29 - 12.8. Simulcast . . . . . . . . . . . . . . . . . . . . . . . . 30 - 12.9. Differentiated Treatment of Flows . . . . . . . . . . . . 30 + 12.4. SSRC Collision Detection . . . . . . . . . . . . . . . . . 26 + 12.5. Contributing Sources . . . . . . . . . . . . . . . . . . . 27 + 12.6. Media Synchronization . . . . . . . . . . . . . . . . . . 28 + 12.7. Multiple RTP End-points . . . . . . . . . . . . . . . . . 28 + 12.8. Simulcast . . . . . . . . . . . . . . . . . . . . . . . . 29 + 12.9. Differentiated Treatment of Flows . . . . . . . . . . . . 29 13. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 31 - 14. Security Considerations . . . . . . . . . . . . . . . . . . . 32 - 15. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 32 + 14. Security Considerations . . . . . . . . . . . . . . . . . . . 31 + 15. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 31 16. References . . . . . . . . . . . . . . . . . . . . . . . . . . 32 16.1. Normative References . . . . . . . . . . . . . . . . . . . 32 - 16.2. Informative References . . . . . . . . . . . . . . . . . . 35 - Appendix A. Supported RTP Topologies . . . . . . . . . . . . . . 37 - A.1. Point to Point . . . . . . . . . . . . . . . . . . . . . . 37 - A.2. Multi-Unicast (Mesh) . . . . . . . . . . . . . . . . . . . 40 - A.3. Mixer Based . . . . . . . . . . . . . . . . . . . . . . . 43 - A.3.1. Media Mixing . . . . . . . . . . . . . . . . . . . . . 43 - A.3.2. Media Switching . . . . . . . . . . . . . . . . . . . 46 - A.3.3. Media Projecting . . . . . . . . . . . . . . . . . . . 49 - A.4. Translator Based . . . . . . . . . . . . . . . . . . . . . 52 - A.4.1. Transcoder . . . . . . . . . . . . . . . . . . . . . . 52 - A.4.2. Gateway / Protocol Translator . . . . . . . . . . . . 53 - A.4.3. Relay . . . . . . . . . . . . . . . . . . . . . . . . 55 - A.5. End-point Forwarding . . . . . . . . . . . . . . . . . . . 59 - A.6. Simulcast . . . . . . . . . . . . . . . . . . . . . . . . 60 - Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 61 + 16.2. Informative References . . . . . . . . . . . . . . . . . . 34 + Appendix A. Supported RTP Topologies . . . . . . . . . . . . . . 36 + A.1. Point to Point . . . . . . . . . . . . . . . . . . . . . . 36 + A.2. Multi-Unicast (Mesh) . . . . . . . . . . . . . . . . . . . 39 + A.3. Mixer Based . . . . . . . . . . . . . . . . . . . . . . . 42 + A.3.1. Media Mixing . . . . . . . . . . . . . . . . . . . . . 42 + A.3.2. Media Switching . . . . . . . . . . . . . . . . . . . 45 + A.3.3. Media Projecting . . . . . . . . . . . . . . . . . . . 48 + A.4. Translator Based . . . . . . . . . . . . . . . . . . . . . 51 + A.4.1. Transcoder . . . . . . . . . . . . . . . . . . . . . . 51 + A.4.2. Gateway / Protocol Translator . . . . . . . . . . . . 52 + A.4.3. Relay . . . . . . . . . . . . . . . . . . . . . . . . 54 + A.5. End-point Forwarding . . . . . . . . . . . . . . . . . . . 58 + A.6. Simulcast . . . . . . . . . . . . . . . . . . . . . . . . 59 + Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 60 1. Introduction 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 is a new protocol - framework that provides support for direct, interactive, real-time + 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 must be implemented by all WebRTC-aware browsers, - along with suggested extensions for enhanced functionality. + RTP features that are to be implemented by all WebRTC-aware end- + points, along with suggested extensions for enhanced functionality. The WebRTC overview [I-D.ietf-rtcweb-overview] outlines the complete WebRTC framework, of which this memo is a part. 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 requirement terminology. Requirements for core RTP protocols are described in Section 4 and recommended RTP extensions are described in Section 5. Section 6 outlines mechanisms that can increase robustness to network problems, while Section 7 describes the required congestion control and rate adaptation mechanisms. The - discussion of required RTP mechanisms concludes in Section 8 with a + 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. This memo concludes with an appendix discussing several different RTP Topologies, and how they affect the RTP session(s) and various implementation details of possible realization of central nodes. 2. Rationale 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 - features should 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 15 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. + 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 15 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. + + RTP and RTCP extensions not discussed in this document can still be + implemented by a WebRTC end-point, but they are considered optional, + are not required for interoperability, and do not provide features + needed to address the WebRTC use cases and requirements + [I-D.ietf-rtcweb-use-cases-and-requirements]. While the baseline set of RTP features and extensions defined in this - memo is targetted at the requirements of the WebRTC framework, it is + 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 apppropriate for other desktop or mobile video + extensions will be appropriate for other desktop or mobile video conferencing systems, or for room-based high-quality telepresence applications. 3. Terminology This memo specifies various requirements levels for implementation or use of RTP features and extensions. When we describe the importance of RTP extensions, or the need for implementation support, we use the following requirement levels to specify the importance of the feature in the WebRTC framework: @@ -211,330 +215,432 @@ that it enhances the product while another vendor may omit the same item. An implementation which does not include a particular option MUST be prepared to interoperate with another implementation which does include the option, though perhaps with reduced functionality. In the same vein an implementation which does include a particular option MUST be prepared to interoperate with another implementation which does not include the option (except, of course, for the feature the option provides.) These key words are used in a manner consistent with their definition - in [RFC2119]. + in [RFC2119]. The above 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: + + RTP Media Stream: A sequence of RTP packets, and associated RTCP + packets, using a single synchronisation source (SSRC) that + together carries part or all of the content of a specific Media + Type from a specific sender source within a given RTP session. + + RTP Session: As defined by [RFC3550], the endpoints belonging to the + same RTP Session are those that share a single SSRC space. That + is, those endpoints can see an SSRC identifier transmitted by any + one of the other endpoints. An endpoint can see an SSRC either + directly in RTP and RTCP packets, or as a contributing source + (CSRC) in RTP packets from a mixer. The RTP Session scope is + hence decided by the endpoints' network interconnection topology, + in combination with RTP and RTCP forwarding strategies deployed by + endpoints and any interconnecting middle nodes. + + WebRTC MediaStream: The MediaStream concept defined by the W3C in + the API. + + Other terms are used according to their definitions from the RTP + Specification [RFC3550] and WebRTC overview + [I-D.ietf-rtcweb-overview] documents. 4. WebRTC Use of RTP: Core Protocols The following sections describe the core features of RTP and RTCP - that MUST be implemented, along with the mandated RTP profiles and + 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 MUST implement to - function effectively on today's networks. + essential features that all WebRTC implementations need to implement + to function effectively on today's networks. 4.1. RTP and RTCP 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: o Support for use of multiple simultaneous SSRC values in a single RTP session, including support for RTP end-points that send many SSRC values simultaneously. o Random choice of SSRC on joining a session; collision detection - and resolution for SSRC values. + and resolution for SSRC values (but see also Section 4.8). - o Support reception of RTP data packets containing CSRC lists, as - generated by RTP mixers. + o Support for reception of RTP data packets containing CSRC lists, + as generated by RTP mixers, and RTCP packets relating to CSRCs. o Support for sending correct synchronization information in the - RTCP Sender Reports, with RECOMMENDED support for the rapid RTP - synchronisation extensions (see Section 5.2.1). + RTCP Sender Reports, to allow a receiver to implement lip-sync, + with RECOMMENDED support for the rapid RTP synchronisation + extensions (see Section 5.2.1). - o Support for standard RTCP packet types, include SR, RR, SDES, and - BYE packets. + o Support for sending and receiving RTCP SR, RR, SDES, and BYE + packet types, with OPTIONAL support for other RTCP packet types; + implementations MUST ignore unknown RTCP packet types. o Support for multiple end-points in a single RTP session, and for scaling the RTCP transmission interval according to the number of - participants in the session; support randomised RTCP transmission - intervals to avoid synchronisation of RTCP reports. + participants in the session; support for randomised RTCP + transmission intervals to avoid synchronisation of RTCP reports; + support for RTCP timer reconsideration. + + o Support for configuring the RTCP bandwidth as a fraction of the + media bandwidth, and for configuring the fraction of the RTCP + bandwidth allocated to senders, e.g., using the SDP "b=" line. It is known that a significant number of legacy RTP implementations, - especially those targetted for purely VoIP systems, do not support - all of the above features. + 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. -4.2. Choice of RTP Profile +4.2. Choice of the RTP Profile 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 Real-time Transport Control Protocol (RTCP)- Based Feedback (RTP/SAVPF)" [RFC5124] is REQUIRED to 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 RTP/AVPF part of RTP/SAVPF is required to get the improved RTCP + The RTCP-based feedback extensions 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. The RTP/ - AVPF profile also saves RTCP bandwidth, and will commonly only use - the full RTCP bandwidth allocation when there are many events that - require feedback. The RTP/AVPF functionality is also needed to make - use of the RTP conferencing extensions discussed in Section 5.1. + 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. Note: The enhanced RTCP timer model defined in the RTP/AVPF profile is backwards compatible with legacy systems that implement only the base RTP/AVP profile, given some constraints on parameter configuration such as the RTCP bandwidth value and "trr-int" (the most important factor for interworking with RTP/AVP end-points via a gateway is to set the trr-int parameter to a value representing 4 seconds). - The RTP/SAVP part of the RTP/SAVPF profile is for support for Secure - RTP (SRTP) [RFC3711]. This provides media encryption, integrity - protection, replay protection and a limited form of source + The secure RTP profile is needed to provide SRTP media encryption, + integrity protection, replay protection and a limited form of source authentication. - WebRTC implementation MUST NOT send packets using the RTP/AVP profile - or the RTP/AVPF profile; they MUST use the RTP/SAVPF profile. WebRTC - implementations MUST support DTLS-SRTP [RFC5764] for key-management. + 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. - (tbd: There is ongoing discussion on what additional keying mechanism - is to be required, what are the mandated cryptographic transforms. - This section needs to be updated based on the results of that - discussion.) + Implementations MUST support DTLS-SRTP [RFC5764] for key-management. + Other key management schemes MAY be supported. 4.3. Choice of RTP Payload Formats - (tbd: say something about the choice of RTP Payload Format for - WebRTC. If there is a mandatory to implement set of codecs, this - should reference them. In any case, it should reference a discussion - of signalling for the choice of codec, once that discussion reaches - closure.) - Endpoints may signal support for multiple media formats, or multiple + The requirement from Section 6 of [RFC3551] that "Audio applications + operating under this profile SHOULD, at a minimum, be able to send + and/or receive payload types 0 (PCMU) and 5 (DVI4)" applies, since + Section 4.2 of this memo mandates the use of the RTP/SAVPF profile, + which inherits this restriction from the RTP/AVP profile. + + (tbd: there is ongoing discussion on whether support for other audio + and video codecs is to be mandated) + + Endpoints MAY signal support for multiple media formats, or multiple configurations of a single format, provided each uses a different RTP - payload type number. An endpoint that has signalled it's support for + payload type number. An endpoint that has signalled its support for multiple formats is REQUIRED to accept data in any of those formats - at any time, unless it has previously signalled limitations on it's - decoding capability. This is modified if several media types are - sent in the same RTP session, in that case a source (SSRC) is - restricted to switch between any RTP payload format established for - the media type that is being sent by that source; see Section 4.4. - To support rapid rate adaptation, RTP does not require signalling in + at any time, unless it has previously signalled limitations on its + decoding capability. + + This requirement is constrained if several media types 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 + media type that is being sent by that source; see Section 4.4. To + support rapid rate adaptation, RTP does not require signalling in advance for changes between payload formats that were signalled during session setup. + 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). + 4.4. RTP Session Multiplexing An association amongst a set of participants communicating with RTP - is known as an RTP session. A participant may be involved in + is known as an RTP session. A participant can be involved in multiple RTP sessions at the same time. In a multimedia session, each medium has typically been carried in a separate RTP session with its own RTCP packets (i.e., one RTP session for the audio, with a - separate RTP session running on a different transport connection for - the video; if SDP is used, this corresponds to one RTP session for - each "m=" line in the SDP). WebRTC implementations of RTP are - REQUIRED to implement support for multimedia sessions in this way, - for compatibility with legacy systems. + separate RTP session using a different transport address for the + video; if SDP is used, this corresponds to one RTP session for each + "m=" line in the SDP). WebRTC implementations of RTP are REQUIRED to + implement support for multimedia sessions in this way, for + compatibility with legacy systems. In today's networks, however, with the widespread use of Network Address/Port Translators (NAT/NAPT) and Firewalls (FW), it is - desirable to reduce the number of transport layer ports used by real- + desirable to reduce the number of transport addresses used by real- time media applications using RTP by combining multimedia traffic in a single RTP session. (Details of how this is to be done are tbd, but see [I-D.lennox-rtcweb-rtp-media-type-mux], [I-D.holmberg-mmusic-sdp-bundle-negotiation] and [I-D.westerlund-avtcore-multiplex-architecture].) Using a single RTP - session also effects the possibility for differentiated treament of + session also effects the possibility for differentiated treatment of media flows. This is further discussed in Section 12.9. WebRTC implementations of RTP are REQUIRED to support multiplexing of a multimedia session onto a single RTP session according to (tbd). If such RTP session multiplexing is to be used, this MUST be negotiated during the signalling phase. Support for multiple RTP sessions over a single UDP flow as defined by - [I-D.westerlund-avtcore-transport-multiplexing] is RECOMMENDED. + [I-D.westerlund-avtcore-transport-multiplexing] is RECOMMENDED/ + OPTIONAL. -4.5. RTP and RTCP Multiplexing + (tbd: No consensus on the level of including support of Multiple RTP + sessions over a single UDP flow.) - Historically, RTP and RTCP have been run on separate transport-layer - ports (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 must be opened to allow RTP - traffic. To reduce these costs and session setup times, support for - multiplexing RTP data packets and RTCP control packets on a single - port [RFC5761] for each RTP session is REQUIRED. +4.5. RTP and RTCP Multiplexing - (tbd: Are WebRTC implementations required to support the case where - the RTP and RTCP are run on separate UDP ports, for interoperability - with legacy systems?) + 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 setup 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 sending of RTP and RTCP to separate destination + ports. 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 may be useful to keep- - alive NAT bindings, and is the recommend method for application level + 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]. 4.6. Reduced Size RTCP 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, these general 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 and allow for more frequent feedback. Frequent - feedback, in turn, is essential to make real-time application quickly - aware of changing network conditions and allow them to adapt their - transmission and encoding behaviour. Support for RFC5506 is - REQUIRED. + 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 sending + RTCP feedback packets as [RFC5506] non-compound packets is REQUIRED + when signalled. For backwards compatibility, implementations are + also REQUIRED to support the use of compound RTCP feedback packets. 4.7. Symmetric RTP/RTCP To ease traversal of NAT and firewall devices, implementations are - REQUIRED to implement Symmetric RTP [RFC4961]. This requires that - the IP address and port used for sending and receiving RTP and RTCP - packets are identical. The reasons for using symmetric RTP is + REQUIRED to implement and use Symmetric RTP [RFC4961]. This requires + that the IP address and port used for sending and receiving RTP and + RTCP packets are identical. 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 in the form of ports at the end-points, but also in the + 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. -4.8. Generation of the RTCP Canonical Name (CNAME) +4.8. Choice of RTP Synchronisation Source (SSRC) + + Implementations are REQUIRED to support signalled RTP SSRC values, + using the "a=ssrc:" SDP attribute defined in Sections 4.1 and 5 of + [RFC5576], and MUST also support the "previous-ssrc" source attribute + defined in Section 6.2 of [RFC5576]. Other attributes defined in + [RFC5576] MAY be supported. + + Use of the "a=ssrc:" attribute is OPTIONAL. Implementations MUST + support random SSRC assignment, and MUST support SSRC collision + detection and resolution, both according to [RFC3550]. + +4.9. Generation of the RTCP Canonical Name (CNAME) 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 may change if a collision is - detected, or when the RTP application is restarted, it's RTCP CNAME - is meant to stay unchanged, so that RTP endpoints can be uniquely - identified and associated with their RTP media streams. For proper - functionality, each RTP endpoint needs to have a unique RTCP CNAME - value. + (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 a unique RTCP CNAME value. 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, some may find long-term persistent identifiers + 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 method (b) specified in Section 4.2 of "Guidelines for Choosing RTP Control - Protocol (RTCP) Canonical Names (CNAMEs)" [RFC6222] is REQUIRED, - since this addresses both concerns. + Protocol (RTCP) Canonical Names (CNAMEs)" [RFC6222] is RECOMMENDED. + Note, however, that this does not resolve the privacy concern as + there is not sufficient randomness to avoid tracking of an end-point. + + 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 according to the recommended + form above. + + (tbd: there seems to be a growing consensus that the working group + wants randomly-chosen CNAME values; need to reference a draft that + describes how this is to be done) 5. WebRTC Use of RTP: Extensions - There are a number of RTP extensions that are either required 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 strongly recommended within WebRTC. + 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. 5.1. Conferencing Extensions RTP is inherently a group communication protocol. Groups can be implemented using a centralised server, multi-unicast, or using IP multicast. While IP multicast was popular in early deployments, in today's practice, overlay-based conferencing dominates, typically using one or more central servers to connect endpoints in a star or flat tree topology. These central servers can be implemented in a number of ways as discussed in Appendix A, and in the memo on RTP Topologies [RFC5117]. As discussed in Section 3.5 of [RFC5117], the use of a video switching MCU makes the use of RTCP for congestion control, or any type of quality reports, very problematic. Also, as discussed in section 3.6 of [RFC5117], the use of a content modifying MCU with RTCP termination breaks RTP loop detection and removes the ability - for receivers to identify active senders. Accordingly, only RTP - Transport Translators (relays), RTP Mixers, and end-point based - forwarding topologies are supported in WebRTC. These RECOMMENDED - topologies are expected to be supported by all WebRTC end-points - (these three topologies require no special support in the end-point, - if the RTP features mandated in this memo are implemented). + for receivers to identify active senders. RTP Transport Translators + (Topo-Translator) are not of immediate interest to WebRTC, although + the main difference compared to point to point is the possibility of + seeing multiple different transport paths in any RTCP feedback. + 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 protocol extensions to be used with conferencing, described - below, are not required for correctness; an RTP endpoint that does - not implement these extensions will work correctly, but offer poor - performance. Support for the listed extensions will greatly improve - the quality of experience, however, in the context of centralised - conferencing, where one RTP Mixer (Conference Focus) receives a - participants media streams and distribute them to the other - participants. These messages are defined in the 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 RTP extensions described below to be used with centralised + conferencing -- where one RTP Mixer (e.g., a conference bridge) + receives a participant's RTP media streams and distributes them to + the other participants -- are not necessary for interoperability; an + RTP endpoint that does not implement these extensions will work + correctly, but may 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. -5.1.1. Full Intra Request + The RTCP packets assisting in such operation are defined in the + 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]. + +5.1.1. Full Intra Request (FIR) 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 have 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 predicted media encoding with - long prediction chains. It is REQUIRED that this feedback message is - supported by RTP senders in WebRTC, since it greatly improves the - user experience when using centralised mixers-based conferencing. + receivers can decode the video or other predictive media encoding + with long prediction chains. It is REQUIRED that this feedback + message is supported by RTP senders in WebRTC, since it greatly + improves the user experience when using centralised mixers-based + conferencing. -5.1.2. Picture Loss Indication +5.1.2. Picture Loss Indication (PLI) 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 can exist multiple methods to fulfil the - request. It is RECOMMENDED that this feedback message is supported - as a loss tolerance mechanism. + Request above as there there may be multiple methods to fulfill the + request. It is REQUIRED that senders understand and react to this + feedback message as a loss tolerance mechanism; receivers MAY send + PLI messages. -5.1.3. Slice Loss Indication +5.1.3. Slice Loss Indication (SLI) 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 macroblocks, and would like to have these repaired somehow. The use of this feedback message is OPTIONAL as a loss tolerance mechanism. -5.1.4. Reference Picture Selection Indication +5.1.4. Reference Picture Selection Indication (RPSI) 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 synchronicity, a - known-as-correct reference picture can be used for future coding. - The RPSI message allows this to be signalled. The use of this RTCP - feedback message is OPTIONAL as a loss tolerance mechanism. + 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. -5.1.5. Temporary Maximum Media Stream Bit Rate Request + Support for RPSI messages is OPTIONAL. + +5.1.5. Temporal-Spatial Trade-off Request (TSTR) + + 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. + +5.1.6. Temporary Maximum Media Stream Bit Rate Request 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 is used by a media receiver, to inform the sending party that + 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 for various reasons, and can for example - be used by an RTP mixer to limit the media sender being forwarded by - the mixer (without doing media transcoding) to fit the bottlenecks - existing towards the other session participants. It is REQUIRED that - this feedback message is supported. + this receiver. This may have various reasons; for example, an RTP + mixer may 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. It + is REQUIRED that this feedback message is supported. A RTP media + stream sender receiving a TMMBR for its SSRC MUST follow the + limitations set by the message; the sending of TMMBR requests is + OPTIONAL. 5.2. Header Extensions 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. @@ -554,331 +660,221 @@ 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. The rapid synchronisation extensions use the general RTP header extension mechanism [RFC5285], which requires signalling, but are otherwise backwards compatible. -5.2.2. Client to Mixer Audio Level +5.2.2. Client-to-Mixer Audio Level - The Client to Mixer Audio Level [RFC6464] is an RTP header extension - used by a client to inform a mixer about the level of audio activity - in the packet the header is attached to. This enables a central node - to make mixing or selection decisions without decoding or detailed - inspection of the payload. Thus reducing the needed complexity in - some types of central RTP nodes. It can also be used to save - decoding resources in a WebRTC receiver in a mesh topology, which if - it has limited decoding resources, may select to decode only the most - relevant media streams based on audio activity levels. + 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 [I-D.ietf-avtcore-srtp-encrypted-header-ext] since the information contained in these header extensions can be considered sensitive. -5.2.3. Mixer to Client Audio Level +5.2.3. Mixer-to-Client Audio Level 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 + [I-D.ietf-avtcore-srtp-encrypted-header-ext] since the information contained in these header extensions can be considered sensitive. 6. WebRTC Use of RTP: Improving Transport Robustness There are some tools that can make RTP flows robust against Packet - loss and reduce the impact on media quality. However they all add + loss and reduce the impact on media quality. However, they all add extra bits compared to a non-robust stream. These extra bits need to - be considered, and the aggregate bit-rate must be rate controlled. - Thus improving robustness might require a lower base encoding - quality, but has the potential to give that quality with fewer + be considered, and the aggregate bit-rate must be rate-controlled. + Thus, 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. -6.1. Retransmission - - Support for RTP retransmission as defined by "RTP Retransmission - Payload Format" [RFC4588] is RECOMMENDED. - - The retransmission scheme in RTP allows flexible application of - retransmissions. Only selected missing packets can be requested by - the receiver. It also allows for the sender to prioritise between - missing packets based on senders knowledge about their content. - Compared to TCP, RTP retransmission also allows one to give up on a - packet that despite retransmission(s) still has not been received - within a time window. - - "WebRTC Media Transport Requirements" [I-D.cbran-rtcweb-data] raises - two issues that they think makes RTP Retransmission unsuitable for - WebRTC. We here consider these issues and explain why they are in - fact not a reason to exclude RTP retransmission from the tool box - available to WebRTC media sessions. +6.1. Negative Acknowledgements and RTP Retransmission - The additional latency added by [RFC4588] will exceed the latency - threshold for interactive voice and video: RTP Retransmission will - require at least one round trip time for a retransmission request - and repair packet to arrive. Thus the general suitability of - using retransmissions will depend on the actual network path - latency between the end-points. In many of the actual usages the - latency between two end-points will be low enough for RTP - retransmission to be effective. Interactive communication with - end-to-end delays of 400 ms still provide a fair quality. Even - removing half of that in end-point delays allows functional - retransmission between end-points on the same continent. In - addition, some applications may accept temporary delay spikes to - allow for retransmission of crucial codec information such an - parameter sets, intra picture etc, rather than getting no media at - all. + As a consequence of supporting the RTP/SAVPF profile, implementations + will support negative acknowlegdements (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. - The undesirable increase in packet transmission at the point when - congestion occurs: Congestion loss will impact the rate controls - view of available bit-rate for transmission. When using - retransmission one will have to prioritise between performing - retransmissions and the quality one can achieve with ones - adaptable codecs. In many use cases one prefer error free or low - rates of error with reduced base quality over high degrees of - error at a higher base quality. + 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 should 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 WebRTC end-point implementations will need to both select when to - enable RTP retransmissions based on API settings and measurements of - the actual round trip time. In addition for each NACK request that a - media sender receives it will need to make a prioritisation based on - the importance of the requested media, the probability that the - packet will reach the receiver in time for being usable, the - consumption of available bit-rate and the impact of the media quality - for new encodings. + 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). 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 may be 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. - To conclude, the issues raised are implementation concerns that an - implementation needs to take into consideration, they are not - arguments against including a highly versatile and efficient packet - loss repair mechanism. + 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 is not expected to retransmit every NACKed packet. 6.2. Forward Error Correction (FEC) - Support of some type of FEC to combat the effects of packet loss is - beneficial, but is heavily application dependent. However, some FEC - mechanisms are encumbered. - - The main benefit from FEC is the relatively low additional delay - needed to protect against packet losses. The transmission of any - repair packets should preferably be done with a time delay that is - just larger than any loss events normally encountered. That way the - repair packet isn't also lost in the same event as the source data. - - The amount of repair packets needed varies depending on the amount - and pattern of packet loss to be recovered, and on the mechanism used - to derive repair data. The later choice also effects the the - additional delay required to both encode the repair packets and in - the receiver to be able to recover the lost packet(s). - -6.2.1. Basic Redundancy - - The method for providing basic redundancy is to simply retransmit a - some time earlier sent packet. This is relatively simple in theory, - i.e. one saves any outgoing source (original) packet in a buffer - marked with a timestamp of actual transmission, some X ms later one - transmit this packet again. Where X is selected to be longer than - the common loss events. Thus any loss events shorter than X can be - recovered assuming that one doesn't get an another loss event before - all the packets lost in the first event has been received. - - The downside of basic redundancy is the overhead. To provide each - packet with once chance of recovery, then the transmission rate - increases with 100% as one needs to send each packet twice. It is - possible to only redundantly send really important packets thus - reducing the overhead below 100% for some other trade-off is - overhead. - - In addition the basic retransmission of the same packet using the - same SSRC in the same RTP session is not possible in RTP context. - The reason is that one would then destroy the RTCP reporting if one - sends the same packet twice with the same sequence number. Thus one - needs more elaborate mechanisms. - - RTP Payload Format Support: Some RTP payload format do support basic - redundancy within the RTP paylaod format itself. Examples are - AMR-WB [RFC4867] and G.719 [RFC5404]. - - RTP Payload for Redundant Audio Data: This audio and text redundancy - format defined in [RFC2198] allows for multiple levels of - redundancy with different delay in their transmissions, as long as - the source plus payload parts to be redundantly transmitted - together fits into one MTU. This should work fine for most - interactive audio and text use cases as both the codec bit-rates - and the framing intervals normally allow for this requirement to - hold. This payload format also don't increase the packet rate, as - original data and redundant data are sent together. This format - does not allow perfect recovery, only recovery of information - deemed necessary for audio, for example the sequence number of the - original data is lost. - - RTP Retransmission Format: The RTP Retransmission Payload format - [RFC4588] can be used to pro-actively send redundant packets using - either SSRC or session multiplexing. By using different SSRCs or - a different session for the redundant packets the RTCP receiver - reports will be correct. The retransmission payload format is - used to recover the packets original data thus enabling a perfect - recovery. - - Duplication Grouping Semantics in the Session Description Protocol: - This [I-D.begen-mmusic-redundancy-grouping] is proposal for new - SDP signalling to indicate media stream duplication using - different RTP sessions, or different SSRCs to separate the source - and the redundant copy of the stream. - -6.2.2. Block Based FEC - - Block based redundancy collects a number of source packets into a - data block for processing. The processing results in some number of - repair packets that is then transmitted to the other end allowing the - receiver to attempt to recover some number of lost packets in the - block. The benefit of block based approaches is the overhead which - can be lower than 100% and still recover one or more lost source - packet from the block. The optimal block codes allows for each - received repair packet to repair a single loss within the block. - Thus 3 repair packets that are received should allow for any set of 3 - packets within the block to be recovered. In reality one commonly - don't reach this level of performance for any block sizes and number - of repair packets, and taking the computational complexity into - account there are even more trade-offs to make among the codes. - - One result of the block based approach is the extra delay, as one - needs to collect enough data together before being able to calculate - the repair packets. In addition sufficient amount of the block needs - to be received prior to recovery. Thus additional delay are added on - both sending and receiving side to ensure possibility to recover any - packet within the block. - - The redundancy overhead and the transmission pattern of source and - repair data can be altered from block to block, thus allowing a - adaptive process adjusting to meet the actual amount of loss seen on - the network path and reported in RTCP. - - The alternatives that exist for block based FEC with RTP are the - following: - - RTP Payload Format for Generic Forward Error Correction: This RTP - payload format [RFC5109] defines an XOR based recovery packet. - This is the simplest processing wise that an block based FEC - scheme can be. It also results in some limited properties, as - each repair packet can only repair a single loss. To handle - multiple close losses a scheme of hierarchical encodings are need. - Thus increasing the overhead significantly. - - Forward Error Correction (FEC) Framework: This framework - [I-D.ietf-fecframe-framework] defines how not only RTP packets but - how arbitrary packet flows can be protected. Some solutions - produced or under development in FECFRAME WG are RTP specific. - There exist alternatives supporting block codes such as Reed- - Salomon and Raptor. + 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 should be noted that using redundancy + encoding or FEC will lead to increased playout delay, which should be + considered when choosing the redundancy or FEC formats and their + respective parameters. -6.2.3. Recommendations for FEC + 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. - Open Issue: Decision of need for FEC and if to be included in - recommendation which FEC scheme to be supported needs to be - documented. + 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. 7. WebRTC Use of RTP: Rate Control and Media Adaptation WebRTC will be used in very varied network environment with a - hetrogenous set of link technologies, including wired and wireless, + heterogeneous set of link technologies, including wired and wireless, interconnecting peers at different topological locations resulting in network paths with widely varying one way delays, bit-rate capacity, - load levels and traffic mixes. In addition individual end-points + load levels and traffic mixes. In addition, individual end-points will open one or more WebRTC sessions between one or more peers. Each of these session may contain different mixes of media and data - flows. Assymetric usage of media bit-rates and number of media + flows. Asymmetric usage of media bit-rates and number of RTP media streams is also to be expected. A single end-point may receive zero - to many simultanous media streams while itself transmitting one or - more streams. + to many simultaneous RTP media streams while itself transmitting one + or more streams. The WebRTC application is very dependent from a quality perspective - on the media adapation working well so that an end-point doesn't + on the media adaptation working well so that an end-point doesn't transmit significantly more than the path is capable of handling. If it would, the result would be high levels of packet loss or delay - spikes causing media degradations. + spikes causing media quality degradation. - WebRTC applications using more than a single media stream of any - media type or data flows has an additional concern. In this case the - different flows should try to avoid affecting each other negatively. - In addition in case there is a resource limiation, the available - resources needs to be shared. How to share them is something the - application should prioritize so that the limiation in quality or - capabilities are the ones that provide the least affect on the - application. + WebRTC applications using more than a single RTP media stream of any + media type or data flows have an additional concern. In this case, + the different flows should try to avoid affecting each other + negatively. In addition, in case there is a resource limitation, the + available resources need to be shared. How to share them is + something the application should prioritize so that the limitations + in quality or capabilities are those that have the least impact on + the application. - This hetrogenous situation results in a requirement to have + Overall, the diversity of operating environments lead to the need for functionality that adapts to the available capacity and that competes fairly with other network flows. If it would not compete fairly enough WebRTC could be used as an attack method for starving out other traffic on specific links as long as the attacker is able to - create traffic across a specific link. This is not far-fetched for a - web-service capable of attracting large number of end-points and use - the service, combined with BGP routing state a server could pick - client pairs to drive traffic to specific paths. + create traffic across the links in question. A possible attack + scenario is to use a web-service capable of attracting large numbers + of end-points, combined with BGP routing state to have the server + pick client pairs to drive traffic to specific paths. - The above estalish a clear need based on several reasons why there - need to be a well working media adaptation mechanism. This mechanism - also have a number of requirements on what services it should provide - and what performance it needs to provide. + The above clearly motivates the need for a well working media + adaptation mechanism. This mechanism also have a number of + requirements on what services it should provide and what performance + it needs to provide. The biggest issue is that there are no standardised and ready to use - mechanism that can simply be included in WebRTC. Thus there will be - need for the IETF to produce such a specification. Therefore the + mechanism that can simply be included in WebRTC. Thus, there will be + a need for the IETF to produce such a specification. Therefore, the suggested way forward is to specify requirements on any solution for - the media adaptation. These requirements is for now proposed to be - documented in this specification. In addition a proposed detailed + the media adaptation. For now, we propose that these requirements be + documented in this specification. In addition, a proposed detailed solution will be developed, but is expected to take longer time to finalize than this document. 7.1. Congestion Control Requirements Requirements for congestion control of WebRTC sessions are discussed in [I-D.jesup-rtp-congestion-reqs]. Implementations are REQUIRED to implement the RTP circuit breakers described in [I-D.perkins-avtcore-rtp-circuit-breakers]. + (tbd: Should add the RTP/RTCP Mechanisms that an WebRTC + implementation is required to support. Potential candidates include + Transmission Timestamps (RFC 5450).) + 7.2. Rate Control Boundary Conditions The session establishment signalling will establish certain boundary that the media bit-rate adaptation can act within. First of all the set of media codecs provide practical limitations in the supported bit-rate span where it can provide useful quality, which packetization choices that exist. Next the signalling can establish maximum media bit-rate boundaries using SDP b=AS or b=CT. -7.3. RTCP Limiations + (tbd: This section needs expanding on how to use these limits) + +7.3. RTCP Limitations for Congestion Control 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 capabilities of real-time media - traffic to adapt to changing path conditions may be less rapid than - for the elastic applications TCP was designed for, but frequent - feedback is still required to allow the congestion control algorithm - to track the path dynamics. + round trip time. We note that the real-time media traffic may 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 required 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 media stream bit rate thus limits the maximum feedback rate as a - function of the mean RTCP packet size. + The RTP media stream bit rate thus limits the maximum feedback rate + as a function of the mean RTCP packet size. Interactive communication may not be able to afford waiting for packet losses to occur to indicate congestion, because an increase in playout delay due to queuing (most prominent in wireless networks) may easily lead to packets being dropped due to late arrival at the receiver. Therefore, more sophisticated cues may 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 @@ -886,240 +882,237 @@ 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. -7.4. Legacy Interop Limitations - - Congestion control interoperability with most type of legacy devices, - even using an translator could be difficult. There are numerous - reasons for this: - - No RTCP Support: There exist legacy implementations that does not - even implement RTCP at all. Thus no feedback at all is provided. - - RTP/AVP Minimal RTCP Interval of 5s: RTP [RFC3550] under the RTP/AVP - profile specifies a recommended minimal fixed interval of 5 - seconds. Sending RTCP report blocks as seldom as 5 seconds makes - it very difficult for a sender to use these reports and react to - any congestion event. +7.4. Congestion Control Interoperability With Legacy Systems - RTP/AVP Scaled Minimal Interval: If a legacy device uses the scaled - minimal RTCP compound interval, the "RECOMMENDED value for the - reduced minimum in seconds is 360 divided by the session bandwidth - in kilobits/second" ([RFC3550], section 6.2). The minimal - interval drops below a second, still several times the RTT in - almost all paths in the Internet, when the session bandwidht - becomes 360 kbps. A session bandwidth of 1 Mbps still has a - minimal interval of 360 ms. Thus, with the exception for rather - high bandwidth sessions, getting frequent enough RTCP Report - Blocks to report on the order of the RTT is very difficult as long - as the legacy device uses the RTP/AVP profile. + 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 must 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. - RTP/AVPF Supporting Legacy Device: If a legacy device supports RTP/ - AVPF, then that enables negotation 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]. + 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 are required + to interwork with such end-points MUST ensure that they keep within + the RTP circuit breaker [I-D.perkins-avtcore-rtp-circuit-breakers] + constraints to limit the congestion they can cause. - It has been suggested on the WebRTC mailing list that if - interoperating with really limited legacy devices an WebRTC end-point - may not send more than 64 kbps of media streams, to avoid it causing - massive congestion on most paths in the Internet when communicating - with a legacy node not providing sufficient feedback for effective - congestion control. This warrants further discussion as there is - clearly a number of link layers that don't even provide that amount - of bit-rate consistently, and that assumes no competing traffic. + 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 are required to interwork with such + end-points MUST ensure that they stay within the RTP circuit breaker + [I-D.perkins-avtcore-rtp-circuit-breakers] constraints to limit the + congestion they can cause, but may find that they can achieve better + congestion response depending on the amount of feedback that is + available. 8. WebRTC Use of RTP: Performance Monitoring - RTCP does contains a basic set of RTP flow monitoring points like - packet loss and jitter. There exist a number of extensions that - could be included in the set to be supported. However, in most cases - which RTP monitoring that is needed depends on the application, which - makes it difficult to select which to include when the set of - applications is very large. + RTCP does contains a basic set of RTP flow monitoring metrics like + packet loss and jitter. There are a number of extensions that could + be included in the set to be supported. However, in most cases which + RTP monitoring that is needed depends on the application, which makes + it difficult to select which to include when the set of applications + is very large. Exposing some metrics in the WebRTC API should be considered allowing the application to gather the measurements of interest. However, security implications for the different data sets exposed will need to be considered in this. + (tbd: If any RTCP XR metrics should be added is still an open + question, but possible to extend at a later stage) + 9. WebRTC Use of RTP: Future Extensions 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 should adopt RTP extensions that are of general utility, to enable easy gatewaying to other applications using RTP, rather than adopt mechanisms that are - narrowly targetted at specific WebRTC use cases. + narrowly targeted at specific WebRTC use cases. 10. Signalling Considerations RTP is built with the assumption of an external signalling channel that can be used to configure the RTP sessions and their features. The basic configuration of an RTP session consists of the following parameters: RTP Profile: The name of the RTP profile to be used in session. The RTP/AVP [RFC3551] and RTP/AVPF [RFC4585] profiles can interoperate on basic level, as can their secure variants RTP/SAVP [RFC3711] and RTP/SAVPF [RFC5124]. The secure variants of the profiles do not directly interoperate with the non-secure variants, due to the presence of additional header fields in addition to any cryptographic transformation of the packet content. As WebRTC - requires the usage of the SAVPF profile only a single profile will - need to be signalled. Interworking functions may transform this - into SAVP for a legacy use case by indicating to the WebRTC end- - point a SAVPF end-point and limiting the usage of the a=rtcp - attribute to indicate a trr-int value of 4 seconds. + requires the usage of the RTP/SAVPF profile this can be inferred + as there is only a single profile, but in SDP this is still + required information to be signalled. Interworking functions may + transform this into RTP/SAVP for a legacy use case by indicating + to the WebRTC end-point a RTP/SAVPF end-point and limiting the + usage of the a=rtcp attribute to indicate a trr-int value of 4 + seconds. - Transport Information: Source and destination address(s) and ports - for RTP and RTCP MUST be signalled for each RTP session. In - WebRTC these end-points will be provided by ICE that signalls - candidates and arrive at nominated candidate pairs. If RTP and - RTCP multiplexing [RFC5761] is to be used, such that a single port - is used for RTP and RTCP flows, this MUST be signalled (see - Section 4.5). If several RTP sessions are to be multiplexed onto - a single transport layer flow, this MUST also be signalled (see - Section 4.4). + Transport Information: Source and destination IP address(s) and + ports for RTP and RTCP MUST be signalled for each RTP session. In + WebRTC these transport addresses will be provided by ICE that + signals candidates and arrives at nominated candidate address + pairs. If RTP and RTCP multiplexing [RFC5761] is to be used, such + that a single port is used for RTP and RTCP flows, this MUST be + signalled (see Section 4.5). If several RTP sessions are to be + multiplexed onto a single transport layer flow, this MUST also be + signalled (see Section 4.4). RTP Payload Types, media formats, and media format parameters: The mapping between media type names (and hence the RTP - payload formats to be used) and the RTP payload type numbers must - be signalled. Each media type may also have a number of media - type parameters that must also be signalled to configure the codec + payload formats to be used) and the RTP payload type numbers MUST + be signalled. Each media type MAY also have a number of media + type parameters that MUST also be signalled to configure the codec and RTP payload format (the "a=fmtp:" line from SDP). - RTP Extensions: The RTP extensions one intends to use need to be - agreed upon, including any parameters for each respective - extension. At the very least, this will help avoiding using - bandwidth for features that the other end-point will ignore. But - for certain mechanisms there is requirement for this to happen as - interoperability failure otherwise happens. + RTP Extensions: The RTP extensions to be used SHOULD be agreed upon, + including any parameters for each respective extension. At the + very least, this will help avoiding using bandwidth for features + that the other end-point will ignore. But for certain mechanisms + there is requirement for this to happen as interoperability + failure otherwise happens. RTCP Bandwidth: Support for exchanging RTCP Bandwidth values to the - end-points will be necessary, as described in "Session Description - Protocol (SDP) Bandwidth Modifiers for RTP Control Protocol (RTCP) - Bandwidth" [RFC3556], or something semantically equivalent. This - also ensures that the end-points have a common view of the RTCP - bandwidth, this is important as too different view of the - bandwidths may lead to failure to interoperate. + end-points will be necessary. This SHALL be done as described in + "Session Description Protocol (SDP) Bandwidth Modifiers for RTP + Control Protocol (RTCP) Bandwidth" [RFC3556], or something + semantically equivalent. This also ensures that the end-points + have a common view of the RTCP bandwidth, this is important as too + different view of the bandwidths may lead to failure to + interoperate. 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 somehow, 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. + 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. 11. WebRTC API Considerations The following sections describe how the WebRTC API features map onto the RTP mechanisms described in this memo. 11.1. API MediaStream to RTP Mapping - The WebRTC API and its media function have the concept of a - MediaStream that consists of zero or more tracks. Where a track is - an individual stream of media from any type of media source like a - microphone or a camera, but also coneptual sources, like a audio mix - or a video composition. The tracks within a MediaStream are expected - to be synchronized. + The WebRTC API and its media function have the concept of a WebRTC + MediaStream that consists of zero or more tracks. A track 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 tracks within a WebRTC + MediaStream are expected to be synchronized. - A track correspondes to the media received with one particular SSRC. + A track correspond to the media received with one particular SSRC. There might be additional SSRCs associated with that SSRC, like for RTP retransmission or Forward Error Correction. However, one SSRC - will identify a media stream and its timing. + will identify an RTP media stream and its timing. - Thus a MediaStream is a collection of SSRCs carrying the different - media included in the synchornized aggregate. Thus also the - synchronization state associated with the included SSRCs are part of - concept. One important thing to consider is that there can be - multiple different MediaStreams containing a given Track (SSRC). - Thus to avoid unnecessary duplication of media at transport level one - need to do the binding of which MediaStreams a given SSRC is - associated with at signalling level. + As a result, a WebRTC MediaStream is a collection of SSRCs carrying + the different media included in the synchronised aggregate. + Therefore, also the synchronization state associated with the + included SSRCs are part of concept. It is important to consider that + there can be multiple different WebRTC MediaStreams containing a + given Track (SSRC). To avoid unnecessary duplication of media at the + transport level in such cases, a need arises for a binding defining + which WebRTC MediaStreams a given SSRC is associated with at the + signalling level. - A proposal for how the binding between MediaStreams and SSRC can be - done exist in "Cross Session Stream Identification in the Session - Description Protocol" [I-D.alvestrand-rtcweb-msid]. + A proposal for how the binding between WebRTC MediaStreams and SSRC + can be done is specified in "Cross Session Stream Identification in + the Session Description Protocol" [I-D.alvestrand-rtcweb-msid]. + + (tbd: This text must be improved and achieved consensus on. Interim + meeting in June 2012 shows large differences in opinions.) 12. RTP Implementation Considerations The following provide some guidance on the implementation of the RTP features described in this memo. This section discusses RTP functionality that is part of the RTP standard, required by decisions made, or to enable use cases raised - and their motivations. This discussion is done from an WebRTC end- - point perspective. It will occassional go into central nodes, but as - the specification is for an end-point that is where the focus lies. + and their motivations. This discussion is from an WebRTC end-point + perspective. It will occasionally talk about central nodes, but as + this specification is for an end-point, this is where the focus lies. For more discussion on the central nodes and details about RTP - topologies please reveiw Appendix A. + topologies please see Appendix A. The section will touch on the relation with certain RTP/RTCP extensions, but will focus on the RTP core functionality. The definition of what functionalities and the level of requirement on implementing it is defined in Section 2. 12.1. RTP Sessions and PeerConnection An RTP session is an association among RTP nodes, which have one common SSRC space. An RTP session can include any number of end- points and nodes sourcing, sinking, manipulating or reporting on the - media streams being sent within the RTP session. A PeerConnection - being a point to point association between an end-point and another - node. That peer node may be both an end-point or centralized - processing node of some type, thus the RTP session may terminate - immediately on the far end of the PeerConnection, but it may also - continue as further discused below in Multiparty (Section 12.3) and - Multiple RTP End-points (Section 12.7). + RTP media streams being sent within the RTP session. A + PeerConnection being a point-to-point association between an end- + point and another node. That peer node may be both an end-point or + centralized processing node of some type; thus, the RTP session may + terminate immediately on the far end of the PeerConnection, but it + may also continue as further discussed below in Multiparty + (Section 12.3) and Multiple RTP End-points (Section 12.7). A PeerConnection can contain one or more RTP session depending on how it is setup and how many UDP flows it uses. A common usage has been to have one RTP session per media type, e.g. one for audio and one - for Video, each sent over different UDP flows. However, the default + for video, each sent over different UDP flows. However, the default usage in WebRTC will be to use one RTP session for all media types. This usage then uses only one UDP flow, as also RTP and RTCP multiplexing is mandated (Section 4.5). However, for legacy - interworking and network prioritization (Section 12.9) based on flows - a WebRTC end-point needs to support a mode of operation where one RTP - session per media type is used. Currently each RTP session must use - its own UDP flow. Discussion are ongoing if a solution enabling - multiple RTP sessions over a single UDP flow, see Section 4.4. + interworking and network prioritization (Section 12.9) based on + flows, a WebRTC end-point needs to support a mode of operation where + one RTP session per media type is used. Currently, each RTP session + must use its own UDP flow. Discussions are ongoing if a solution + enabling multiple RTP sessions over a single UDP flow, see + Section 4.4. - The multi-unicast or mesh based multi-party topology (Figure 1) is - best to raise in this section as it concers the relation between RTP + The multi-unicast- or mesh-based multi-party topology (Figure 1) is a + good example for this section as it concerns the relation between RTP sessions and PeerConnections. In this topology, each participant sends individual unicast RTP/UDP/IP flows to each of the other participants using independent PeerConnections in a full mesh. This topology has the benefit of not requiring central nodes. The downside is that it increases the used bandwidth at each sender by - requiring one copy of the media streams for each participant that are - part of the same session beyond the sender itself. Hence, this + requiring one copy of the RTP media streams for each participant that + are part of the same session beyond the sender itself. Hence, this topology is limited to scenarios with few participants unless the media is very low bandwidth. +---+ +---+ | A |<---->| B | +---+ +---+ ^ ^ \ / \ / v v @@ -1133,346 +1126,337 @@ session, spanning multiple peer-to-peer transport layer connections, or as several pairwise RTP sessions, one between each pair of peers. To maintain a coherent mapping between the relation between RTP sessions and PeerConnections we recommend that one implements this as individual RTP sessions. The only downside is that end-point A will not learn of the quality of any transmission happening between B and C based on RTCP. This has not been seen as a significant downside as no one has yet seen a clear need for why A would need to know about the B's and C's communication. An advantage of using separate RTP sessions is that it enables using different media bit-rates to the - differnt peers, thus not forcing B to endure the same quality - reductions if there are limiations in the transport from A to C as C + different peers, thus not forcing B to endure the same quality + reductions if there are limitations in the transport from A to C as C will. 12.2. Multiple Sources A WebRTC end-point may have multiple cameras, microphones or audio - inputs thus a single end-point can source multiple media streams - concurrently of the same media type. In addition the above discussed - criteria to support multiple media types in one single RTP session - results that also an end-point that has one audio and one video - source still need two transmit using two SSRCs concurrently. As - multi-party conferences are supported, as discussed below in - Section 12.3, a WebRTC end-point will need to be capable of - receiving, decoding and playout multiple media streams of the same - type concurrently. + inputs and thus a single end-point can source multiple RTP media + streams of the same media type concurrently. Even if an end-point + does not have multiple media sources of the same media type it will + be required to support transmission using multiple SSRCs concurrently + in the same RTP session. This is due to the requirement on an WebRTC + end-point to support multiple media types in one RTP session. For + example, one audio and one video source can result in the end-point + sending with two different SSRCs in the same RTP session. As multi- + party conferences are supported, as discussed below in Section 12.3, + a WebRTC end-point will need to be capable of receiving, decoding and + playout multiple RTP media streams of the same type concurrently. - Open Issue:Are any mechanism needed to signal limiations in the - number of SSRC that an end-point can handle? + tbd: Are any mechanism needed to signal limitations in the number of + SSRC that an end-point can handle? 12.3. Multiparty - There exist numerous situations and clear use cases for WebRTC - supporting sessions supoprting multi-party. This can be realized in - a number of ways using a number of different implementations - strategies. This focus on the different set of WebRTC end-point - requirements that arise from different sets of multi-party - topologies. + There are numerous situations and clear use cases for WebRTC + supporting RTP sessions supporting multi-party. This can be realized + in a number of ways using a number of different implementation + strategies. In the following, the focus is on the different set of + WebRTC end-point requirements that arise from different sets of + multi-party topologies. - The multi-unicast mesh (Figure 1) based multi-party topoology - discussed above provides a non-centralized solution but can easily - tax the end-points outgoing paths. It may also consume large amount - of encoding resources if each outgoing stream is specifically - encoded. If an encoding is transmitted to multiple parties, either - as in the mesh case or when using relaying central nodes (see below) - a requirement on the end-point becomes to be able to create media - streams suitable to multiple destinations requirements. These - requirements may both be dependent on transport path and the - different end-points preferences related to playout of the media. + The multi-unicast mesh (Figure 1)-based multi-party topology + discussed above provides a non-centralized solution but may incur a + heavy tax on the end-points' outgoing paths. It may also consume + large amount of encoding resources if each outgoing stream is + specifically encoded. If an encoding is transmitted to multiple + parties, as in some implementations of the mesh case, a requirement + on the end-point becomes to be able to create RTP media streams + suitable for multiple destinations requirements. These requirements + may both be dependent on transport path and the different end-points + preferences related to playout of the media. +---+ +------------+ +---+ | A |<---->| |<---->| B | +---+ | | +---+ | Mixer | +---+ | | +---+ | C |<---->| |<---->| D | +---+ +------------+ +---+ Figure 2: RTP Mixer with Only Unicast Paths A Mixer (Figure 2) is an RTP end-point that optimizes the - transmission of media streams from certain perspectives, either by - only sending some of the received media stream to any given receiver - or by providing a combined media stream out of a set of contributing - streams. There exist various methods of implementation as discussed - in Appendix A.3. A common aspect is that these central nodes a - number of tools to control the media encoding provided by a WebRTC - end-point. This includes functions like requesting breaking the - encoding chain and have the encoder produce a so called Intra frame. - Another is limiting the bit-rate of a given stream to better suit the - mixer view of the multiple down-streams. Others are controling the - most suitable frame-rate, picture resultion, the trade-off between - frame-rate and spatial quality. + transmission of RTP media streams from certain perspectives, either + by only sending some of the received RTP media stream to any given + receiver or by providing a combined RTP media stream out of a set of + contributing streams. There are various methods of implementation as + discussed in Appendix A.3. A common aspect is that these central + nodes may use a number of tools to control the media encoding + provided by a WebRTC end-point. This includes functions like + requesting breaking the encoding chain and have the encoder produce a + so called Intra frame. Another is limiting the bit-rate of a given + stream to better suit the mixer view of the multiple down-streams. + Others are controlling the most suitable frame-rate, picture + resolution, the trade-off between frame-rate and spatial quality. A mixer gets a significant responsibility to correctly perform - congestion control, identity management, manage synchronization while - providing a for the application suitable media optimization. + congestion control, source identification, manage synchronization + while providing the application with suitable media optimizations. - Mixers also need to be a trusted node when it comes to security as it + Mixers also need to be trusted nodes when it comes to security as it manipulates either RTP or the media itself before sending it on - towards the end-point(s) thus must be able to decrypt and then - encrypt it before sending it out. There exist one type of central - node, the relay that one doesn't need to trust with the keys to the - media. The relay operates only on the IP/UDP level of the transport. - It is configured so that it would forward any RTP/RTCP packets from A - to the other participants B-D. - - +---+ +---+ - | | +-----------+ | | - | A |<------->| DTLS-SRTP |<------->| C | - | |<-- -->| HOST |<-- -->| | - +---+ \ / +-----------+ \ / +---+ - X X - +---+ / \ +-----------+ / \ +---+ - | |<-- -->| RTP |<-- -->| | - | B |<------->| RELAY |<------->| D | - | | +-----------+ | | - +---+ +---+ - - Figure 3: DTLS-SRTP host and RTP Relay Separated - - To accomplish the security properties discussed above using a relay - one need to have a separate key handling server and also support for - distribute the different keys such as Encrypted Key Transport - [I-D.ietf-avt-srtp-ekt]. The relay also creates a situation where - there is multiple end-points visible in the RTCP reporting and any - feedback events. Thus becoming yet another situation in addition to - Mesh where the end-point will have to have logic for merging - different requirements and preferences. This is more detail - discussed in Section 12.7. - - +---+ +---+ +---+ - | A |--->| B |--->| C | - +---+ +---+ +---+ - - Figure 4: MediaStream Forwarding - - The above Figure 4 depicts a possible scenario where an WebRTC end- - point (A) sends a media stream to B. B decides to forward the media - stream to C. This can either be realized in B (WebRTC end-point) - using a simple relay functionality creating similar consideration and - implementation requirements. Another implmentation strategy in B - could be to select to transcode the media from A to C, thus breaking - most of the dependecies between A and C. In that case A is not - required to be aware of B forwarding the media to C. + towards the end-point(s), thus they must be able to decrypt and then + encrypt it before sending it out. 12.4. SSRC Collision Detection The RTP standard [RFC3550] requires any RTP implementation to have - support for detecting and handling SSRC collisions, i.e. when two - different end-points uses the same SSRC value. This requirement - applies also to WebRTC end-points. There exist several scenarios - where SSRC collisions may occur. + 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 may occur. - In a point to point session where each SSRC are associated with - either of the two end-points and where the main media carrying SSRC - identifier will be announced in the signalling there is less likely - to occur due to the information about used SSRCs provided by Source- - Specific SDP Attributes [RFC5576]. Still if both end-points starts - 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. + 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 unsignalled SSRCs, this may be considered a - bug. This is more likely than it appears as certain RTP - functionalities need extra SSRCs to provide functionality related to - another SSRC, for example SSRC multiplexed RTP retransmission - [RFC4588]. In those cases an end-point can create a new SSRC which - strictly don't need to be announced over the signalling channel to - function correctly on both RTP and PeerConnection level. + There could also appear unsignalled SSRCs. 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 PeerConnection level. - The more likely cases for SSRC collision is that multiple end-points - in an multiparty creates new soruces and signalls those towards the - central server. In cases where the SSRC/CSRC are propogated between - the different end-points from the central node collisions can occur. + 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 manage to connect an end- - points PeerConnection to another PeerConnectio the end-point it has. - Thus forming a loop where the end-point will receive its own traffic. - This must be considered a bug, but still if it occurs it is important - that the end-point can handle the situation. + Another scenario is when the central node manages to connect an end- + point's PeerConnection to another PeerConnection 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. 12.5. Contributing Sources - Contributing Sources (CSRC) is a functionality in RTP header that - enables a RTP node combing multiple sources into one to identify the - sources that has gone into the combination. For WebRTC end-point the - support of contributing sources are trivial. The set of CSRC are - provided for a given RTP packet. This information can then be - exposed towards the applications using some form of API, most likely - a mapping back into MediaStream identities to avoid having to expose - two namespaces and the handling of SSRC collision handling to the - JavaScript. + Contributing Sources (CSRC) is a functionality in the RTP header that + allows an RTP node to combine media packets from multiple sources + into one and to identify which sources yielded the result. For + WebRTC end-points, supporting contributing sources is trivial. The + set of CSRCs is provided in a given RTP packet. This information can + then be exposed to the applications using some form of API, possibly + a mapping back into WebRTC MediaStream identities to avoid having to + expose two namespaces and the handling of SSRC collision handling to + the JavaScript. - There are also at least one extension that is dependent on the CRSRC - list being used, that is the Mixer to client audio level [RFC6465], - that enhances the information provided by the CSRC to actual energy - levels for audio for each contributing source. + (tbd: should the API provide the ability to add a CSRC list to an + outgoing packet? this is only useful if the sender is mixing content) + + There are also at least one extension that depends on the CRSRC list + being used: the Mixer-to-client audio level [RFC6465], which enhances + the information provided by the CSRC to actual energy levels for + audio for each contributing source. 12.6. Media Synchronization - When an end-point has more than one media source being sent one need - to consider if these media source are to be synchronized. In RTP/ - RTCP synchronziation is provided by having a set of media streams be - indicated as comming from the same synchroniztion context and logical - end-point by using the same CNAME identifier. + 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 CNAME + identifier. - The next provision is that all media sources internal clock, i.e. - what drives the RTP timestamp can be correlated with a system clock - that is provided in RTCP Sender Reports encoded in an NTP format. By - having the RTP timestamp to system clock being provided for all - sources the relation of the different media stream, also across - multiple RTP sessions can if chosen to be synchronized. The - requirement is for the media sender to provide the information, the - receiver can chose to use it or not. + 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. 12.7. Multiple RTP End-points - A number of usages of RTP discussed here results in that an WebRTC - end-point sending media in an RTP session out over an PeerConnection - will receive receiver reports from multiple RTP receiving nodes. - Note that receiving multiple receiver reports are expected due to - that any RTP node that has multiple SSRCs are required to report on + Some usages of RTP beyond the recommend topologies result in that an + WebRTC end-point sending media in an RTP session out over a single + PeerConnection will receive receiver reports from multiple RTP + receivers. Note that receiving multiple receiver reports is expected + because any RTP node that has multiple SSRCs is required to report to the media sender. The difference here is that they are multiple - nodes, and thus will have different path characteristics. + nodes, and thus will likely have different path characteristics. - The topologies relevant to WebRTC when this can occur are centralized - relay and a end-point forwarding a media stream. Mixers are expected - to not forward media stream reports across itself due to the - difference in the media stream provided to different end-points which - the original media source lacks information about the mixers - manipulation. + RTP Mixers may create a situation where an end-point experiences a + situation in-between a session with only two end-points and multiple + end-points. Mixers are expected to not forward RTCP reports + regarding RTP media streams across themselves. This is due to the + difference in the RTP media streams provided to the different end- + points. The original media source lacks information about a mixer's + manipulations prior to sending it the different receivers. This + setup also results in that an end-point's feedback or requests goes + to the mixer. When the mixer can't act on this by itself, it is + forced to go to the original media source to fulfill the receivers + request. This will not necessarily be explicitly visible any RTP and + RTCP traffic, but the interactions and the time to complete them will + indicate such dependencies. - Having multiple RTP nodes receive ones RTP flow and send reports and - feedback about it has several impacts. As previously discussed - (Section 12.3) any codec control and rate control needs to be capable - of merging the requirements and preferences to provide a single best - according to the situation media stream. Specifically when it comes - to congestion control it needs to be capable of identifying the + The topologies in which an end-point receives receiver reports from + multiple other end-points are the centralized relay, multicast and an + end-point forwarding an RTP media stream. Having multiple RTP nodes + receive an RTP flow and send reports and feedback about it has + several impacts. As previously discussed (Section 12.3) any codec + control and rate control needs to be capable of merging the + requirements and preferences to provide a single best encoding + according to the situation RTP media stream. Specifically, when it + comes to congestion control it needs to be capable of identifying the different end-points to form independent congestion state information for each different path. - Providing source authentication in multi-party is a challange. In - the mixer based topologies an end-points source authentication is - based on verifying that media comes from the mixer by cryptographic - verification and secondly trust the mixer to correctly identify any - source towards the end-point. In RTP sessions where multiple end- - points are directly visible to an end-point all end-points have - knowledge about each others master keys, and can thus inject packets - claimed to come from another end-point in the session. Any node - performing relay can perform non-cryptographic mitigation by - preventing forwarding of packets that has SSRC fields that has - previously come from other end-points. For cryptographic - verification of the source SRTP will require additional security - mechanisms, like TESLA for SRTP [RFC4383]. + Providing source authentication in multi-party scenarios is a + challenge. In the mixer-based topologies, end-points source + authentication is based on, firstly, verifying that media comes from + the mixer by cryptographic verification and, secondly, trust in the + mixer to correctly identify any source towards the end-point. In RTP + sessions where multiple end-points are directly visible to an end- + point, all end-points will have knowledge about each others' master + keys, and can thus inject packets claimed to come from another end- + point in the session. Any node performing relay can perform non- + cryptographic mitigation by preventing forwarding of packets that + have SSRC fields that came from other end-points before. For + cryptographic verification of the source SRTP would require + additional security mechanisms, like TESLA for SRTP [RFC4383]. 12.8. Simulcast This section discusses simulcast in the meaning of providing a node, - for example a Mixer, with multiple different encoded version of the - same media source. In the WebRTC context that appears to be most - easily accomplished by establishing mutliple PeerConnection all being - feed the same set of MediaStreams. Each PeerConnection is then - configured to deliver a particular media quality and thus media bit- - rate. This will work well as long as the end-point implements - indepdentent media encoding for each PeerConnection and not share the - encoder. Simulcast will fail if the end-point uses a common encoder - instance to multiple PeerConnections. + for example a Mixer, with multiple different encoded versions of the + same media source. In the WebRTC context, this could be accomplished + in two ways. One is to establish multiple PeerConnection all being + feed the same set of WebRTC MediaStreams. Another method is to use + multiple WebRTC MediaStreams that are differently configured when it + comes to the media parameters. This would result in that multiple + different RTP Media Streams (SSRCs) being in used with different + encoding based on the same media source (camera, microphone). - Thus it should be considered to explicitly signal which of the two - implementation strategies that are desired and which will be done. - At least making the application and possible the central node - interested in receiving simulcast of an end-points media streams to - be aware if it will function or not. + When intending to use simulcast it is important that this is made + explicit so that the end-points don't automatically try to optimize + away the different encodings and provide a single common version. + Thus, some explicit indications that the intent really is to have + different media encodings is likely required. It should be noted + that it might be a central node, rather than an WebRTC end-point that + would benefit from receiving simulcasted media sources. + + tbd: How to perform simulcast needs to be determined and the + appropriate API or signalling for its usage needs to be defined. 12.9. Differentiated Treatment of Flows - There exist use cases for differentiated treatment of media streams. - Such differentiation can happen at several places in the system. - First of all is the prioritization within the end-point for which - media streams that should be sent, there allocation of bit-rate out - of the current available aggregate as determined by the congestion - control. + 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. - Secondly, the transport can prioritize a media streams. This is done - according to three methods; + 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 + which should be treated differently, the second the actual mechanism + to prioritize packets. This is done according to three methods; - Diffserv: The end-point could mark the packet with a diffserv code - point to indicate to the network how the WebRTC application and - browser would like this particular packet treated. + Diffserv: The end-point marks a packet with a diffserv code point to + indicate to the network that the packet belongs to a particular + class. - Flow based: Prioritization of all packets belonging to a particular - media flow or RTP session by keeping them in separated UDP flows. - Thus enabling either end-point initiated or network initiated - prioritization of the flow. + Flow based: Packets that shall be given a particular treatment are + identified using a combination of IP and port address. Deep Packet Inspection: A network classifier (DPI) inspects the packet and tries to determine if the packet represents a particular application and type that is to be prioritized. With the exception of diffserv both flow based and DPI have issues with running multiple media types and flows on a single UDP flow, especially when combined with data transport (SCTP/DTLS). DPI has - issues due to that multiple different type of flows are aggregated - and thus becomes more difficult to apply analysis on. The flow based + issues because multiple types of flows are aggregated and thus it + becomes more difficult to analyse them. The flow-based differentiation will provide the same treatment to all packets within - the flow. Thus relative prioritization is not possible. In addition + the flow, i.e., relative prioritization is not possible. Moreover, if the resources are limited it may 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 so that it can provide the separation of the - media streams onto different UDP flows to enable a more granular + 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. - Diffserv is based on that either the end-point or a classifier can - mark the packets with an appropriate DSCP so the packets is treated + 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 - there exist two requirements in the WebRTC context. The first is - that the WebRTC application or browser knows which DSCP to use and - that it can use them on some set of media streams. Secondly the - information needs to be propagated to the operating system when - transmitting the packet. + 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. - Open Issue: How will the WebRTC application and/or browser know that - differentiated treatment is desired and available and ensure that it - gets the information required to correctly configure the WebRTC - multimedia conference. + tbd: The model for providing differentiated treatment needs to be + evolved. This includes: + + 1. How the application can prioritize MediaStreamTracks differently + in the API + + 2. How the browser or application determine availability of + transport differentiation + + 3. How to learn about any configuration information for transport + differentiation, such as DSCPs. 13. IANA Considerations This memo makes no request of IANA. Note to RFC Editor: this section may be removed on publication as an RFC. 14. Security Considerations RTP and its various extensions each have their own security considerations. These should be taken into account when considering the security properties of the complete suite. We currently don't think this suite creates any additional security issues or properties. The use of SRTP [RFC3711] will provide protection or - mitigation against all the fundamental issues by offering + mitigation against most of the fundamental issues by offering confidentiality, integrity and partial source authentication. A mandatory to implement media security solution will be required to be picked. We currently don't discuss the key-management aspect of SRTP in this memo, that needs to be done taking the WebRTC communication model into account. - The guidelines in [I-D.ietf-avtcore-srtp-vbr-audio] apply when using - variable bit rate (VBR) audio codecs, for example Opus or the Mixer - audio level header extensions. + Privacy concerns are under discussion and the generation of non- + trackable CNAMEs are under discussion. + + The guidelines in [RFC6562] apply when using variable bit rate (VBR) + audio codecs, for example Opus or the Mixer audio level header + extensions. Security considerations for the WebRTC work are discussed in [I-D.ietf-rtcweb-security]. 15. Acknowledgements The authors would like to thank Harald Alvestrand, Cary Bran, Charles Eckel and Cullen Jennings for valuable feedback. 16. References @@ -1484,60 +1468,48 @@ Using Session Description Protocol (SDP) Port Numbers", draft-holmberg-mmusic-sdp-bundle-negotiation-00 (work in progress), October 2011. [I-D.ietf-avtcore-srtp-encrypted-header-ext] Lennox, J., "Encryption of Header Extensions in the Secure Real-Time Transport Protocol (SRTP)", draft-ietf-avtcore-srtp-encrypted-header-ext-01 (work in progress), October 2011. - [I-D.ietf-avtcore-srtp-vbr-audio] - Perkins, C. and J. Valin, "Guidelines for the use of - Variable Bit Rate Audio with Secure RTP", - draft-ietf-avtcore-srtp-vbr-audio-04 (work in progress), - December 2011. + [I-D.ietf-avtext-multiple-clock-rates] + Petit-Huguenin, M. and G. Zorn, "Support for Multiple + Clock Rates in an RTP Session", + draft-ietf-avtext-multiple-clock-rates-05 (work in + progress), May 2012. [I-D.ietf-rtcweb-overview] Alvestrand, H., "Overview: Real Time Protocols for Brower- - based Applications", draft-ietf-rtcweb-overview-03 (work - in progress), March 2012. + based Applications", draft-ietf-rtcweb-overview-04 (work + in progress), June 2012. [I-D.ietf-rtcweb-security] Rescorla, E., "Security Considerations for RTC-Web", - draft-ietf-rtcweb-security-02 (work in progress), - March 2012. - - [I-D.jesup-rtp-congestion-reqs] - Jesup, R. and H. Alvestrand, "Congestion Control - Requirements For Real Time Media", - draft-jesup-rtp-congestion-reqs-00 (work in progress), - March 2012. + draft-ietf-rtcweb-security-03 (work in progress), + June 2012. [I-D.lennox-rtcweb-rtp-media-type-mux] Rosenberg, J. and J. Lennox, "Multiplexing Multiple Media Types In a Single Real-Time Transport Protocol (RTP) Session", draft-lennox-rtcweb-rtp-media-type-mux-00 (work in progress), October 2011. [I-D.perkins-avtcore-rtp-circuit-breakers] Perkins, C. and V. Singh, "RTP Congestion Control: Circuit Breakers for Unicast Sessions", draft-perkins-avtcore-rtp-circuit-breakers-00 (work in progress), March 2012. - [I-D.westerlund-avtcore-multiplex-architecture] - Westerlund, M., Burman, B., and C. Perkins, "RTP - Multiplexing Architecture", - draft-westerlund-avtcore-multiplex-architecture-01 (work - in progress), March 2012. - [I-D.westerlund-avtcore-transport-multiplexing] Westerlund, M. and C. Perkins, "Multiple RTP Sessions on a Single Lower-Layer Transport", draft-westerlund-avtcore-transport-multiplexing-02 (work in progress), March 2012. [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997. [RFC2736] Handley, M. and C. Perkins, "Guidelines for Writers of RTP @@ -1569,23 +1541,20 @@ Hakenberg, "RTP Retransmission Payload Format", RFC 4588, July 2006. [RFC4961] Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)", BCP 131, RFC 4961, July 2007. [RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman, "Codec Control Messages in the RTP Audio-Visual Profile with Feedback (AVPF)", RFC 5104, February 2008. - [RFC5109] Li, A., "RTP Payload Format for Generic Forward Error - Correction", RFC 5109, December 2007. - [RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF)", RFC 5124, February 2008. [RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP Header Extensions", RFC 5285, July 2008. [RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size Real-Time Transport Control Protocol (RTCP): Opportunities and Consequences", RFC 5506, April 2009. @@ -1605,88 +1574,80 @@ (CNAMEs)", RFC 6222, April 2011. [RFC6464] Lennox, J., Ivov, E., and E. Marocco, "A Real-time Transport Protocol (RTP) Header Extension for Client-to- Mixer Audio Level Indication", RFC 6464, December 2011. [RFC6465] Ivov, E., Marocco, E., and J. Lennox, "A Real-time Transport Protocol (RTP) Header Extension for Mixer-to- Client Audio Level Indication", RFC 6465, December 2011. + [RFC6562] Perkins, C. and JM. Valin, "Guidelines for the Use of + Variable Bit Rate Audio with Secure RTP", RFC 6562, + March 2012. + 16.2. Informative References [I-D.alvestrand-rtcweb-msid] Alvestrand, H., "Cross Session Stream Identification in the Session Description Protocol", draft-alvestrand-rtcweb-msid-02 (work in progress), May 2012. - [I-D.begen-mmusic-redundancy-grouping] - Begen, A., Cai, Y., and H. Ou, "Duplication Grouping - Semantics in the Session Description Protocol", - draft-begen-mmusic-redundancy-grouping-03 (work in - progress), March 2012. - - [I-D.cbran-rtcweb-data] - Bran, C. and C. Jennings, "RTC-Web Non-Media Data - Transport Requirements", draft-cbran-rtcweb-data-00 (work - in progress), July 2011. - [I-D.ietf-avt-srtp-ekt] Wing, D., McGrew, D., and K. Fischer, "Encrypted Key Transport for Secure RTP", draft-ietf-avt-srtp-ekt-03 (work in progress), October 2011. - [I-D.ietf-fecframe-framework] - Watson, M., Begen, A., and V. Roca, "Forward Error - Correction (FEC) Framework", - draft-ietf-fecframe-framework-15 (work in progress), - June 2011. + [I-D.ietf-rtcweb-use-cases-and-requirements] + Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- + Time Communication Use-cases and Requirements", + draft-ietf-rtcweb-use-cases-and-requirements-09 (work in + progress), June 2012. - [RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V., - Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse- - Parisis, "RTP Payload for Redundant Audio Data", RFC 2198, - September 1997. + [I-D.jesup-rtp-congestion-reqs] + Jesup, R. and H. Alvestrand, "Congestion Control + Requirements For Real Time Media", + draft-jesup-rtp-congestion-reqs-00 (work in progress), + March 2012. + + [I-D.westerlund-avtcore-multiplex-architecture] + Westerlund, M., Burman, B., and C. Perkins, "RTP + Multiplexing Architecture", + draft-westerlund-avtcore-multiplex-architecture-01 (work + in progress), March 2012. [RFC4341] Floyd, S. and E. Kohler, "Profile for Datagram Congestion Control Protocol (DCCP) Congestion Control ID 2: TCP-like Congestion Control", RFC 4341, March 2006. [RFC4342] Floyd, S., Kohler, E., and J. Padhye, "Profile for Datagram Congestion Control Protocol (DCCP) Congestion Control ID 3: TCP-Friendly Rate Control (TFRC)", RFC 4342, March 2006. [RFC4383] Baugher, M. and E. Carrara, "The Use of Timed Efficient Stream Loss-Tolerant Authentication (TESLA) in the Secure Real-time Transport Protocol (SRTP)", RFC 4383, February 2006. [RFC4828] Floyd, S. and E. Kohler, "TCP Friendly Rate Control (TFRC): The Small-Packet (SP) Variant", RFC 4828, April 2007. - [RFC4867] Sjoberg, J., Westerlund, M., Lakaniemi, A., and Q. Xie, - "RTP Payload Format and File Storage Format for the - Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband - (AMR-WB) Audio Codecs", RFC 4867, April 2007. - [RFC5117] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117, January 2008. [RFC5348] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP Friendly Rate Control (TFRC): Protocol Specification", RFC 5348, September 2008. - [RFC5404] Westerlund, M. and I. Johansson, "RTP Payload Format for - G.719", RFC 5404, January 2009. - [RFC5576] Lennox, J., Ott, J., and T. Schierl, "Source-Specific Media Attributes in the Session Description Protocol (SDP)", RFC 5576, June 2009. [RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion Control", RFC 5681, September 2009. [RFC5968] Ott, J. and C. Perkins, "Guidelines for Extending the RTP Control Protocol (RTCP)", RFC 5968, September 2010. @@ -1698,31 +1659,36 @@ RTP supports both unicast and group communication, with participants being connected using wide range of transport-layer topologies. Some of these topologies involve only the end-points, while others use RTP translators and mixers to provide in-network processing. Properties of some RTP topologies are discussed in [RFC5117], and we further describe those expected to be useful for WebRTC in the following. We also goes into important RTP session aspects that the topology or implementation variant can place on a WebRTC end-point. + This section includes RTP topologies beyond the recommended ones. + This in an attempt to highlight the differencies and the in many case + small differences in implementation to support a larger set of + possible topologies. + A.1. Point to Point - The point-to-point RTP topology (Figure 5) is the simplest scenario + The point-to-point RTP topology (Figure 3) is the simplest scenario for WebRTC applications. This is going to be very common for user to user calls. +---+ +---+ | A |<------->| B | +---+ +---+ - Figure 5: Point to Point + Figure 3: Point to Point This being the basic one lets use the topology to high-light a couple of details that are common for all RTP usage in the WebRTC context. First is the intention to multiplex RTP and RTCP over the same UDP- flow. Secondly is the question of using only a single RTP session or one per media type for legacy interoperability. Thirdly is the question of using multiple sender sources (SSRCs) per end-point. Historically, RTP and RTCP have been run on separate UDP ports. With the increased use of Network Address/Port Translation (NAPT) this has @@ -1739,21 +1705,21 @@ (e.g., audio and video), then each type media can be sent as a separate RTP session using a different 5-tuple, allowing for separate transport level treatment of each type of media. Alternatively, all types of media can be multiplexed onto a single 5-tuple as a single RTP session, or as several RTP sessions if using a demultiplexing shim. Multiplexing different types of media onto a single 5-tuple places some limitations on how RTP is used, as described in "RTP Multiplexing Architecture" [I-D.westerlund-avtcore-multiplex-architecture]. It is not expected that these limitations will significantly affect the scenarios - targetted by WebRTC, but they may impact interoperability with legacy + targeted by WebRTC, but they may impact interoperability with legacy systems. An RTP session have good support for simultanously transport multiple media sources. Each media source uses an unique SSRC identifier and each SSRC has independent RTP sequence number and timestamp spaces. This is being utilized in WebRTC for several cases. One is to enable multiple media sources of the same type, an end-point that has two video cameras can potentially transmitt video from both to its peer(s). Another usage is when a single RTP session is being used for both multiple media types, thus an end-point can transmit both @@ -1789,24 +1755,24 @@ | | | | +-Video-| |-Video-+ | | | | | | | | | AV1|---------------->| | | | | | | | | | | AV2|---------------->| | | | | | | | | | | |<----------------|BV1 | | | | | | | | | +-------| |-------+ | | | | | | | +---------| |---------+ | | | | | +-----------| |-----------+ | | | +-------------| |-------------+ | +---------------+ +---------------+ - Figure 6: Point to Point: Multiple RTP sessions + Figure 4: Point to Point: Multiple RTP sessions As can be seen above in the Point to Point: Multiple RTP sessions - (Figure 6) the single Peer Connection contains two RTP sessions over + (Figure 4) the single Peer Connection contains two RTP sessions over different UDP flows UDP 1 and UDP 2, i.e. their 5-tuples will be different, normally on source and destination ports. The first RTP session (RTP1) carries audio, one stream in each direction AA1 and BA1. The second RTP session contains two video streams from A (AV1 and AV2) and one from B to A (BV1). +-A-------------+ +-B-------------+ | +-PeerC1------| |-PeerC1------+ | | | +-UDP1------| |-UDP1------+ | | | | | +-RTP1----| |-RTP1----+ | | | @@ -1818,52 +1784,52 @@ | | | | +-Video-| |-Video-+ | | | | | | | | | AV1|---------------->| | | | | | | | | | | AV2|---------------->| | | | | | | | | | | |<----------------|BV1 | | | | | | | | | +-------| |-------+ | | | | | | | +---------| |---------+ | | | | | +-----------| |-----------+ | | | +-------------| |-------------+ | +---------------+ +---------------+ - Figure 7: Point to Point: Single RTP session. + Figure 5: Point to Point: Single RTP session. - In (Figure 7) there is only a single UDP flow and RTP session (RTP1). - This RTP session carries a total of five (5) media streams (SSRCs). - From A to B there is Audio (AA1) and two video (AV1 and AV2). From B - to A there is Audio (BA1) and Video (BV1). + In (Figure 5) there is only a single UDP flow and RTP session (RTP1). + This RTP session carries a total of five (5) RTP media streams + (SSRCs). From A to B there is Audio (AA1) and two video (AV1 and + AV2). From B to A there is Audio (BA1) and Video (BV1). A.2. Multi-Unicast (Mesh) For small multiparty calls, it is practical to set up a multi-unicast - topology (Figure 8); unfortunately not discussed in the RTP + topology (Figure 6); unfortunately not discussed in the RTP Topologies RFC [RFC5117]. In this topology, each participant sends individual unicast RTP/UDP/IP flows to each of the other participants using independent PeerConnections in a full mesh. +---+ +---+ | A |<---->| B | +---+ +---+ ^ ^ \ / \ / v v +---+ | C | +---+ - Figure 8: Multi-unicast + Figure 6: Multi-unicast This topology has the benefit of not requiring central nodes. The downside is that it increases the used bandwidth at each sender by - requiring one copy of the media streams for each participant that are - part of the same session beyond the sender itself. Hence, this + requiring one copy of the RTP media streams for each participant that + are part of the same session beyond the sender itself. Hence, this topology is limited to scenarios with few participants unless the media is very low bandwidth. The multi-unicast topology could be implemented as a single RTP session, spanning multiple peer-to-peer transport layer connections, or as several pairwise RTP sessions, one between each pair of peers. To maintain a coherent mapping between the relation between RTP sessions and PeerConnections we recommend that one implements this as individual RTP sessions. The only downside is that end-point A will not learn of the quality of any transmission happening between B and C based on RTCP. This has not been seen as a significant downside as now one has yet seen a need @@ -1892,38 +1858,38 @@ | | | | +-RTP2----| |-RTP2----+ | | | | | +----+ | | | +-Audio-| |-Audio-+ | | | | | +->|ENC2|--+-+-+-+--->AA2|------------->| | | | | | | +----+ | | | | |<-------------|CA1 | | | | | | | | | +-------| |-------+ | | | | | | | +---------| |---------+ | | | | | +-----------| |-----------+ | | | +-------------| |-------------+ | +--------------------------+ +---------------+ - Figure 9: Session strcuture for Multi-Unicast Setup + Figure 7: Session structure for Multi-Unicast Setup Lets review how the RTP sessions looks from A's perspective by considering both how the media is a handled and what PeerConnections - and RTP sessions that are setup in Figure 9. A's microphone is + and RTP sessions that are setup in Figure 7. A's microphone is captured and the digital audio can then be feed into two different encoder instances each beeing associated with two different PeerConnections (PeerC1 and PeerC2) each containing independent RTP sessions (RTP1 and RTP2). The SSRCs in each RTP session will be completely independent and the media bit-rate produced by the encoder can also be tuned to address any congestion control requirements between A and B differently then for the path A to C. For media encodings which are more resource consuming, like video, one could expect that it will be common that end-points that are resource costrained will use a different implementation strategy where the encoder is shared between the different PeerConnections as - shown below Figure 10. + shown below Figure 8. +-A----------------------+ +-B-------------+ |+---+ | | | ||CAM| +-PeerC1------| |-PeerC1------+ | |+---+ | +-UDP1------| |-UDP1------+ | | | | | | +-RTP1----| |-RTP1----+ | | | | V | | | +-Video-| |-Video-+ | | | | |+----+ | | | | |<----------------|BV1 | | | | | ||ENC |----+-+-+-+--->AV1|---------------->| | | | | | |+----+ | | | +-------| |-------+ | | | | | | | | +---------| |---------+ | | | @@ -1937,79 +1903,79 @@ | | | | +-RTP2----| |-RTP2----+ | | | | | | | | +-Video-| |-Video-+ | | | | | +-------+-+-+-+--->AV2|---------------->| | | | | | | | | | | |<----------------|CV1 | | | | | | | | | +-------| |-------+ | | | | | | | +---------| |---------+ | | | | | +-----------| |-----------+ | | | +-------------| |-------------+ | +------------------------+ +---------------+ - Figure 10: Single Encoder Multi-Unicast Setup + Figure 8: Single Encoder Multi-Unicast Setup This will clearly save resources consumed by encoding but does introduce the need for the end-point A to make decisions on how it encodes the media so it suites delivery to both B and C. This is not limited to congestion control, also prefered resolution to receive based on dispaly area available is another aspect requiring consideration. The need for this type of descion logic does arise in several different topologies and implementation. A.3. Mixer Based - An mixer (Figure 11) is a centralised point that selects or mixes + An mixer (Figure 9) is a centralised point that selects or mixes content in a conference to optimise the RTP session so that each end- point only needs connect to one entity, the mixer. The mixer can also reduce the bit-rate needed from the mixer down to a conference participants as 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 in - stead of 3 in the depicted scenario (Figure 11). + stead of 3 in the depicted scenario (Figure 9). +---+ +------------+ +---+ | A |<---->| |<---->| B | +---+ | | +---+ | Mixer | +---+ | | +---+ | C |<---->| |<---->| D | +---+ +------------+ +---+ - Figure 11: RTP Mixer with Only Unicast Paths + Figure 9: RTP Mixer with Only Unicast Paths Mixers has two downsides, the first is that the mixer must be a trusted node as they either performs media operations or at least repacketize the media. Both type of operations requires when using SRTP that the mixer verifies integrity, decrypts the content, perform its operation and form new RTP packets, encrypts and integegrity protect them. This applies to all types of mixers described below. The second downside is that all these operations and optimization of the session requires processing. How much depends on the implementation as will become evident below. The implementation of an 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 on a particular media stream. + or codec used on a particular RTP media stream. A.3.1. Media Mixing This type of mixer is one which clearly can be called RTP mixer is likely the one that most thinks of when they hear the term mixer. Its basic patter of operation is that it will receive the different - participants media stream. Select which that are to be included in a - media domain mix of the incomming media streams. Then create a - single outgoing stream from this mix. + participants RTP media stream. Select which that are to be included + in a media domain mix of the incomming RTP media streams. Then + create a single outgoing stream from this mix. Audio mixing is straight forward and commonly possible to do for a number of participants. Lets assume that you want to mix N number of streams from different participants. Then the mixer need to perform N decodings. Then it needs to produce N or N+1 mixes, the reasons that different mixes are needed are so that each contributing source get a mix which don't contain themselves, as this would result in an echo. When N is lower than the number of all participants one may produce a Mix of all N streams for the group that are curently not included in the mix, thus N+1 mixes. These audio streams are then @@ -2020,21 +1986,21 @@ video streams can be done. In fact it can be done in a number of ways, tiling the different streams creating a chessboard, selecting someone as more important and showing them large and a number of other sources as smaller is another. Also here one commonly need to produce a number of different compositions so that the contributing part doesn't need to see themselves. Then the mixer re-encodes the created video stream, RTP packetize it and send it out The problem with media mixing is that it both consume large amount of media processing and encoding resources. The second is the quality - degradation created by decoding and re-encoding the media stream. + degradation created by decoding and re-encoding the RTP media stream. Its advantage is that it is quite simplistic for the clients to handle as they don't need to handle local mixing and composition. +-A-------------+ +-MIXER--------------------------+ | +-PeerC1------| |-PeerC1--------+ | | | +-UDP1------| |-UDP1--------+ | | | | | +-RTP1----| |-RTP1------+ | | +-----+ | | | | | +-Audio-| |-Audio---+ | | | +---+ | | | | | | | | AA1|------------>|---------+-+-+-+-|DEC|->| | | | | | | | |<------------|MA1 <----+ | | | +---+ | | | @@ -2065,24 +2031,24 @@ | | | | +-Audio-| |-Audio---+ | | | +---+ | | | | | | | | CA1|------------>|---------+-+-+-+-|DEC|->| | | | | | | | |<------------|MA3 <----+ | | | +---+ | | | | | | | +-------| |(BA1+CA1)|\| | | +---+ | | | | | | +---------| |---------+ +-+-+-|ENC|<-| A+B | | | | +-----------| |-----------+ | | +---+ | | | | +-------------| |-------------+ | +-----+ | +---------------+ |---------------+ | +--------------------------------+ - Figure 12: Session and SSRC details for Media Mixer + Figure 10: Session and SSRC details for Media Mixer From an RTP perspective media mixing can be very straight forward as - can be seen in Figure 12. The mixer present one SSRC towards the + can be seen in Figure 10. The mixer present one SSRC towards the peer client, e.g. MA1 to Peer A, which is the media mix of the other particpants. As each peer receives a different version produced by the mixer there are no actual relation between the different RTP sessions in the actual media or the transport level information. There is however one connection between RTP1-RTP3 in this figure. It has to do with the SSRC space and the identity information. When A receives the MA1 stream which is a combination of BA1 and CA1 streams in the other PeerConnections RTP could enable the mixer to include CSRC information in the MA1 stream to identify the contributing source BA1 and CA1. @@ -2096,28 +2062,28 @@ the different legs. For the above situation commonly nothing more than the Source Description (SDES) information and RTCP BYE for CSRC need to be exposed. The main goal would be to enable the correct binding against the application logic and other information sources. This also enables loop detection in the RTP session. A.3.1.1. RTP Session Termination There exist an possible implementation choice to have the RTP sessions being separated between the different legs in the multi- - party communication session and only generate media streams in each - without carrying on RTP/RTCP level any identity information about the - contributing sources. This removes both the functionaltiy that CSRC - can provide and the possibility to use any extensions that build on - CSRC and the loop detection. It may appear a simplification if SSRC - collision would occur between two different end-points as they can be - avoide to be resolved and instead remapped between the independent - sessions if at all exposed. However, SSRC/CSRC remapping + party communication session and only generate RTP media streams in + each without carrying on RTP/RTCP level any identity information + about the contributing sources. This removes both the functionaltiy + that CSRC can provide and the possibility to use any extensions that + build on CSRC and the loop detection. It may appear a simplification + if SSRC collision would occur between two different end-points as + they can be avoide to be resolved and instead remapped between the + independent sessions if at all exposed. However, SSRC/CSRC remapping requiresthat SSRC/CSRC are never exposed to the WebRTC javascript client to use as reference. This as they only have local importance if they are used on a multi-party session scope the result would be missreferencing. Also SSRC collision handling will still be needed as it may occur between the mixer and the end-point. Session termination may appear to resolve some issues, it however creates other issues that needs resolving, like loop detection, identification of contributing sources and the need to handle mapped identities and ensure that the right one is used towards the right @@ -2125,24 +2091,24 @@ A.3.2. Media Switching An RTP Mixer based on media switching avoids the media decoding and encoding cycle in the mixer, but not the decryption and re-encryption cycle as one rewrites RTP headers. This both reduces the amount of computational resources needed in the mixer and increases the media quality per transmitted bit. This is achieve by letting the mixer have a number of SSRCs that represents conceptual or functional streams the mixer produces. These streams are created by selecting - media from one of the by the mixer received media streams and forward - the media using the mixers own SSRCs. The mixer can then switch - between available sources if that is required by the concept for the - source, like currently active speaker. + media from one of the by the mixer received RTP media streams and + forward the media using the mixers own SSRCs. The mixer can then + switch between available sources if that is required by the concept + for the source, like currently active speaker. To achieve a coherent RTP media stream from the mixer's SSRC the mixer is forced to rewrite the incoming RTP packet's header. First the SSRC field must be set to the value of the Mixer's SSRC. Secondly, the sequence number must be the next in the sequence of outgoing packets it sent. Thirdly the RTP timestamp value needs to be adjusted using an offset that changes each time one switch media source. Finally depending on the negotiation the RTP payload type value representing this particular RTP payload configuration may have to be changed if the different PeerConnections have not arrived on @@ -2192,34 +2158,34 @@ | | | | +-Video-| |-Video---+ | | | | | | | | | | | CV1|------------>|---------+-+-+-+------->| | | | | | | | |<------------|MV11 <---+-+-+-+-AV1----| | | | | | | | |<------------|MV12 <---+-+-+-+-EV1----| | | | | | | +-------| |---------+ | | | | | | | | | +---------| |-----------+ | | | | | | | +-----------| |-------------+ | +-----+ | | +-------------| |---------------+ | +---------------+ +--------------------------------+ - Figure 13: Media Switching RTP Mixer + Figure 11: Media Switching RTP Mixer The Media Switching RTP mixer can similar to the Media Mixing one reduce the bit-rate needed towards the different peers by selecting - and switching in a sub-set of media streams out of the ones it + and switching in a sub-set of RTP media streams out of the ones it receives from the conference participations. - To ensure that a media receiver can correctly decode the media stream - after a switch, it becomes necessary to ensure for state saving - codecs that they start from default state at the point of switching. - Thus one common tool for video is to request that the encoding - creates an intra picture, something that isn't dependent on earlier - state. This can be done using Full Intra Request RTCP codec control - message as discussed in Section 5.1.1. + To ensure that a media receiver can correctly decode the RTP media + stream after a switch, it becomes necessary to ensure for state + saving codecs that they start from default state at the point of + switching. Thus one common tool for video is to request that the + encoding creates an intra picture, something that isn't dependent on + earlier state. This can be done using Full Intra Request RTCP codec + control message as discussed in Section 5.1.1. Also in this type of mixer one could consider to terminate the RTP sessions fully between the different PeerConnection. The same arguments and conisderations as discussed in Appendix A.3.1.1 applies here. A.3.3. Media Projecting Another method for handling media in the RTP mixer is to project all potential sources (SSRCs) into a per end-point independent RTP @@ -2270,107 +2236,108 @@ | | | | +-Video-| |-Video---+ | | | | | | | | | | | CV1|------------>|---------+-+-+-+------->| | | | | | | | |<------------|AV1 <----+-+-+-+--------| | | | | | | | | : : : |: : : : : : : : : : :| | | | | | | | |<------------|EV1 <----+-+-+-+--------| | | | | | | +-------| |---------+ | | | | | | | | | +---------| |-----------+ | | | | | | | +-----------| |-------------+ | +-----+ | | +-------------| |---------------+ | +---------------+ +--------------------------------+ - Figure 14: Media Projecting Mixer + Figure 12: Media Projecting Mixer - So in this six participant conference depicted above in (Figure 14) + So in this six participant conference depicted above in (Figure 12) one can see that end-point A will in this case be aware of 5 incoming SSRCs, BV1-FV1. If this mixer intend to have the same behavior as in Appendix A.3.2 where the mixer provides the end-points with the two latest speaking end-points, then only two out of these five SSRCs will concurrently transmitt media to A. As the mixer selects which source in the different RTP sessions that transmit media to the end- - points each media stream will require some rewriting when being + points each RTP media stream will require some rewriting when being projected from one session into another. The main thing is that the sequence number will need to be consequitvely incremented based on the packet actually being transmitted in each RTP session. Thus the RTP sequence number offset will change each time a source is turned on in RTP session. As the RTP sessions are independent the SSRC numbers used can be handled indepdentently also thus working around any SSRC collisions by having remapping tables between the RTP sessions. However the - related MediaStream signalling must be correspondlingly changed to - ensure consistent MediaStream to SSRC mappings between the different - PeerConnections and the same comment that higher functions must not - use SSRC as references to media streams applies also here. + related WebRTC MediaStream signalling must be correspondlingly + changed to ensure consistent WebRTC MediaStream to SSRC mappings + between the different PeerConnections and the same comment that + higher functions must not use SSRC as references to RTP media streams + applies also here. The mixer will also be responsible to act on any RTCP codec control requests comming from an end-point and decide if it can act on it locally or needs to translate the request into the RTP session that contains the media source. Both end-points and the mixer will need to implement conference related codec control functionalities to provide a good experience. Full Intra Request to request from the media source to provide switching points between the sources, Temporary Maximum Media Bit-rate Request (TMMBR) to enable the mixer to aggregate congestion control response towards the media source and have it adjust its bit-rate in case the limitation is not in the source to mixer link. This version of the mixer also puts different requirements on the - end-point when it comes to decoder instances and handling of the + end-point when it comes to decoder instances and handling of the RTP media streams providing media. As each projected SSRC can at any time provide media the end-point either needs to handle having thus many allocated decoder instances or have efficient switching of decoder contexts in a more limited set of actual decoder instances to cope with the switches. The WebRTC application also gets more responsibility to update how the media provides is to be presented to the user. A.4. Translator Based There is also a variety of translators. The core commonality is that they do not need to make themselves visible in the RTP level by having an SSRC themselves. Instead they sit between one or more end- point and perform translation at some level. It can be media transcoding, protocol translation or covering missing functionality - for a legacy device or simply relay packets between transport domains - or to realize multi-party. We will go in details below. + for a legacy end-point or simply relay packets between transport + domains or to realize multi-party. We will go in details below. A.4.1. Transcoder A transcoder operates on media level and really used for two purposes, the first is to allow two end-points that doesn't have a common set of media codecs to communicate by translating from one codec to another. The second is to change the bit-rate to a lower one. For WebRTC end-points communicating with each other only the first one should at all be relevant. In certain legacy deployment media transcoder will be necessary to ensure both codecs and bit-rate - falls within the envelope the legacy device supports. + falls within the envelope the legacy end-point supports. As transcoding requires access to the media the transcoder must within the security context and access any media encryption and integrity keys. On the RTP plane a media transcoder will in practice fork the RTP session into two different domains that are highly decoupled when it comes to media parameters and reporting, but not identities. To maintain signalling bindings to SSRCs a transcoder is likely needing to use the SSRC of one end-point to represent the - transcoded media stream to the other end-point(s). The congestion - control loop can be terminated in the transcoder as the media bit- - rate being sent by the transcoder can be adjusted independently of - the incoming bit-rate. However, for optimizing performance and - resource consumption the translator needs to consider what signals or - bit-rate reductions it should send towards the source end-point. For - example receving a 2.5 mbps video stream and then send out a 250 kbps - video stream after transcoding is a vaste of resources. In most - cases a 500 kbps video stream from the source in the right resolution - is likely to provide equal quality after transcoding as the 2.5 mbps - source stream. At the same time increasing media bit-rate futher - than what is needed to represent the incoming quality accurate is - also wasted resources. + transcoded RTP media stream to the other end-point(s). The + congestion control loop can be terminated in the transcoder as the + media bit-rate being sent by the transcoder can be adjusted + independently of the incoming bit-rate. However, for optimizing + performance and resource consumption the translator needs to consider + what signals or bit-rate reductions it should send towards the source + end-point. For example receving a 2.5 mbps video stream and then + send out a 250 kbps video stream after transcoding is a vaste of + resources. In most cases a 500 kbps video stream from the source in + the right resolution is likely to provide equal quality after + transcoding as the 2.5 mbps source stream. At the same time + increasing media bit-rate futher than what is needed to represent the + incoming quality accurate is also wasted resources. +-A-------------+ +-Translator------------------+ | +-PeerC1------| |-PeerC1--------+ | | | +-UDP1------| |-UDP1--------+ | | | | | +-RTP1----| |-RTP1------+ | | | | | | | +-Audio-| |-Audio---+ | | | +---+ | | | | | | AA1|------------>|---------+-+-+-+-|DEC|----+ | | | | | | |<------------|BA1 <----+ | | | +---+ | | | | | | | | | |\| | | +---+ | | | | | | +-------| |---------+ +-+-+-|ENC|<-+ | | @@ -2386,50 +2353,50 @@ | | | | +-Audio-| |-Audio---+ | | | +---+ | | | | | | | | BA1|------------>|---------+-+-+-+-|DEC|--+ | | | | | | | |<------------|AA1 <----+ | | | +---+ | | | | | | | | | |\| | | +---+ | | | | | | +-------| |---------+ +-+-+-|ENC|<---+ | | | | +---------| |-----------+ | | +---+ | | | +-----------| |-------------+ | | | +-------------| |---------------+ | +---------------+ +-----------------------------+ - Figure 15: Media Transcoder + Figure 13: Media Transcoder - Figure 15 exposes some important details. First of all you can see + Figure 13 exposes some important details. First of all you can see the SSRC identifiers used by the translator are the corresponding end-points. Secondly, there is a relation between the RTP sessions in the two different PeerConnections that are represtented by having both parts be identified by the same level and they need to share certain contexts. Also certain type of RTCP messages will need to be bridged between the two parts. Certain RTCP feedback messages are likely needed to be soruced by the translator in response to actions by the translator and its media encoder. A.4.2. Gateway / Protocol Translator Gateways are used when some protocol feature that is required is not supported by an end-point wants to participate in session. This RTP - translator in Figure 16 takes on the role of ensuring that from the + translator in Figure 14 takes on the role of ensuring that from the perspective of participant A, participant B appears as a fully compliant WebRTC end-point (that is, it is the combination of the Translator and participant B that looks like a WebRTC end point). +------------+ | | +---+ | Translator | +---+ | A |<---->| to legacy |<---->| B | +---+ | end-point | +---+ WebRTC | | Legacy +------------+ - Figure 16: Gateway (RTP translator) towards legacy end-point + Figure 14: Gateway (RTP translator) towards legacy end-point For WebRTC there are a number of requirements that could force the need for a gateway if a WebRTC end-point is to communicate with a legacy end-point, such as support of ICE and DTLS-SRTP for keymanagement. On RTP level the main functions that may be missing in a legacy implementation that otherswise support RTP are RTCP in general, SRTP implementation, congestion control and feedback messages required to make it work. +-A-------------+ +-Translator------------------+ @@ -2453,25 +2420,25 @@ | | | +-Audio-| |-Audio---+ +---+-+ | | || | | | | |<---RTCP---->|<--------+----------+ | | || | | | | BA1|------------>|---------+--------------+ | || | | | | |<------------|AA1 <----+----------------+ || | | | +-------| |---------+ || | | +---------| |----------------------------+| | +-----------| |-----------+ | | | | | +---------------+ +-----------------------------+ - Figure 17: RTP/RTCP Protocol Translator + Figure 15: RTP/RTCP Protocol Translator The legacy gateway may be implemented in several ways and what it need to change is higly dependent on what functions it need to proxy - for the legacy end-point. One possibility is depicted in Figure 17 + for the legacy end-point. One possibility is depicted in Figure 15 where the RTP media streams are compatible and forward without changes. However, their RTP header values are captured to enable the RTCP translator to create RTCP reception information related to the leg between the end-point and the translator. This can then be combined with the more basic RTCP reports that the legacy endpoint (B) provides to give compatible and expected RTCP reporting to A. Thus enabling at least full congestion control on the path between A and the translator. If B has limited possibilities for congestion response for the media then the translator may need the capabilities to perform media transcoding to address cases where it otherwise @@ -2488,56 +2455,56 @@ encryption and integirty protection operation to resolve missmatch in security systems. A.4.3. Relay There exist a class of translators that operates on transport level below RTP and thus do not effect RTP/RTCP packets directly. They come in two distinct flavors, the one used to bridge between two different transport or address domains to more function as a gateway and the second one which is to to provide a group communication - feature as depicted below in Figure 18. + feature as depicted below in Figure 16. +---+ +------------+ +---+ | A |<---->| |<---->| B | +---+ | | +---+ | Translator | +---+ | | +---+ | C |<---->| |<---->| D | +---+ +------------+ +---+ - Figure 18: RTP Translator (Relay) with Only Unicast Paths + Figure 16: RTP Translator (Relay) with Only Unicast Paths The first kind is straight forward and is likely to exist in WebRTC context when an legacy end-point is compatible with the exception for ICE, and thus needs a gateway that terminates the ICE and then forwards all the RTP/RTCP traffic and keymanagment to the end-point only rewriting the IP/UDP to forward the packet to the legacy node. The second type is useful if one wants a less complex central node or a central node that is outside of the security context and thus do not have access to the media. This relay takes on the role of forwarding the media (RTP and RTCP) packets to the other end-points but doesn't perform any RTP or media processing. Such a device simply forwards the media from each sender to all of the other particpants, and is sometimes called a transport-layer translator. - In Figure 18, participant A will only need to send a media once to + In Figure 16, participant A will only need to send a media once to the relay, which will redistribute it by sending a copy of the stream to participants B, C, and D. Participant A will still receive three RTP streams with the media from B, C and D if they transmit simultaneously. This is from an RTP perspective resulting in an RTP session that behaves equivalent to one transporter over an IP Any Source Multicast (ASM). This results in one common RTP session between all participants despite that there will be independent PeerConnections created to the - translator as depicted below Figure 19. + translator as depicted below Figure 17. +-A-------------+ +-RELAY--------------------------+ | +-PeerC1------| |-PeerC1--------+ | | | +-UDP1------| |-UDP1--------+ | | | | | +-RTP1----| |-RTP1-------------------------+ | | | | | +-Video-| |-Video---+ | | | | | | | AV1|------------>|---------------------------+ | | | | | | | |<------------|BV1 <--------------------+ | | | | | | | | |<------------|CV1 <------------------+ | | | | | | | | +-------| |---------+ | | | | | @@ -2568,32 +2535,32 @@ | | | | +-Video-| |-Video---+ | | | | | | | | | | CV1|------------>|-------------------------+ | | | | | | | | |<------------|AV1 <----------------------+ | | | | | | | |<------------|BV1 <------------------+ | | | | | | +-------| |---------+ | | | | | +---------| |------------------------------+ | | | +-----------| |-------------+ | | | +-------------| |---------------+ | +---------------+ +--------------------------------+ - Figure 19: Transport Multi-party Relay + Figure 17: Transport Multi-party Relay As the Relay RTP and RTCP packets between the UDP flows as indicated by the arrows for the media flow a given WebRTC end-point, like A will see the remote sources BV1 and CV1. There will be also two different network paths between A, and B or C. This results in that the client A must be capable of handlilng that when determining congestion state that there might exist multiple destinations on the far side of a PeerConnection and that these paths shall be treated differently. It also results in a requirement to combine the different congestion states into a decision to transmit a particular - media stream suitable to all participants. + RTP media stream suitable to all participants. It is also important to note that the relay can not perform selective relaying of some sources and not others. The reason is that the RTCP reporting in that case becomes incosistent and without explicit information about it being blocked must be interpret as severe congestion. In this usage it is also necessary that the session management has configured a common set of RTP configuration including RTP payload formats as when A sends a packet with pt=97 it will arrive at both B @@ -2609,123 +2576,124 @@ RTP session. The second problem can basically be solved in two ways. Either a common master key from which all derive their per source key for SRTP. The second alternative which might be more practical is that each end-point has its own key used to protects all RTP/RTCP packets it sends. Each participants key are then distributed to the other participants. This second method could be implemented using DTLS- SRTP to a special key server and then use Encrypted Key Transport [I-D.ietf-avt-srtp-ekt] to distribute the actual used key to the - other participants in the RTP session Figure 20. The first one could + other participants in the RTP session Figure 18. The first one could be achieved using MIKEY messages in SDP. +---+ +---+ | | +-----------+ | | | A |<------->| DTLS-SRTP |<------->| C | | |<-- -->| HOST |<-- -->| | +---+ \ / +-----------+ \ / +---+ X X +---+ / \ +-----------+ / \ +---+ | |<-- -->| RTP |<-- -->| | | B |<------->| RELAY |<------->| D | | | +-----------+ | | +---+ +---+ - Figure 20: DTLS-SRTP host and RTP Relay Separated + Figure 18: DTLS-SRTP host and RTP Relay Separated The relay can still verify that a given SSRC isn't used or spoofed by another participant within the multi-party session by binding SSRCs on their first usage to a given source address and port pair. Packets carrying that source SSRC from other addresses can be suppressed to prevent spoofing. This is possible as long as SRTP is used which leaves the SSRC of the packet originator in RTP and RTCP packets in the clear. If such packet level method for enforcing source authentication within the group, then there exist cryptographic methods such as TESLA [RFC4383] that could be used for true source authentication. A.5. End-point Forwarding - An WebRTC end-point (B in Figure 21) will receive a MediaStream (set - of SSRCs) over a PeerConnection (from A). For the moment is not - decided if the end-point is allowed or not to in its turn send that - MediaStream over another PeerConnection to C. This section discusses - the RTP and end-point implications of allowing such functionality, - which on the API level is extremely simplistic to perform. + An WebRTC end-point (B in Figure 19) will receive a WebRTC + MediaStream (set of SSRCs) over a PeerConnection (from A). For the + moment is not decided if the end-point is allowed or not to in its + turn send that WebRTC MediaStream over another PeerConnection to C. + This section discusses the RTP and end-point implications of allowing + such functionality, which on the API level is extremely simplistic to + perform. +---+ +---+ +---+ | A |--->| B |--->| C | +---+ +---+ +---+ - Figure 21: MediaStream Forwarding + Figure 19: MediaStream Forwarding There exist two main approaches to how B forwards the media from A to - C. The first one is to simply relay the media stream. The second one - is for B to act as a transcoder. Lets consider both approaches. + C. The first one is to simply relay the RTP media stream. The second + one is for B to act as a transcoder. Lets consider both approaches. A relay approache will result in that the WebRTC end-points will have to have the same capabilities as being discussed in Relay (Appendix A.4.3). Thus A will see an RTP session that is extended beyond the PeerConnection and see two different receiving end-points with different path characteristics (B and C). Thus A's congestion control needs to be capable of handling this. The security solution can either support mechanism that allows A to inform C about the key A is using despite B and C having agreed on another set of keys. Alternatively B will decrypt and then re-encrypt using a new key. The relay based approach has the advantage that B does not need to transcode the media thus both maintaining the quality of the encoding and reducing B's complexity requirements. If the right security solutions are supported then also C will be able to verify the authenticity of the media comming from A. As downside A are forced to take both B and C into consideration when delivering content. The media transcoder approach is similar to having B act as Mixer terminating the RTP session combined with the transcoder as discussed in Appendix A.4.1. A will only see B as receiver of its media. B - will responsible to produce a media stream suitable for the B to C - PeerConnection. This may require media transcoding for congestion + will responsible to produce a RTP media stream suitable for the B to + C PeerConnection. This may require media transcoding for congestion control purpose to produce a suitable bit-rate. Thus loosing media quality in the transcoding and forcing B to spend the resource on the transcoding. The media transcoding does result in a separation of the two different legs removing almost all dependencies. B could choice to implement logic to optimize its media transcoding operation, by for example requesting media properties that are suitable for C also, thus trying to avoid it having to transcode the content and only forward the media payloads between the two sides. For that optimization to be practical WebRTC end-points must support sufficiently good tools for codec control. A.6. Simulcast This section discusses simulcast in the meaning of providing a node, for example a stream switching Mixer, with multiple different encoded version of the same media source. In the WebRTC context that appears to be most easily accomplished by establishing mutliple - PeerConnection all being feed the same set of MediaStreams. Each - PeerConnection is then configured to deliver a particular media + PeerConnection all being feed the same set of WebRTC MediaStreams. + Each PeerConnection is then configured to deliver a particular media quality and thus media bit-rate. This will work well as long as the - end-point implements media encoding according to Figure 9. Then each + end-point implements media encoding according to Figure 7. Then each PeerConnection will receive an independently encoded version and the codec parameters can be agreed specifically in the context of this PeerConnection. For simulcast to work one needs to prevent that the end-point deliver - content encoded as depicted in Figure 10. If a single encoder + content encoded as depicted in Figure 8. If a single encoder instance is feed to multiple PeerConnections the intention of performing simulcast will fail. Thus it should be considered to explicitly signal which of the two implementation strategies that are desired and which will be done. At least making the application and possible the central node - interested in receiving simulcast of an end-points media streams to - be aware if it will function or not. + interested in receiving simulcast of an end-points RTP media streams + to be aware if it will function or not. Authors' Addresses Colin Perkins University of Glasgow School of Computing Science Glasgow G12 8QQ United Kingdom Email: csp@csperkins.org