--- 1/draft-ietf-rtcweb-transports-15.txt 2016-10-04 02:15:51.195137041 -0700 +++ 2/draft-ietf-rtcweb-transports-16.txt 2016-10-04 02:15:51.231137963 -0700 @@ -1,18 +1,18 @@ Network Working Group H. Alvestrand Internet-Draft Google -Intended status: Standards Track August 4, 2016 -Expires: February 5, 2017 +Intended status: Standards Track October 4, 2016 +Expires: April 7, 2017 Transports for WebRTC - draft-ietf-rtcweb-transports-15 + draft-ietf-rtcweb-transports-16 Abstract This document describes the data transport protocols used by WebRTC, including the protocols used for interaction with intermediate boxes such as firewalls, relays and NAT boxes. Status of This Memo This Internet-Draft is submitted in full conformance with the @@ -21,21 +21,21 @@ 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 February 5, 2017. + This Internet-Draft will expire on April 7, 2017. Copyright Notice Copyright (c) 2016 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 @@ -49,69 +49,70 @@ 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 2. Requirements language . . . . . . . . . . . . . . . . . . . . 3 3. Transport and Middlebox specification . . . . . . . . . . . . 3 3.1. System-provided interfaces . . . . . . . . . . . . . . . 3 3.2. Ability to use IPv4 and IPv6 . . . . . . . . . . . . . . 4 3.3. Usage of temporary IPv6 addresses . . . . . . . . . . . . 4 3.4. Middle box related functions . . . . . . . . . . . . . . 5 3.5. Transport protocols implemented . . . . . . . . . . . . . 6 4. Media Prioritization . . . . . . . . . . . . . . . . . . . . 7 - 4.1. Local prioritization . . . . . . . . . . . . . . . . . . 7 - 4.2. Usage of Quality of Service - DSCP and Multiplexing . . . 8 - 5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 10 + 4.1. Local prioritization . . . . . . . . . . . . . . . . . . 8 + 4.2. Usage of Quality of Service - DSCP and Multiplexing . . . 9 + 5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 11 6. Security Considerations . . . . . . . . . . . . . . . . . . . 11 7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 11 8. References . . . . . . . . . . . . . . . . . . . . . . . . . 11 8.1. Normative References . . . . . . . . . . . . . . . . . . 11 - 8.2. Informative References . . . . . . . . . . . . . . . . . 14 - Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 15 - A.1. Changes from -00 to -01 . . . . . . . . . . . . . . . . . 15 - A.2. Changes from -01 to -02 . . . . . . . . . . . . . . . . . 16 - A.3. Changes from -02 to -03 . . . . . . . . . . . . . . . . . 16 - A.4. Changes from -03 to -04 . . . . . . . . . . . . . . . . . 16 - A.5. Changes from -04 to -05 . . . . . . . . . . . . . . . . . 17 - A.6. Changes from -05 to -06 . . . . . . . . . . . . . . . . . 17 - A.7. Changes from -06 to -07 . . . . . . . . . . . . . . . . . 17 - A.8. Changes from -07 to -08 . . . . . . . . . . . . . . . . . 17 - A.9. Changes from -08 to -09 . . . . . . . . . . . . . . . . . 18 - A.10. Changes from -09 to -10 . . . . . . . . . . . . . . . . . 18 - A.11. Changes from -10 to -11 . . . . . . . . . . . . . . . . . 18 - A.12. Changes from -11 to -12 . . . . . . . . . . . . . . . . . 18 - A.13. Changes from -12 to -13 . . . . . . . . . . . . . . . . . 18 - A.14. Changes from -13 to -14 . . . . . . . . . . . . . . . . . 18 - A.15. Changes from -14 to -15 . . . . . . . . . . . . . . . . . 18 - Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 19 + 8.2. Informative References . . . . . . . . . . . . . . . . . 15 + Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 16 + A.1. Changes from -00 to -01 . . . . . . . . . . . . . . . . . 16 + A.2. Changes from -01 to -02 . . . . . . . . . . . . . . . . . 17 + A.3. Changes from -02 to -03 . . . . . . . . . . . . . . . . . 17 + A.4. Changes from -03 to -04 . . . . . . . . . . . . . . . . . 17 + A.5. Changes from -04 to -05 . . . . . . . . . . . . . . . . . 18 + A.6. Changes from -05 to -06 . . . . . . . . . . . . . . . . . 18 + A.7. Changes from -06 to -07 . . . . . . . . . . . . . . . . . 18 + A.8. Changes from -07 to -08 . . . . . . . . . . . . . . . . . 18 + A.9. Changes from -08 to -09 . . . . . . . . . . . . . . . . . 19 + A.10. Changes from -09 to -10 . . . . . . . . . . . . . . . . . 19 + A.11. Changes from -10 to -11 . . . . . . . . . . . . . . . . . 19 + A.12. Changes from -11 to -12 . . . . . . . . . . . . . . . . . 19 + A.13. Changes from -12 to -13 . . . . . . . . . . . . . . . . . 19 + A.14. Changes from -13 to -14 . . . . . . . . . . . . . . . . . 19 + A.15. Changes from -14 to -15 . . . . . . . . . . . . . . . . . 19 + A.16. Changes from -15 to -16 . . . . . . . . . . . . . . . . . 20 + Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 20 1. Introduction WebRTC is a protocol suite aimed at real time multimedia exchange between browsers, and between browsers and other entities. WebRTC is described in the WebRTC overview document, [I-D.ietf-rtcweb-overview], which also defines terminology used in - this document, including the terms "WebRTC device" and "WebRTC + this document, including the terms "WebRTC endpoint" and "WebRTC browser". Terminology for RTP sources is taken from[RFC7656] . This document focuses on the data transport protocols that are used by conforming implementations, including the protocols used for interaction with intermediate boxes such as firewalls, relays and NAT boxes. This protocol suite intends to satisfy the security considerations described in the WebRTC security documents, [I-D.ietf-rtcweb-security] and [I-D.ietf-rtcweb-security-arch]. This document describes requirements that apply to all WebRTC - devices. When there are requirements that apply only to WebRTC + endpoints. When there are requirements that apply only to WebRTC browsers, this is called out explicitly. 2. Requirements language The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in RFC 2119 [RFC2119]. 3. Transport and Middlebox specification @@ -131,21 +132,21 @@ For UDP, this specification assumes the ability to set the DSCP code point of the sockets opened on a per-packet basis, in order to achieve the prioritizations described in [I-D.ietf-tsvwg-rtcweb-qos] (see Section 4.2) when multiple media types are multiplexed. It does not assume that the DSCP codepoints will be honored, and does assume that they may be zeroed or changed, since this is a local configuration issue. Platforms that do not give access to these interfaces will not be - able to support a conforming WebRTC implementation. + able to support a conforming WebRTC endpoint. This specification does not assume that the implementation will have access to ICMP or raw IP. The following protocols may be used, but can be implemented by a WebRTC endpoint, and are therefore not defined as "system-provided interfaces": o TURN - Traversal Using Relays Around NAT, [RFC5766] @@ -176,29 +177,31 @@ that temporary addresses [RFC4941] are to be preferred over permanent addresses. This is a change from the rules specified by [RFC3484]. For applications that select a single address, this is usually done by the IPV6_PREFER_SRC_TMP preference flag specified in [RFC5014]. However, this rule, which is intended to ensure that privacy-enhanced addresses are used in preference to static addresses, doesn't have the right effect in ICE, where all addresses are gathered and therefore revealed to the application. Therefore, the following rule is applied instead: - When a client gathers all IPv6 addresses on a host, and both non- - deprecated temporary addresses and permanent addresses of the same - scope are present, the client SHOULD discard the permanent addresses - before exposing addresses to the application or using them in ICE. - This is consistent with the default policy described in [RFC6724]. + When a WebRTC endpoint gathers all IPv6 addresses on its host, and + both non-deprecated temporary addresses and permanent addresses of + the same scope are present, the WebRTC endpoint SHOULD discard the + permanent addresses before exposing addresses to the application or + using them in ICE. This is consistent with the default policy + described in [RFC6724]. If some of the temporary IPv6 addresses, but not all, are marked - deprecated, the client SHOULD discard the deprecated addresses. In - an ICE restart, deprecated addresses that are currently in use MAY be + deprecated, the WebRTC endpoint SHOULD discard the deprecated + addresses, unless they are used by an ongoing connection. In an ICE + restart, deprecated addresses that are currently in use MAY be retained. 3.4. Middle box related functions The primary mechanism to deal with middle boxes is ICE, which is an appropriate way to deal with NAT boxes and firewalls that accept traffic from the inside, but only from the outside if it is in response to inside traffic (simple stateful firewalls). ICE [RFC5245] MUST be supported. The implementation MUST be a full @@ -206,102 +209,110 @@ interworking with both ICE and ICE-Lite implementations when they are deployed appropriately. In order to deal with situations where both parties are behind NATs of the type that perform endpoint-dependent mapping (as defined in [RFC5128] section 2.4), TURN [RFC5766] MUST be supported. WebRTC browsers MUST support configuration of STUN and TURN servers, both from browser configuration and from an application. + Note that there is other work around STUN and TURN sever discovery + and management, including [I-D.ietf-tram-turn-server-discovery] for + server discovery, as well as [I-D.ietf-rtcweb-return]. + In order to deal with firewalls that block all UDP traffic, the mode - of TURN that uses TCP between the client and the server MUST be - supported, and the mode of TURN that uses TLS over TCP between the - client and the server MUST be supported. See [RFC5766] section 2.1 - for details. + of TURN that uses TCP between the WebRTC endpoint and the TURN server + MUST be supported, and the mode of TURN that uses TLS over TCP + between the WebRTC endpoint and the TURN server MUST be supported. + See [RFC5766] section 2.1 for details. In order to deal with situations where one party is on an IPv4 network and the other party is on an IPv6 network, TURN extensions for IPv6 [RFC6156] MUST be supported. - TURN TCP candidates, where the connection from the client's TURN - server to the peer is a TCP connection, [RFC6062] MAY be supported. + TURN TCP candidates, where the connection from the WebRTC endpoint's + TURN server to the peer is a TCP connection, [RFC6062] MAY be + supported. However, such candidates are not seen as providing any significant benefit, for the following reasons. First, use of TURN TCP candidates would only be relevant in cases which both peers are required to use TCP to establish a PeerConnection. Second, that use case is supported in a different way by both sides establishing UDP relay candidates using TURN over TCP to connect to their respective relay servers. - Third, using TCP between the client's TURN server and the peer may - result in more performance problems than using UDP, e.g. due to head - of line blocking. + Third, using TCP between the WebRTC endpoint's TURN server and the + peer may result in more performance problems than using UDP, e.g. due + to head of line blocking. ICE-TCP candidates [RFC6544] MUST be supported; this may allow applications to communicate to peers with public IP addresses across UDP-blocking firewalls without using a TURN server. If TCP connections are used, RTP framing according to [RFC4571] MUST be used for all packets. This includes the RTP packets, DTLS packets used to carry data channels, and STUN connectivity check packets. The ALTERNATE-SERVER mechanism specified in [RFC5389] (STUN) section 11 (300 Try Alternate) MUST be supported. - The WebRTC implementation MAY support accessing the Internet through - an HTTP proxy. If it does so, it MUST include the "ALPN" header as + The WebRTC endpoint MAY support accessing the Internet through an + HTTP proxy. If it does so, it MUST include the "ALPN" header as specified in [RFC7639], and proxy authentication as described in Section 4.3.6 of [RFC7231] and [RFC7235] MUST also be supported. 3.5. Transport protocols implemented For transport of media, secure RTP is used. The details of the profile of RTP used are described in "RTP Usage" - [I-D.ietf-rtcweb-rtp-usage]. Key exchange MUST be done using DTLS- - SRTP, as described in [I-D.ietf-rtcweb-security-arch]. + [I-D.ietf-rtcweb-rtp-usage], which mandates the use of a circuit + breaker [I-D.ietf-avtcore-rtp-circuit-breakers] and congstion control + (see [I-D.ietf-rmcat-cc-requirements] for further guidance). + + Key exchange MUST be done using DTLS-SRTP, as described in + [I-D.ietf-rtcweb-security-arch]. For data transport over the WebRTC data channel - [I-D.ietf-rtcweb-data-channel], WebRTC implementations MUST support - SCTP over DTLS over ICE. This encapsulation is specified in + [I-D.ietf-rtcweb-data-channel], WebRTC endpoints MUST support SCTP + over DTLS over ICE. This encapsulation is specified in [I-D.ietf-tsvwg-sctp-dtls-encaps]. Negotiation of this transport in SDP is defined in [I-D.ietf-mmusic-sctp-sdp]. The SCTP extension for NDATA, [I-D.ietf-tsvwg-sctp-ndata], MUST be supported. The setup protocol for WebRTC data channels described in [I-D.ietf-rtcweb-data-protocol] MUST be supported. Note: DTLS-SRTP as defined in [RFC5764] section 6.7.1 defines the interaction between DTLS and ICE ( [RFC5245]). The effect of this specification is that all ICE candidate pairs associated with a single component are part of the same DTLS association. Thus, there will only be one DTLS handshake even if there are multiple valid candidate pairs. - WebRTC implementations MUST support multiplexing of DTLS and RTP over - the same port pair, as described in the DTLS-SRTP specification + WebRTC endpoints MUST support multiplexing of DTLS and RTP over the + same port pair, as described in the DTLS-SRTP specification [RFC5764], section 5.1.2, with clarifications in - [I-D.ietf-avtcore-rfc5764-mux-fixes]. All application layer protocol payloads over this DTLS connection are SCTP packets. Protocol identification MUST be supplied as part of the DTLS handshake, as specified in [I-D.ietf-rtcweb-alpn]. 4. Media Prioritization The WebRTC prioritization model is that the application tells the - WebRTC implementation about the priority of media and data that is + WebRTC endpoint about the priority of media and data that is controlled from the API. In this context, a "flow" is used for the units that are given a specific priority through the WebRTC API. For media, a "media flow", which can be an "audio flow" or a "video flow", is what [RFC7656] calls a "media source", which results in a "source RTP stream" and one or more "redundancy RTP streams". This specification does not describe prioritization between the RTP streams that come from a single "media source". @@ -320,24 +331,24 @@ sequence decisions and packet markings. Each is described in its own section below. 4.1. Local prioritization Local prioritization is applied at the local node, before the packet is sent. This means that the prioritization has full access to the data about the individual packets, and can choose differing treatment based on the stream a packet belongs to. - When an WebRTC implementation has packets to send on multiple streams - that are congestion-controlled under the same congestion control - regime, the WebRTC implementation SHOULD cause data to be emitted in - such a way that each stream at each level of priority is being given + When an WebRTC endpoint has packets to send on multiple streams that + are congestion-controlled under the same congestion control regime, + the WebRTC endpoint SHOULD cause data to be emitted in such a way + that each stream at each level of priority is being given approximately twice the transmission capacity (measured in payload bytes) of the level below. Thus, when congestion occurs, a "high" priority flow will have the ability to send 8 times as much data as a "very-low" priority flow if both have data to send. This prioritization is independent of the media type. The details of which packet to send first are implementation defined. For example: If there is a high priority audio flow sending 100 byte @@ -369,20 +380,25 @@ Any combination of these, or other schemes that have the same effect, is valid, as long as the distribution of transmission capacity is approximately correct. For media, it is usually inappropriate to use deep queues for sending; it is more useful to, for instance, skip intermediate frames that have no dependencies on them in order to achieve a lower bitrate. For reliable data, queues are useful. + Note that this specification doesn't dictate when disparate streams + are to be "congestion controlled under the same congestion control + regime". The issue of coupling congestion controllers is explored + further in [I-D.ietf-rmcat-coupled-cc]. + 4.2. Usage of Quality of Service - DSCP and Multiplexing When the packet is sent, the network will make decisions about queueing and/or discarding the packet that can affect the quality of the communication. The sender can attempt to set the DSCP field of the packet to influence these decisions. Implementations SHOULD attempt to set QoS on the packets sent, according to the guidelines in [I-D.ietf-tsvwg-rtcweb-qos]. It is appropriate to depart from this recommendation when running on @@ -502,46 +518,62 @@ 8.1. Normative References [I-D.ietf-avtcore-rfc5764-mux-fixes] Petit-Huguenin, M. and G. Salgueiro, "Multiplexing Scheme Updates for Secure Real-time Transport Protocol (SRTP) Extension for Datagram Transport Layer Security (DTLS)", draft-ietf-avtcore-rfc5764-mux-fixes-10 (work in progress), July 2016. + [I-D.ietf-avtcore-rtp-circuit-breakers] + Perkins, C. and V. Singh, "Multimedia Congestion Control: + Circuit Breakers for Unicast RTP Sessions", draft-ietf- + avtcore-rtp-circuit-breakers-18 (work in progress), August + 2016. + [I-D.ietf-mmusic-ice-dualstack-fairness] Martinsen, P., Reddy, T., and P. Patil, "ICE Multihomed and IPv4/IPv6 Dual Stack Fairness", draft-ietf-mmusic-ice- dualstack-fairness-02 (work in progress), September 2015. [I-D.ietf-mmusic-sctp-sdp] Holmberg, C., Loreto, S., and G. Camarillo, "Stream Control Transmission Protocol (SCTP)-Based Media Transport in the Session Description Protocol (SDP)", draft-ietf- mmusic-sctp-sdp-16 (work in progress), February 2016. + [I-D.ietf-rmcat-cc-requirements] + Jesup, R. and Z. Sarker, "Congestion Control Requirements + for Interactive Real-Time Media", draft-ietf-rmcat-cc- + requirements-09 (work in progress), December 2014. + [I-D.ietf-rtcweb-alpn] Thomson, M., "Application Layer Protocol Negotiation for Web Real-Time Communications (WebRTC)", draft-ietf-rtcweb- alpn-04 (work in progress), May 2016. [I-D.ietf-rtcweb-data-channel] Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channels", draft-ietf-rtcweb-data-channel-13 (work in progress), January 2015. [I-D.ietf-rtcweb-data-protocol] Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel Establishment Protocol", draft-ietf-rtcweb-data- protocol-09 (work in progress), January 2015. + [I-D.ietf-rtcweb-overview] + Alvestrand, H., "Overview: Real Time Protocols for + Browser-based Applications", draft-ietf-rtcweb-overview-15 + (work in progress), January 2016. + [I-D.ietf-rtcweb-rtp-usage] Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time Communication (WebRTC): Media Transport and Use of RTP", draft-ietf-rtcweb-rtp-usage-26 (work in progress), March 2016. [I-D.ietf-rtcweb-security] Rescorla, E., "Security Considerations for WebRTC", draft- ietf-rtcweb-security-08 (work in progress), February 2015. @@ -576,20 +608,25 @@ [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/ RFC2119, March 1997, . [RFC4571] Lazzaro, J., "Framing Real-time Transport Protocol (RTP) and RTP Control Protocol (RTCP) Packets over Connection- Oriented Transport", RFC 4571, DOI 10.17487/RFC4571, July 2006, . + [RFC4594] Babiarz, J., Chan, K., and F. Baker, "Configuration + Guidelines for DiffServ Service Classes", RFC 4594, DOI + 10.17487/RFC4594, August 2006, + . + [RFC4941] Narten, T., Draves, R., and S. Krishnan, "Privacy Extensions for Stateless Address Autoconfiguration in IPv6", RFC 4941, DOI 10.17487/RFC4941, September 2007, . [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols", RFC 5245, DOI 10.17487/RFC5245, April 2010, . @@ -647,53 +684,58 @@ [RFC7235] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer Protocol (HTTP/1.1): Authentication", RFC 7235, DOI 10.17487/RFC7235, June 2014, . [RFC7639] Hutton, A., Uberti, J., and M. Thomson, "The ALPN HTTP Header Field", RFC 7639, DOI 10.17487/RFC7639, August 2015, . + [RFC7656] Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and + B. Burman, Ed., "A Taxonomy of Semantics and Mechanisms + for Real-Time Transport Protocol (RTP) Sources", RFC 7656, + DOI 10.17487/RFC7656, November 2015, + . + 8.2. Informative References - [I-D.ietf-rtcweb-overview] - Alvestrand, H., "Overview: Real Time Protocols for - Browser-based Applications", draft-ietf-rtcweb-overview-15 - (work in progress), January 2016. + [I-D.ietf-rmcat-coupled-cc] + Islam, S., Welzl, M., and S. Gjessing, "Coupled congestion + control for RTP media", draft-ietf-rmcat-coupled-cc-03 + (work in progress), July 2016. + + [I-D.ietf-rtcweb-return] + Schwartz, B. and J. Uberti, "Recursively Encapsulated TURN + (RETURN) for Connectivity and Privacy in WebRTC", draft- + ietf-rtcweb-return-01 (work in progress), January 2016. + + [I-D.ietf-tram-turn-server-discovery] + Patil, P., Reddy, T., and D. Wing, "TURN Server Auto + Discovery", draft-ietf-tram-turn-server-discovery-09 (work + in progress), August 2016. [RFC3484] Draves, R., "Default Address Selection for Internet Protocol version 6 (IPv6)", RFC 3484, DOI 10.17487/ RFC3484, February 2003, . - [RFC4594] Babiarz, J., Chan, K., and F. Baker, "Configuration - Guidelines for DiffServ Service Classes", RFC 4594, DOI - 10.17487/RFC4594, August 2006, - . - [RFC5014] Nordmark, E., Chakrabarti, S., and J. Laganier, "IPv6 Socket API for Source Address Selection", RFC 5014, DOI 10.17487/RFC5014, September 2007, . [RFC5128] Srisuresh, P., Ford, B., and D. Kegel, "State of Peer-to- Peer (P2P) Communication across Network Address Translators (NATs)", RFC 5128, DOI 10.17487/RFC5128, March 2008, . - [RFC7656] Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and - B. Burman, Ed., "A Taxonomy of Semantics and Mechanisms - for Real-Time Transport Protocol (RTP) Sources", RFC 7656, - DOI 10.17487/RFC7656, November 2015, - . - [RFC7657] Black, D., Ed. and P. Jones, "Differentiated Services (Diffserv) and Real-Time Communication", RFC 7657, DOI 10.17487/RFC7657, November 2015, . Appendix A. Change log This section should be removed before publication as an RFC. A.1. Changes from -00 to -01 @@ -853,16 +895,39 @@ o Various text clarifications based on comments in Last Call and IESG review o Clarified that only non-deprecated IPv6 addresses are used o Described handling of downgrading of DSCP markings when blackholes are detected o Expanded acronyms in a new protocol list +A.16. Changes from -15 to -16 + + These changes are done post IESG approval, and address IESG comments + and other late comments. Issue numbers refer to https://github.com/ + rtcweb-wg/rtcweb-transport/issues. + + o Moved RFC 4594, 7656 and -overview to normative (issue #28) + + o Changed the terms "client", "WebRTC implementation" and "WebRTC + device" to consistently be "WebRTC endpoint", as defined in + -overview. (issue #40) + + o Added a note mentioning TURN service discovery and RETURN (issue + #42) + + o Added a note mentioning that rtp-usage requires circut breaker and + congestion control (issue #43) + + o Added mention of the "don't discard temporary IPv6 addresses that + are in use" (issue #44) + + o Added a reference to draft-ietf-rmcat-coupled-cc (issue #46) + Author's Address Harald Alvestrand Google Email: harald@alvestrand.no