Network Working Group                                      H. Alvestrand
Internet-Draft                                                    Google
Intended status: Standards Track                        January 28, 2016
Expires: July 31, 2016

                         Transports for WebRTC


   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
   provisions of BCP 78 and BCP 79.

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   This Internet-Draft will expire on July 31, 2016.

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Table of Contents

   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  . . . . . . . . . . . . . .   4
     3.5.  Transport protocols implemented . . . . . . . . . . . . .   6
   4.  Media Prioritization  . . . . . . . . . . . . . . . . . . . .   6
     4.1.  Local prioritization  . . . . . . . . . . . . . . . . . .   7
     4.2.  Usage of Quality of Service - DSCP and Multiplexing . . .   8
   5.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .   9
   6.  Security Considerations . . . . . . . . . . . . . . . . . . .  10
   7.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  10
   8.  References  . . . . . . . . . . . . . . . . . . . . . . . . .  10
     8.1.  Normative References  . . . . . . . . . . . . . . . . . .  10
     8.2.  Informative References  . . . . . . . . . . . . . . . . .  12
   Appendix A.  Change log . . . . . . . . . . . . . . . . . . . . .  13
     A.1.  Changes from -00 to -01 . . . . . . . . . . . . . . . . .  13
     A.2.  Changes from -01 to -02 . . . . . . . . . . . . . . . . .  13
     A.3.  Changes from -02 to -03 . . . . . . . . . . . . . . . . .  14
     A.4.  Changes from -03 to -04 . . . . . . . . . . . . . . . . .  14
     A.5.  Changes from -04 to -05 . . . . . . . . . . . . . . . . .  14
     A.6.  Changes from -05 to -06 . . . . . . . . . . . . . . . . .  15
     A.7.  Changes from -06 to -07 . . . . . . . . . . . . . . . . .  15
     A.8.  Changes from -07 to -08 . . . . . . . . . . . . . . . . .  15
     A.9.  Changes from -08 to -09 . . . . . . . . . . . . . . . . .  15
     A.10. Changes from -09 to -10 . . . . . . . . . . . . . . . . .  16
     A.11. Changes from -10 to -11 . . . . . . . . . . . . . . . . .  16
   Author's Address  . . . . . . . . . . . . . . . . . . . . . . . .  16

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

   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

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   interaction with intermediate boxes such as firewalls, relays and NAT

   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
   browsers, this is called out explicitly.

2.  Requirements language

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   document are to be interpreted as described in RFC 2119 [RFC2119].

3.  Transport and Middlebox specification

3.1.  System-provided interfaces

   The protocol specifications used here assume that the following
   protocols are available to the implementations of the WebRTC

   o  UDP [RFC0768].  This is the protocol assumed by most protocol
      elements described.

   o  TCP [RFC0793].  This is used for HTTP/WebSockets, as well as for
      TURN/SSL and ICE-TCP.

   For both protocols, IPv4 and IPv6 support is assumed.

   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.

   This specification does not assume that the implementation will have
   access to ICMP or raw IP.

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3.2.  Ability to use IPv4 and IPv6

   Web applications running in a WebRTC browser MUST be able to utilize
   both IPv4 and IPv6 where available - that is, when two peers have
   only IPv4 connectivity to each other, or they have only IPv6
   connectivity to each other, applications running in the WebRTC
   browser MUST be able to communicate.

   When TURN is used, and the TURN server has IPv4 or IPv6 connectivity
   to the peer or its TURN server, candidates of the appropriate types
   MUST be supported.  The "Happy Eyeballs" specification for ICE
   [I-D.martinsen-mmusic-ice-dualstack-fairness] SHOULD be supported.

3.3.  Usage of temporary IPv6 addresses

   The IPv6 default address selection specification [RFC6724] specifies
   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 is not completely obvious in the ICE scope.  This
   is therefore clarified as follows:

   When a client gathers all IPv6 addresses on a host, and both
   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].

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
   ICE implementation, not ICE-Lite.  A full ICE implementation allows
   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.

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   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.

   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.

   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

   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 only between the endpoint and its relay may result
   in less issues with TCP in regards to real-time constraints, 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, both for the RTP packets and for the DTLS packets used to
   carry data channels.

   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 support 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.

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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].

   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-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
   [RFC5764], section 5.1.2.  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
   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".

   A "data flow" is the outgoing data on a single WebRTC data channel.

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   The priority associated with a media flow or data flow is classified
   as "very-low", "low", "medium or "high".  There are only four
   priority levels at the API.

   The priority settings affect two pieces of behavior: Packet send
   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 controller,
   the WebRTC implementation 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
   packets, and a low priority video flow sending 1000 byte packets, and
   outgoing capacity exists for sending >5000 payload bytes, it would be
   appropriate to send 4000 bytes (40 packets) of audio and 1000 bytes
   (one packet) of video as the result of a single pass of sending

   Conversely, if the audio flow is marked low priority and the video
   flow is marked high priority, the scheduler may decide to send 2
   video packets (2000 bytes) and 5 audio packets (500 bytes) when
   outgoing capacity exists for sending > 2500 payload bytes.

   If there are two high priority audio flows, each will be able to send
   4000 bytes in the same period where a low priority video flow is able
   to send 1000 bytes.

   Two example implementation strategies are:

   o  When the available bandwidth is known from the congestion control
      algorithm, configure each codec and each data channel with a

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      target send rate that is appropriate to its share of the available

   o  When congestion control indicates that a specified number of
      packets can be sent, send packets that are available to send using
      a weighted round robin scheme across the connections.

   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.

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
   platforms where QoS marking is not implemented.

   The implementation MAY turn off use of DSCP markings if it detects
   symptoms of unexpected behaviour like priority inversion or blocking
   of packets with certain DSCP markings.  The detection of these
   conditions is implementation dependent.

   All packets carrying data from the SCTP association supporting the
   data channels MUST use a single DSCP code point.  The code point used
   SHOULD be that recommended by [I-D.ietf-tsvwg-rtcweb-qos] for the
   highest priority data channel carried.  Note that this means that all
   data packets, no matter what their relative priority is, will be
   treated the same by the network.

   All packets on one TCP connection, no matter what it carries, MUST
   use a single DSCP code point.

   More advice on the use of DSCP code points with RTP is given in

   There exist a number of schemes for achieving quality of service that
   do not depend solely on DSCP code points.  Some of these schemes

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   depend on classifying the traffic into flows based on 5-tuple (source
   address, source port, protocol, destination address, destination
   port) or 6-tuple (5-tuple + DSCP code point).  Under differing
   conditions, it may therefore make sense for a sending application to
   choose any of the configurations:

   o  Each media stream carried on its own 5-tuple

   o  Media streams grouped by media type into 5-tuples (such as
      carrying all audio on one 5-tuple)

   o  All media sent over a single 5-tuple, with or without
      differentiation into 6-tuples based on DSCP code points

   In each of the configurations mentioned, data channels may be carried
   in its own 5-tuple, or multiplexed together with one of the media

   More complex configurations, such as sending a high priority video
   stream on one 5-tuple and sending all other video streams multiplexed
   together over another 5-tuple, can also be envisioned.  More
   information on mapping media flows to 5-tuples can be found in

   A sending implementation MUST be able to support the following

   o  multiplex all media and data on a single 5-tuple (fully bundled)

   o  send each media stream on its own 5-tuple and data on its own
      5-tuple (fully unbundled)

   It MAY choose to support other configurations, such as bundling each
   media type (audio, video or data) into its own 5-tuple (bundling by
   media type).

   Sending data over multiple 5-tuples is not supported.

   A receiving implementation MUST be able to receive media and data in
   all these configurations.

5.  IANA Considerations

   This document makes no request of IANA.

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

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6.  Security Considerations

   Security considerations are enumerated in [I-D.ietf-rtcweb-security].

7.  Acknowledgements

   This document is based on earlier versions embedded in
   [I-D.ietf-rtcweb-overview], which were the results of contributions
   from many RTCWEB WG members.

   Special thanks for reviews of earlier versions of this draft go to
   Eduardo Gueiros, Magnus Westerlund, Markus Isomaki and Dan Wing; the
   contributions from Andrew Hutton also deserve special mention.

8.  References

8.1.  Normative References

              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-12 (work in progress), January 2015.

              Thomson, M., "Application Layer Protocol Negotiation for
              Web Real-Time Communications (WebRTC)", draft-ietf-rtcweb-
              alpn-00 (work in progress), July 2014.

              Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data
              Channels", draft-ietf-rtcweb-data-channel-13 (work in
              progress), January 2015.

              Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel
              Establishment Protocol", draft-ietf-rtcweb-data-
              protocol-09 (work in progress), January 2015.

              Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
              Communication (WebRTC): Media Transport and Use of RTP",
              draft-ietf-rtcweb-rtp-usage-22 (work in progress),
              February 2015.

              Rescorla, E., "Security Considerations for WebRTC", draft-
              ietf-rtcweb-security-07 (work in progress), July 2014.

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              Rescorla, E., "WebRTC Security Architecture", draft-ietf-
              rtcweb-security-arch-10 (work in progress), July 2014.

              Dhesikan, S., Jennings, C., Druta, D., Jones, P., and J.
              Polk, "DSCP and other packet markings for RTCWeb QoS",
              draft-ietf-tsvwg-rtcweb-qos-03 (work in progress),
              November 2014.

              Tuexen, M., Stewart, R., Jesup, R., and S. Loreto, "DTLS
              Encapsulation of SCTP Packets", draft-ietf-tsvwg-sctp-
              dtls-encaps-09 (work in progress), January 2015.

              Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann,
              "Stream Schedulers and a New Data Chunk for the Stream
              Control Transmission Protocol", draft-ietf-tsvwg-sctp-
              ndata-02 (work in progress), January 2015.

              Martinsen, P., Reddy, T., and P. Patil, "ICE IPv4/IPv6
              Dual Stack Fairness", draft-martinsen-mmusic-ice-
              dualstack-fairness-02 (work in progress), February 2015.

   [RFC0768]  Postel, J., "User Datagram Protocol", STD 6, RFC 768,
              August 1980.

   [RFC0793]  Postel, J., "Transmission Control Protocol", STD 7, RFC
              793, September 1981.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC4571]  Lazzaro, J., "Framing Real-time Transport Protocol (RTP)
              and RTP Control Protocol (RTCP) Packets over Connection-
              Oriented Transport", RFC 4571, July 2006.

   [RFC4941]  Narten, T., Draves, R., and S. Krishnan, "Privacy
              Extensions for Stateless Address Autoconfiguration in
              IPv6", RFC 4941, September 2007.

   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245, April

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   [RFC5389]  Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
              "Session Traversal Utilities for NAT (STUN)", RFC 5389,
              October 2008.

   [RFC5764]  McGrew, D. and E. Rescorla, "Datagram Transport Layer
              Security (DTLS) Extension to Establish Keys for the Secure
              Real-time Transport Protocol (SRTP)", RFC 5764, May 2010.

   [RFC5766]  Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using
              Relays around NAT (TURN): Relay Extensions to Session
              Traversal Utilities for NAT (STUN)", RFC 5766, April 2010.

   [RFC6062]  Perreault, S. and J. Rosenberg, "Traversal Using Relays
              around NAT (TURN) Extensions for TCP Allocations", RFC
              6062, November 2010.

   [RFC6156]  Camarillo, G., Novo, O., and S. Perreault, "Traversal
              Using Relays around NAT (TURN) Extension for IPv6", RFC
              6156, April 2011.

   [RFC6544]  Rosenberg, J., Keranen, A., Lowekamp, B., and A. Roach,
              "TCP Candidates with Interactive Connectivity
              Establishment (ICE)", RFC 6544, March 2012.

   [RFC6724]  Thaler, D., Draves, R., Matsumoto, A., and T. Chown,
              "Default Address Selection for Internet Protocol Version 6
              (IPv6)", RFC 6724, September 2012.

   [RFC7231]  Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
              Protocol (HTTP/1.1): Semantics and Content", RFC 7231, DOI
              10.17487/RFC7231, June 2014,

   [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, <>.

8.2.  Informative References

              Black, D. and P. Jones, "Differentiated Services
              (DiffServ) and Real-time Communication", draft-ietf-dart-
              dscp-rtp-10 (work in progress), November 2014.

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              Alvestrand, H., "Overview: Real Time Protocols for
              Browser-based Applications", draft-ietf-rtcweb-overview-13
              (work in progress), November 2014.

   [RFC3484]  Draves, R., "Default Address Selection for Internet
              Protocol version 6 (IPv6)", RFC 3484, February 2003.

   [RFC5014]  Nordmark, E., Chakrabarti, S., and J. Laganier, "IPv6
              Socket API for Source Address Selection", RFC 5014,
              September 2007.

   [RFC5128]  Srisuresh, P., Ford, B., and D. Kegel, "State of Peer-to-
              Peer (P2P) Communication across Network Address
              Translators (NATs)", RFC 5128, 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,

Appendix A.  Change log

   This section should be removed before publication as an RFC.

A.1.  Changes from -00 to -01

   o  Clarified DSCP requirements, with reference to -qos-

   o  Clarified "symmetric NAT" -> "NATs which perform endpoint-
      dependent mapping"

   o  Made support of TURN over TCP mandatory

   o  Made support of TURN over TLS a MAY, and added open question

   o  Added an informative reference to -firewalls-

   o  Called out that we don't make requirements on HTTP proxy
      interaction (yet

A.2.  Changes from -01 to -02

   o  Required support for 300 Alternate Server from STUN.

   o  Separated the ICE-TCP candidate requirement from the TURN-TCP

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   o  Added new sections on using QoS functions, and on multiplexing

   o  Removed all mention of RTP profiles.  Those are the business of
      the RTP usage draft, not this one.

   o  Required support for TURN IPv6 extensions.

   o  Removed reference to the TURN URI scheme, as it was unnecessary.

   o  Made an explicit statement that multiplexing (or not) is an
      application matter.


A.3.  Changes from -02 to -03

   o  Added required support for draft-ietf-tsvwg-sctp-ndata

   o  Removed discussion of multiplexing, since this is present in rtp-

   o  Added RFC 4571 reference for framing RTP packets over TCP.

   o  Downgraded TURN TCP candidates from SHOULD to MAY, and added more
      language discussing TCP usage.

   o  Added language on IPv6 temporary addresses.

   o  Added language describing multiplexing choices.

   o  Added a separate section detailing what it means when we say that
      an WebRTC implementation MUST support both IPv4 and IPv6.

A.4.  Changes from -03 to -04

   o  Added a section on prioritization, moved the DSCP section into it,
      and added a section on local prioritization, giving a specific
      algorithm for interpreting "priority" in local prioritization.

   o  ICE-TCP candidates was changed from MAY to MUST, in recognition of
      the sense of the room at the London IETF.

A.5.  Changes from -04 to -05

   o  Reworded introduction

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   o  Removed all references to "WebRTC".  It now uses only the term

   o  Addressed a number of clarity / language comments

   o  Rewrote the prioritization to cover data channels and to describe
      multiple ways of prioritizing flows

   o  Made explicit reference to "MUST do DTLS-SRTP", and referred to
      security-arch for details

A.6.  Changes from -05 to -06

   o  Changed all references to "RTCWEB" to "WebRTC", except one
      reference to the working group

   o  Added reference to the httpbis "connect" protocol (being adopted
      by HTTPBIS)

   o  Added reference to the ALPN header (being adopted by RTCWEB)

   o  Added reference to the DART RTP document

   o  Said explicitly that SCTP for data channels has a single DSCP

A.7.  Changes from -06 to -07

   o  Updated references

   o  Removed reference to draft-hutton-rtcweb-nat-firewall-

A.8.  Changes from -07 to -08

   o  Updated references

   o  Deleted "bundle each media type (audio, video or data) into its
      own 5-tuple (bundling by media type)" from MUST support
      configuration, since JSEP does not have a means to negotiate this

A.9.  Changes from -08 to -09

   o  Added a clarifying note about DTLS-SRTP and ICE interaction.

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A.10.  Changes from -09 to -10

   o  Re-added references to proxy authentication lost in 07-08
      transition (Bug #5)

   o  Rearranged and rephrased text in section 4 about prioritization to
      reflect discussions in TSVWG.

   o  Changed the "Connect" header to "ALPN", and updated reference.
      (Bug #6)

A.11.  Changes from -10 to -11

   o  Added a definition of the term "flow" used in the prioritization

   o  Changed the names of the four priority levels to conform to other

Author's Address

   Harald Alvestrand


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