draft-ietf-rtcweb-transports-05.txt   draft-ietf-rtcweb-transports-06.txt 
Network Working Group H. Alvestrand Network Working Group H. Alvestrand
Internet-Draft Google Internet-Draft Google
Intended status: Standards Track June 11, 2014 Intended status: Standards Track August 11, 2014
Expires: December 13, 2014 Expires: February 12, 2015
Transports for RTCWEB Transports for WebRTC
draft-ietf-rtcweb-transports-05 draft-ietf-rtcweb-transports-06
Abstract Abstract
This document describes the data transport protocols used by RTCWEB, This document describes the data transport protocols used by WebRTC,
including the protocols used for interaction with intermediate boxes including the protocols used for interaction with intermediate boxes
such as firewalls, relays and NAT boxes. such as firewalls, relays and NAT boxes.
Status of This Memo Status of This Memo
This Internet-Draft is submitted in full conformance with the This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79. provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet- working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/. Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress." material or to cite them other than as "work in progress."
This Internet-Draft will expire on December 13, 2014. This Internet-Draft will expire on February 12, 2015.
Copyright Notice Copyright Notice
Copyright (c) 2014 IETF Trust and the persons identified as the Copyright (c) 2014 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of (http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents publication of this document. Please review these documents
skipping to change at page 2, line 17 skipping to change at page 2, line 17
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Requirements language . . . . . . . . . . . . . . . . . . . . 3 2. Requirements language . . . . . . . . . . . . . . . . . . . . 3
3. Transport and Middlebox specification . . . . . . . . . . . . 3 3. Transport and Middlebox specification . . . . . . . . . . . . 3
3.1. System-provided interfaces . . . . . . . . . . . . . . . 3 3.1. System-provided interfaces . . . . . . . . . . . . . . . 3
3.2. Ability to use IPv4 and IPv6 . . . . . . . . . . . . . . 3 3.2. Ability to use IPv4 and IPv6 . . . . . . . . . . . . . . 3
3.3. Usage of temporary IPv6 addresses . . . . . . . . . . . . 4 3.3. Usage of temporary IPv6 addresses . . . . . . . . . . . . 4
3.4. Middle box related functions . . . . . . . . . . . . . . 4 3.4. Middle box related functions . . . . . . . . . . . . . . 4
3.5. Transport protocols implemented . . . . . . . . . . . . . 5 3.5. Transport protocols implemented . . . . . . . . . . . . . 5
4. Media Prioritization . . . . . . . . . . . . . . . . . . . . 6 4. Media Prioritization . . . . . . . . . . . . . . . . . . . . 6
4.1. Usage of Quality of Service - DSCP and Multiplexing . . . 6 4.1. Usage of Quality of Service - DSCP and Multiplexing . . . 6
4.2. Local prioritization . . . . . . . . . . . . . . . . . . 7 4.2. Local prioritization . . . . . . . . . . . . . . . . . . 8
5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 8 5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 9
6. Security Considerations . . . . . . . . . . . . . . . . . . . 8 6. Security Considerations . . . . . . . . . . . . . . . . . . . 9
7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 9 7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 9
8. References . . . . . . . . . . . . . . . . . . . . . . . . . 9 8. References . . . . . . . . . . . . . . . . . . . . . . . . . 9
8.1. Normative References . . . . . . . . . . . . . . . . . . 9 8.1. Normative References . . . . . . . . . . . . . . . . . . 9
8.2. Informative References . . . . . . . . . . . . . . . . . 11 8.2. Informative References . . . . . . . . . . . . . . . . . 11
Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 11 Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 12
A.1. Changes from -00 to -01 . . . . . . . . . . . . . . . . . 11 A.1. Changes from -00 to -01 . . . . . . . . . . . . . . . . . 12
A.2. Changes from -01 to -02 . . . . . . . . . . . . . . . . . 12 A.2. Changes from -01 to -02 . . . . . . . . . . . . . . . . . 13
A.3. Changes from -02 to -03 . . . . . . . . . . . . . . . . . 12 A.3. Changes from -02 to -03 . . . . . . . . . . . . . . . . . 13
A.4. Changes from -03 to -04 . . . . . . . . . . . . . . . . . 13 A.4. Changes from -03 to -04 . . . . . . . . . . . . . . . . . 13
A.5. Changes from -04 to -05 . . . . . . . . . . . . . . . . . 13 A.5. Changes from -04 to -05 . . . . . . . . . . . . . . . . . 14
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 13 A.6. Changes from -05 to -06 . . . . . . . . . . . . . . . . . 14
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 14
1. Introduction 1. Introduction
RTCWEB is a protocol suite aimed at real time multimedia exchange WebRTC is a protocol suite aimed at real time multimedia exchange
between browsers, and between browsers and other entities. between browsers, and between browsers and other entities.
RTCWEB is described in the RTCWEB overview document, WebRTC is described in the WebRTC overview document,
[I-D.ietf-rtcweb-overview]. [I-D.ietf-rtcweb-overview], which also defines terminology used in
this document.
This document focuses on the data transport protocols that are used This document focuses on the data transport protocols that are used
by conforming implementations, including the protocols used for by conforming implementations, including the protocols used for
interaction with intermediate boxes such as firewalls, relays and NAT interaction with intermediate boxes such as firewalls, relays and NAT
boxes. boxes.
This protocol suite intends to satisfy the security considerations This protocol suite intends to satisfy the security considerations
described in the RTCWEB security documents, described in the WebRTC security documents,
[I-D.ietf-rtcweb-security] and [I-D.ietf-rtcweb-security-arch]. [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 by using the word "browser".
2. Requirements language 2. Requirements language
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [RFC2119]. document are to be interpreted as described in RFC 2119 [RFC2119].
3. Transport and Middlebox specification 3. Transport and Middlebox specification
3.1. System-provided interfaces 3.1. System-provided interfaces
The protocol specifications used here assume that the following The protocol specifications used here assume that the following
protocols are available to the implementations of the RTCWEB protocols are available to the implementations of the WebRTC
protocols: protocols:
o UDP. This is the protocol assumed by most protocol elements o UDP. This is the protocol assumed by most protocol elements
described. described.
o TCP. This is used for HTTP/WebSockets, as well as for TURN/SSL o TCP. This is used for HTTP/WebSockets, as well as for TURN/SSL
and ICE-TCP. and ICE-TCP.
For both protocols, IPv4 and IPv6 support is assumed. For both protocols, IPv4 and IPv6 support is assumed.
For UDP, this specification assumes the ability to set the DSCP code 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 point of the sockets opened on a per-packet basis, in order to
achieve the prioritizations described in [I-D.ietf-tsvwg-rtcweb-qos] achieve the prioritizations described in [I-D.ietf-tsvwg-rtcweb-qos]
(see Section 4.1) when multiple media types are multiplexed. It does (see Section 4.1) when multiple media types are multiplexed. It does
not assume that the DSCP codepoints will be honored, and does assume not assume that the DSCP codepoints will be honored, and does assume
that they may be zeroed or changed, since this is a local that they may be zeroed or changed, since this is a local
configuration issue. configuration issue.
Platforms that do not give access to these interfaces will not be Platforms that do not give access to these interfaces will not be
able to support a conforming RTCWEB implementation. able to support a conforming WebRTC implementation.
This specification does not assume that the implementation will have This specification does not assume that the implementation will have
access to ICMP or raw IP. access to ICMP or raw IP.
3.2. Ability to use IPv4 and IPv6 3.2. Ability to use IPv4 and IPv6
Web applications running on top of the RTCWEB implementation MUST be Web applications running in a WebRTC browser MUST be able to utilize
able to utilize both IPv4 and IPv6 where available - that is, when both IPv4 and IPv6 where available - that is, when two peers have
two peers have only IPv4 connectivity to each other, or they have only IPv4 connectivity to each other, or they have only IPv6
only IPv6 connectivity to each other, applications running on top of connectivity to each other, applications running in the WebRTC
the RTCWEB implementation MUST be able to communicate. browser MUST be able to communicate.
When TURN is used, and the TURN server has IPv4 or IPv6 connectivity 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 to the peer or its TURN server, candidates of the appropriate types
MUST be supported. The "Happy Eyeballs" specification for ICE MUST be supported. The "Happy Eyeballs" specification for ICE
[I-D.reddy-mmusic-ice-happy-eyeballs] SHOULD be supported. [I-D.reddy-mmusic-ice-happy-eyeballs] SHOULD be supported.
3.3. Usage of temporary IPv6 addresses 3.3. Usage of temporary IPv6 addresses
The IPv6 default address selection specification [RFC6724] specifies The IPv6 default address selection specification [RFC6724] specifies
that temporary addresses [RFC4941] are to be preferred over permanent that temporary addresses [RFC4941] are to be preferred over permanent
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ICE [RFC5245] MUST be supported. The implementation MUST be a full ICE [RFC5245] MUST be supported. The implementation MUST be a full
ICE implementation, not ICE-Lite. A full ICE implementation allows ICE implementation, not ICE-Lite. A full ICE implementation allows
interworking with both ICE and ICE-Lite implementations when they are interworking with both ICE and ICE-Lite implementations when they are
deployed appropriately. deployed appropriately.
In order to deal with situations where both parties are behind NATs In order to deal with situations where both parties are behind NATs
of the type that perform endpoint-dependent mapping (as defined in of the type that perform endpoint-dependent mapping (as defined in
[RFC5128] section 2.4), TURN [RFC5766] MUST be supported. [RFC5128] section 2.4), TURN [RFC5766] MUST be supported.
Configuration of STUN and TURN servers, both from browser WebRTC browsers MUST support configuration of STUN and TURN servers,
configuration and from an application, MUST be supported. both from browser configuration and from an application.
In order to deal with firewalls that block all UDP traffic, the mode 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 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 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 client and the server MUST be supported. See [RFC5766] section 2.1
for details. for details.
In order to deal with situations where one party is on an IPv4 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 network and the other party is on an IPv6 network, TURN extensions
for IPv6 [RFC6156] MUST be supported. for IPv6 [RFC6156] MUST be supported.
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applications to communicate to peers with public IP addresses across applications to communicate to peers with public IP addresses across
UDP-blocking firewalls without using a TURN server. UDP-blocking firewalls without using a TURN server.
If TCP connections are used, RTP framing according to [RFC4571] MUST 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 be used, both for the RTP packets and for the DTLS packets used to
carry data channels. carry data channels.
The ALTERNATE-SERVER mechanism specified in [RFC5389] (STUN) section The ALTERNATE-SERVER mechanism specified in [RFC5389] (STUN) section
11 (300 Try Alternate) MUST be supported. 11 (300 Try Alternate) MUST be supported.
Further discussion of the interaction of RTCWEB with firewalls is Further discussion of the interaction of WebRTC with firewalls is
contained in [I-D.hutton-rtcweb-nat-firewall-considerations]. This contained in [I-D.hutton-rtcweb-nat-firewall-considerations]. This
document makes no requirements on interacting with HTTP proxies or document makes no requirements on interacting with HTTP proxies or
HTTP proxy configuration methods. HTTP proxy configuration methods.
NOTE IN DRAFT: This may be added. The WebRTC implementation MAY support accessing the Internet through
an HTTP proxy. If it does so, it MUST support the "connect" header
as specified in [I-D.hutton-httpbis-connect-protocol].
3.5. Transport protocols implemented 3.5. Transport protocols implemented
For transport of media, secure RTP is used. The details of the For transport of media, secure RTP is used. The details of the
profile of RTP used are described in "RTP Usage" profile of RTP used are described in "RTP Usage"
[I-D.ietf-rtcweb-rtp-usage]. Key exchange MUST be done using DTLS- [I-D.ietf-rtcweb-rtp-usage]. Key exchange MUST be done using DTLS-
SRTP, as described in [I-D.ietf-rtcweb-security-arch]. SRTP, as described in [I-D.ietf-rtcweb-security-arch].
For data transport over the RTCWEB data channel For data transport over the WebRTC data channel
[I-D.ietf-rtcweb-data-channel], RTCWEB implementations MUST support [I-D.ietf-rtcweb-data-channel], WebRTC implementations MUST support
SCTP over DTLS over ICE. This encapsulation is specified in SCTP over DTLS over ICE. This encapsulation is specified in
[I-D.ietf-tsvwg-sctp-dtls-encaps]. Negotiation of this transport 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 SDP is defined in [I-D.ietf-mmusic-sctp-sdp]. The SCTP extension for
NDATA, [I-D.ietf-tsvwg-sctp-ndata], MUST be supported. NDATA, [I-D.ietf-tsvwg-sctp-ndata], MUST be supported.
The setup protocol for RTCWEB data channels is described in The setup protocol for WebRTC data channels is described in
[I-D.jesup-rtcweb-data-protocol]. [I-D.jesup-rtcweb-data-protocol].
RTCWEB implementations MUST support multiplexing of DTLS and RTP over WebRTC implementations MUST support multiplexing of DTLS and RTP over
the same port pair, as described in the DTLS_SRTP specification the same port pair, as described in the DTLS_SRTP specification
[RFC5764], section 5.1.2. All application layer protocol payloads [RFC5764], section 5.1.2. All application layer protocol payloads
over this DTLS connection are SCTP packets. over this DTLS connection are SCTP packets.
Protocol identification MUST be supplied as part of the DTLS
handshake, as specified in [I-D.thomson-rtcweb-alpn].
4. Media Prioritization 4. Media Prioritization
The RTCWEB prioritization model is that the application tells the The WebRTC prioritization model is that the application tells the
RTCWEB implementation about the priority of media and data flows WebRTC implementation about the priority of media and data flows
through an API. through an API.
The priority associated with a media or data flow is classified as The priority associated with a media or data flow is classified as
"normal", "below normal", "high" or "very high". There are only four "normal", "below normal", "high" or "very high". There are only four
priority levels at the API. priority levels at the API.
The priority settings affect two pieces of behavior: Packet markings The priority settings affect two pieces of behavior: Packet markings
and packet send sequence decisions. Each is described in its own and packet send sequence decisions. Each is described in its own
section below. section below.
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according to the guidelines in [I-D.ietf-tsvwg-rtcweb-qos]. It is according to the guidelines in [I-D.ietf-tsvwg-rtcweb-qos]. It is
appropriate to depart from this recommendation when running on appropriate to depart from this recommendation when running on
platforms where QoS marking is not implemented. platforms where QoS marking is not implemented.
The implementation MAY turn off use of DSCP markings if it detects The implementation MAY turn off use of DSCP markings if it detects
symptoms of unexpected behaviour like priority inversion or blocking symptoms of unexpected behaviour like priority inversion or blocking
of packets with certain DSCP markings. The detection of these of packets with certain DSCP markings. The detection of these
conditions is implementation dependent. (Question: Does there need conditions is implementation dependent. (Question: Does there need
to be an API knob to turn off DSCP markings?) to be an API knob to turn off DSCP markings?)
All packets arrying data from the SCTP association supporting the
data channels MUST use a single DSCP code point.
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
[I-D.ietf-dart-dscp-rtp].
There exist a number of schemes for achieving quality of service that There exist a number of schemes for achieving quality of service that
do not depend solely on DSCP code points. Some of these schemes do not depend solely on DSCP code points. Some of these schemes
depend on classifying the traffic into flows based on 5-tuple (source depend on classifying the traffic into flows based on 5-tuple (source
address, source port, protocol, destination address, destination address, source port, protocol, destination address, destination
port) or 6-tuple (5-tuple + DSCP code point). Under differing port) or 6-tuple (5-tuple + DSCP code point). Under differing
conditions, it may therefore make sense for a sending application to conditions, it may therefore make sense for a sending application to
choose any of the configurations: choose any of the configurations:
o Each media stream carried on its own 5-tuple o Each media stream carried on its own 5-tuple
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It MAY choose to support other configurations. It MAY choose to support other configurations.
Sending data over multiple 5-tuples is not supported. Sending data over multiple 5-tuples is not supported.
A receiving implementation MUST be able to receive media and data in A receiving implementation MUST be able to receive media and data in
all these configurations. all these configurations.
4.2. Local prioritization 4.2. Local prioritization
When an RTCWEB implementation has packets to send on multiple streams When an WebRTC implementation has packets to send on multiple streams
(with each media stream and each data channel considered as one (with each media stream and each data channel considered as one
"stream" for this purpose) that are congestion-controlled under the "stream" for this purpose) that are congestion-controlled under the
same congestion controller, the RTCWEB implementation SHOULD cause same congestion controller, the WebRTC implementation SHOULD cause
data to be emitted in such a way that each stream at each level of data to be emitted in such a way that each stream at each level of
priority is being given approximately twice the transmission capacity priority is being given approximately twice the transmission capacity
(measured in payload bytes) of the level below. (measured in payload bytes) of the level below.
Thus, when congestion occurs, a "very high" priority flow will have Thus, when congestion occurs, a "very high" priority flow will have
the ability to send 8 times as much data as a "below normal" flow if the ability to send 8 times as much data as a "below normal" flow if
both have data to send. This prioritization is independent of the both have data to send. This prioritization is independent of the
media type. The details of which packet to send first are media type. The details of which packet to send first are
implementation defined. implementation defined.
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from many RTCWEB WG members. from many RTCWEB WG members.
Special thanks for reviews of earlier versions of this draft go to Special thanks for reviews of earlier versions of this draft go to
Eduardo Gueiros, Magnus Westerlund, Markus Isomaki and Dan Wing; the Eduardo Gueiros, Magnus Westerlund, Markus Isomaki and Dan Wing; the
contributions from Andrew Hutton also deserve special mention. contributions from Andrew Hutton also deserve special mention.
8. References 8. References
8.1. Normative References 8.1. Normative References
[I-D.hutton-httpbis-connect-protocol]
Hutton, A., Uberti, J., and M. Thomson, "HTTP Connect -
Tunnel Protocol For WebRTC", draft-hutton-httpbis-connect-
protocol-00 (work in progress), June 2014.
[I-D.ietf-mmusic-sctp-sdp] [I-D.ietf-mmusic-sctp-sdp]
Loreto, S. and G. Camarillo, "Stream Control Transmission Loreto, S. and G. Camarillo, "Stream Control Transmission
Protocol (SCTP)-Based Media Transport in the Session Protocol (SCTP)-Based Media Transport in the Session
Description Protocol (SDP)", draft-ietf-mmusic-sctp-sdp-06 Description Protocol (SDP)", draft-ietf-mmusic-sctp-sdp-07
(work in progress), February 2014. (work in progress), July 2014.
[I-D.ietf-rtcweb-data-channel] [I-D.ietf-rtcweb-data-channel]
Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data
Channels", draft-ietf-rtcweb-data-channel-08 (work in Channels", draft-ietf-rtcweb-data-channel-11 (work in
progress), April 2014. progress), July 2014.
[I-D.ietf-rtcweb-rtp-usage] [I-D.ietf-rtcweb-rtp-usage]
Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
Communication (WebRTC): Media Transport and Use of RTP", Communication (WebRTC): Media Transport and Use of RTP",
draft-ietf-rtcweb-rtp-usage-13 (work in progress), April draft-ietf-rtcweb-rtp-usage-16 (work in progress), July
2014. 2014.
[I-D.ietf-rtcweb-security] [I-D.ietf-rtcweb-security]
Rescorla, E., "Security Considerations for WebRTC", draft- Rescorla, E., "Security Considerations for WebRTC", draft-
ietf-rtcweb-security-06 (work in progress), January 2014. ietf-rtcweb-security-07 (work in progress), July 2014.
[I-D.ietf-rtcweb-security-arch] [I-D.ietf-rtcweb-security-arch]
Rescorla, E., "WebRTC Security Architecture", draft-ietf- Rescorla, E., "WebRTC Security Architecture", draft-ietf-
rtcweb-security-arch-09 (work in progress), February 2014. rtcweb-security-arch-10 (work in progress), July 2014.
[I-D.ietf-tsvwg-rtcweb-qos] [I-D.ietf-tsvwg-rtcweb-qos]
Dhesikan, S., Druta, D., Jones, P., and J. Polk, "DSCP and Dhesikan, S., Jennings, C., Druta, D., Jones, P., and J.
other packet markings for RTCWeb QoS", draft-ietf-tsvwg- Polk, "DSCP and other packet markings for RTCWeb QoS",
rtcweb-qos-00 (work in progress), April 2014. draft-ietf-tsvwg-rtcweb-qos-02 (work in progress), June
2014.
[I-D.ietf-tsvwg-sctp-dtls-encaps] [I-D.ietf-tsvwg-sctp-dtls-encaps]
Tuexen, M., Stewart, R., Jesup, R., and S. Loreto, "DTLS Tuexen, M., Stewart, R., Jesup, R., and S. Loreto, "DTLS
Encapsulation of SCTP Packets", draft-ietf-tsvwg-sctp- Encapsulation of SCTP Packets", draft-ietf-tsvwg-sctp-
dtls-encaps-03 (work in progress), February 2014. dtls-encaps-05 (work in progress), July 2014.
[I-D.ietf-tsvwg-sctp-ndata] [I-D.ietf-tsvwg-sctp-ndata]
Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann, "A Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann,
New Data Chunk for Stream Control Transmission Protocol", "Stream Schedulers and a New Data Chunk for the Stream
draft-ietf-tsvwg-sctp-ndata-00 (work in progress), Control Transmission Protocol", draft-ietf-tsvwg-sctp-
February 2014. ndata-01 (work in progress), July 2014.
[I-D.reddy-mmusic-ice-happy-eyeballs] [I-D.reddy-mmusic-ice-happy-eyeballs]
Reddy, T., Patil, P., and P. Martinsen, "Happy Eyeballs Reddy, T., Patil, P., and P. Martinsen, "Happy Eyeballs
Extension for ICE", draft-reddy-mmusic-ice-happy- Extension for ICE", draft-reddy-mmusic-ice-happy-
eyeballs-06 (work in progress), February 2014. eyeballs-07 (work in progress), June 2014.
[I-D.thomson-rtcweb-alpn]
Thomson, M., "Application Layer Protocol Negotiation for
Web Real-Time Communications (WebRTC)", draft-thomson-
rtcweb-alpn-00 (work in progress), April 2014.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997. Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC4571] Lazzaro, J., "Framing Real-time Transport Protocol (RTP) [RFC4571] Lazzaro, J., "Framing Real-time Transport Protocol (RTP)
and RTP Control Protocol (RTCP) Packets over Connection- and RTP Control Protocol (RTCP) Packets over Connection-
Oriented Transport", RFC 4571, July 2006. Oriented Transport", RFC 4571, July 2006.
[RFC4941] Narten, T., Draves, R., and S. Krishnan, "Privacy [RFC4941] Narten, T., Draves, R., and S. Krishnan, "Privacy
Extensions for Stateless Address Autoconfiguration in Extensions for Stateless Address Autoconfiguration in
skipping to change at page 11, line 21 skipping to change at page 12, line 5
(IPv6)", RFC 6724, September 2012. (IPv6)", RFC 6724, September 2012.
8.2. Informative References 8.2. Informative References
[I-D.hutton-rtcweb-nat-firewall-considerations] [I-D.hutton-rtcweb-nat-firewall-considerations]
Stach, T., Hutton, A., and J. Uberti, "RTCWEB Stach, T., Hutton, A., and J. Uberti, "RTCWEB
Considerations for NATs, Firewalls and HTTP proxies", Considerations for NATs, Firewalls and HTTP proxies",
draft-hutton-rtcweb-nat-firewall-considerations-03 (work draft-hutton-rtcweb-nat-firewall-considerations-03 (work
in progress), January 2014. in progress), January 2014.
[I-D.ietf-dart-dscp-rtp]
Black, D. and P. Jones, "Differentiated Services
(DiffServ) and Real-time Communication", draft-ietf-dart-
dscp-rtp-02 (work in progress), August 2014.
[I-D.ietf-rtcweb-overview] [I-D.ietf-rtcweb-overview]
Alvestrand, H., "Overview: Real Time Protocols for Brower- Alvestrand, H., "Overview: Real Time Protocols for
based Applications", draft-ietf-rtcweb-overview-09 (work Browser-based Applications", draft-ietf-rtcweb-overview-10
in progress), February 2014. (work in progress), June 2014.
[I-D.jesup-rtcweb-data-protocol] [I-D.jesup-rtcweb-data-protocol]
Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel
Protocol", draft-jesup-rtcweb-data-protocol-04 (work in Protocol", draft-jesup-rtcweb-data-protocol-04 (work in
progress), February 2013. progress), February 2013.
[RFC3484] Draves, R., "Default Address Selection for Internet [RFC3484] Draves, R., "Default Address Selection for Internet
Protocol version 6 (IPv6)", RFC 3484, February 2003. Protocol version 6 (IPv6)", RFC 3484, February 2003.
[RFC5014] Nordmark, E., Chakrabarti, S., and J. Laganier, "IPv6 [RFC5014] Nordmark, E., Chakrabarti, S., and J. Laganier, "IPv6
skipping to change at page 13, line 6 skipping to change at page 13, line 44
o Added RFC 4571 reference for framing RTP packets over TCP. o Added RFC 4571 reference for framing RTP packets over TCP.
o Downgraded TURN TCP candidates from SHOULD to MAY, and added more o Downgraded TURN TCP candidates from SHOULD to MAY, and added more
language discussing TCP usage. language discussing TCP usage.
o Added language on IPv6 temporary addresses. o Added language on IPv6 temporary addresses.
o Added language describing multiplexing choices. o Added language describing multiplexing choices.
o Added a separate section detailing what it means when we say that o Added a separate section detailing what it means when we say that
an RTCWEB implementation MUST support both IPv4 and IPv6. an WebRTC implementation MUST support both IPv4 and IPv6.
A.4. Changes from -03 to -04 A.4. Changes from -03 to -04
o Added a section on prioritization, moved the DSCP section into it, o Added a section on prioritization, moved the DSCP section into it,
and added a section on local prioritization, giving a specific and added a section on local prioritization, giving a specific
algorithm for interpreting "priority" in local prioritization. algorithm for interpreting "priority" in local prioritization.
o ICE-TCP candidates was changed from MAY to MUST, in recognition of o ICE-TCP candidates was changed from MAY to MUST, in recognition of
the sense of the room at the London IETF. the sense of the room at the London IETF.
skipping to change at page 13, line 32 skipping to change at page 14, line 23
RTCWEB. RTCWEB.
o Addressed a number of clarity / language comments o Addressed a number of clarity / language comments
o Rewrote the prioritization to cover data channels and to describe o Rewrote the prioritization to cover data channels and to describe
multiple ways of prioritizing flows multiple ways of prioritizing flows
o Made explicit reference to "MUST do DTLS-SRTP", and referred to o Made explicit reference to "MUST do DTLS-SRTP", and referred to
security-arch for details 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
codepoint
Author's Address Author's Address
Harald Alvestrand Harald Alvestrand
Google Google
Email: harald@alvestrand.no Email: harald@alvestrand.no
 End of changes. 39 change blocks. 
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