draft-ietf-rtcweb-transports-03.txt   draft-ietf-rtcweb-transports-04.txt 
Network Working Group H. Alvestrand Network Working Group H. Alvestrand
Internet-Draft Google Internet-Draft Google
Intended status: Standards Track March 31, 2014 Intended status: Standards Track April 25, 2014
Expires: October 2, 2014 Expires: October 27, 2014
Transports for RTCWEB Transports for RTCWEB
draft-ietf-rtcweb-transports-03 draft-ietf-rtcweb-transports-04
Abstract Abstract
This document describes the data transport protocols used by RTCWEB, This document describes the data transport protocols used by RTCWEB,
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
skipping to change at page 1, line 32 skipping to change at page 1, line 32
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 October 2, 2014. This Internet-Draft will expire on October 27, 2014.
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
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described in the Simplified BSD License. described in the Simplified BSD License.
Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
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 . . . . . . . . . . . . . . . 4 3.2. Ability to use IPv4 and IPv6 . . . . . . . . . . . . . . . 4
3.3. Usage of temporary IPv6 addresses . . . . . . . . . . . . 4 3.3. Usage of temporary IPv6 addresses . . . . . . . . . . . . 4
3.4. Usage of Quality of Service - DSCP and Multiplexing . . . 4 3.4. Middle box related functions . . . . . . . . . . . . . . . 4
3.5. Middle box related functions . . . . . . . . . . . . . . . 5 3.5. Transport protocols implemented . . . . . . . . . . . . . 5
3.6. Transport protocols implemented . . . . . . . . . . . . . 6 4. Media Prioritization . . . . . . . . . . . . . . . . . . . . . 6
4. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 7 4.1. Usage of Quality of Service - DSCP and Multiplexing . . . 6
5. Security Considerations . . . . . . . . . . . . . . . . . . . 7 4.2. Local prioritization . . . . . . . . . . . . . . . . . . . 7
6. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 7 5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 8
7. References . . . . . . . . . . . . . . . . . . . . . . . . . . 7 6. Security Considerations . . . . . . . . . . . . . . . . . . . 8
7.1. Normative References . . . . . . . . . . . . . . . . . . . 7 7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 8
7.2. Informative References . . . . . . . . . . . . . . . . . . 9 8. References . . . . . . . . . . . . . . . . . . . . . . . . . . 9
Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 10 8.1. Normative References . . . . . . . . . . . . . . . . . . . 9
A.1. Changes from -00 to -01 . . . . . . . . . . . . . . . . . 10 8.2. Informative References . . . . . . . . . . . . . . . . . . 11
A.2. Changes from -01 to -02 . . . . . . . . . . . . . . . . . 10 Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 11
A.3. Changes from -02 to -03 . . . . . . . . . . . . . . . . . 11 A.1. Changes from -00 to -01 . . . . . . . . . . . . . . . . . 11
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 11 A.2. Changes from -01 to -02 . . . . . . . . . . . . . . . . . 12
A.3. Changes from -02 to -03 . . . . . . . . . . . . . . . . . 12
A.4. Changes from -03 to -04 . . . . . . . . . . . . . . . . . 13
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 13
1. Introduction 1. Introduction
The IETF RTCWEB effort, part of the WebRTC effort carried out in The IETF RTCWEB effort, part of the WebRTC effort carried out in
cooperation between the IETF and the W3C, is aimed at specifying a cooperation between the IETF and the W3C, is aimed at specifying a
protocol suite that is useful for real time multimedia exchange protocol suite that is useful for real time multimedia exchange
between browsers. between browsers.
The overall effort is described in the RTCWEB overview document, The overall effort is described in the RTCWEB overview document,
[I-D.ietf-rtcweb-overview]. This document focuses on the data [I-D.ietf-rtcweb-overview]. This document focuses on the data
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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 achieve the prioritizations described in [I-D.ietf-tsvwg-rtcweb-qos]
[I-D.dhesikan-tsvwg-rtcweb-qos] (see Section 3.4) when multiple media (see Section 4.1) when multiple media types are multiplexed. It does
types are multiplexed. It does not assume that the DSCP codepoints not assume that the DSCP codepoints will be honored, and does assume
will be honored, and does assume that they may be zeroed or changed, that they may be zeroed or changed, since this is a local
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 RTCWEB 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 on top of the RTCWEB implementation MUST be
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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
addresses. This is a change from the rules specified by [RFC3484]. addresses. This is a change from the rules specified by [RFC3484].
For applications that select a single address, this is usually done For applications that select a single address, this is usually done
by the IPV6_PREFER_SRC_TMP specified in [RFC5014]. However, this by the IPV6_PREFER_SRC_TMP preference flag specified in [RFC5014].
rule is not completely obvious in the ICE scope. This is therefore However, this rule is not completely obvious in the ICE scope. This
clarified as follows: is therefore clarified as follows:
When a client gathers all IPv6 addresses on a host, and both When a client gathers all IPv6 addresses on a host, and both
temporary addresses and permanent addresses of the same scope are temporary addresses and permanent addresses of the same scope are
present, the client SHOULD discard the permanent addresses before present, the client SHOULD discard the permanent addresses before
forming pairs. This is consistent with the default policy described forming pairs. This is consistent with the default policy described
in [RFC6724]. in [RFC6724].
3.4. Usage of Quality of Service - DSCP and Multiplexing 3.4. Middle box related functions
WebRTC implementations SHOULD attempt to set QoS on the packets sent,
according to the guidelines in [I-D.dhesikan-tsvwg-rtcweb-qos]. It
is appropriate to depart from this recommendation when running on
platforms where QoS marking is not implemented.
There exist a number of schemes for achieving quality of service that
do not depend solely on DSCP code points. Some of these schemes
depend on classifying the traffic into flows based on 5-tuple (source
address, source port, protocol, destination address, destination
port) or 6-tuple (same as above + 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
flows.
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.
A sending implementation MUST be able to multiplex all media and data
on a single 5-tuple (fully bundled), MUST be able to send each media
stream and data on their own 5-tuple (fully unbundled), and MAY
choose to support other configurations.
NOTE IN DRAFT: is there a need to place the "group by media type,
with data multiplexed on the video" as a MUST or SHOULD
configuration?
A receiving implementation MUST be able to receive media and data in
all these configurations.
3.5. Middle box related functions
The primary mechanism to deal with middle boxes is ICE, which is an The primary mechanism to deal with middle boxes is ICE, which is an
appropriate way to deal with NAT boxes and firewalls that accept appropriate way to deal with NAT boxes and firewalls that accept
traffic from the inside, but only from the outside if it's in traffic from the inside, but only from the outside if it's in
response to inside traffic (simple stateful firewalls). response to inside traffic (simple stateful firewalls).
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. ICE implementation, not ICE-Lite; this 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 In order to deal with situations where both parties are behind NATs
which perform endpoint-dependent mapping (as defined in [RFC5128] which perform endpoint-dependent mapping (as defined in [RFC5128]
section 2.4), TURN [RFC5766] MUST be supported. section 2.4), TURN [RFC5766] MUST be supported.
Configuration of STUN and TURN servers, both from browser Configuration of STUN and TURN servers, both from browser
configuration and from an applicaiton, MUST be supported. configuration and from an applicaiton, MUST be supported.
In order to deal with firewalls that block all UDP traffic, TURN In order to deal with firewalls that block all UDP traffic, TURN
using TCP between the client and the server MUST be supported, and using TCP between the client and the server MUST be supported, and
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However, such candidates are not seen as providing any significant However, such candidates are not seen as providing any significant
benefit. First, use of TURN TCP would only be relevant in cases benefit. First, use of TURN TCP would only be relevant in cases
which both peers are required to use TCP to establish a which both peers are required to use TCP to establish a
PeerConnection. Secondly, that use case is anyway supported by both PeerConnection. Secondly, that use case is anyway supported by both
sides establishing UDP relay candidates using TURN over TCP to sides establishing UDP relay candidates using TURN over TCP to
connect to the relay server. Thirdly, using TCP only between the connect to the relay server. Thirdly, using TCP only between the
endpoint and its relay may result in less issues with TCP in regards 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. to real-time constraints, e.g. due to head of line blocking.
ICE-TCP candidates [RFC6544] MAY be supported; this may allow ICE-TCP candidates [RFC6544] MUST be supported; this may allow
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 RTCWEB 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. NOTE IN DRAFT: This may be added.
3.6. 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]. [I-D.ietf-rtcweb-rtp-usage].
For data transport over the RTCWEB data channel For data transport over the RTCWEB data channel
[I-D.ietf-rtcweb-data-channel], RTCWEB implementations MUST support [I-D.ietf-rtcweb-data-channel], RTCWEB 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 RTCWEB data channels is described in
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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 RTCWEB 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 RTCWEB 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.
4. IANA Considerations 4. Media Prioritization
The RTCWEB prioritization model is that the application tells the
RTCWEB implementation about the priority of media and data flows
through an API.
The priority associated with a media or data flow is classified as
"normal", "below normal", "high" or "very high". There are only four
priority levels at the API.
The priority settings affect two pieces of behavior: Packet markings
and packet send sequence decisions. Each is described in its own
section below.
4.1. Usage of Quality of Service - DSCP and Multiplexing
WebRTC 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. (Question: Does there need
to be an API knob to turn off DSCP markings?)
There exist a number of schemes for achieving quality of service that
do not depend solely on DSCP code points. Some of these schemes
depend on classifying the traffic into flows based on 5-tuple (source
address, source port, protocol, destination address, destination
port) or 6-tuple (same as above + 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
flows.
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
[I-D.ietf-rtcweb-rtp-usage].
A sending implementation MUST be able to multiplex all media and data
on a single 5-tuple (fully bundled), MUST be able to send each media
stream on its own 5-tuple and data on its own 5-tuple (fully
unbundled), and MAY choose to support other configurations.
Sending data over multiple 5-tuples is not supported.
NOTE IN DRAFT: is there a need to place the "group by media type,
with data multiplexed on the video" as a MUST or SHOULD
configuration? Are there other MUST configurations?
NOTE IN DRAFT: It's been suggested that at least one "MUST"
configuration should be with data channels on its own 5-tuple,
separate from the media. Opinions sought.
A receiving implementation MUST be able to receive media and data in
all these configurations.
4.2. Local prioritization
When an RTCWEB implementation has packets to send on multiple streams
that are congestion-controlled under the same congestion controller,
the RTCWEB implementation SHOULD serve the streams in a weighted
round-robin fashion, with each stream at each level of priority being
given approximately twice the transmission capacity (measured in
payload bytes) of the level below.
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
both have data to send. This prioritization is independent of the
media type, but will lead to packet loss due to full send buffers
occuring first on the highest volume flows at any given priority
level. The details of which packet to send first are implementation
defined.
For example: If there is a very high priority audio flow sending 100
byte packets, and a normal 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 decisions.
Conversely, if the audio flow is marked normal priority and the video
flow is marked very 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 very high priority audio flows, each will be able to
send 4000 bytes in the same period where a normal priority video flow
is able to send 1000 bytes.
NOTE: The appropriate algorithm for deciding when to send SCTP data
vs media data is not described yet.
5. IANA Considerations
This document makes no request of IANA. This document makes no request of IANA.
Note to RFC Editor: this section may be removed on publication as an Note to RFC Editor: this section may be removed on publication as an
RFC. RFC.
5. Security Considerations 6. Security Considerations
Security considerations are enumerated in [I-D.ietf-rtcweb-security]. Security considerations are enumerated in [I-D.ietf-rtcweb-security].
6. Acknowledgements 7. Acknowledgements
This document is based on earlier versions embedded in This document is based on earlier versions embedded in
[I-D.ietf-rtcweb-overview], which were the results of contributions [I-D.ietf-rtcweb-overview], which were the results of contributions
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
Magnus Westerlund, Markus Isomaki and Dan Wing; the contributions Magnus Westerlund, Markus Isomaki and Dan Wing; the contributions
from Andrew Hutton also deserve special mention. from Andrew Hutton also deserve special mention.
7. References 8. References
7.1. Normative References
[I-D.dhesikan-tsvwg-rtcweb-qos] 8.1. Normative References
Dhesikan, S., Druta, D., Jones, P., and J. Polk, "DSCP and
other packet markings for RTCWeb QoS",
draft-dhesikan-tsvwg-rtcweb-qos-06 (work in progress),
March 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-06
(work in progress), February 2014. (work in progress), February 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-07 (work in Channels", draft-ietf-rtcweb-data-channel-08 (work in
progress), February 2014. progress), April 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-12 (work in progress), draft-ietf-rtcweb-rtp-usage-13 (work in progress),
February 2014. April 2014.
[I-D.ietf-rtcweb-security] [I-D.ietf-rtcweb-security]
Rescorla, E., "Security Considerations for WebRTC", Rescorla, E., "Security Considerations for WebRTC",
draft-ietf-rtcweb-security-06 (work in progress), draft-ietf-rtcweb-security-06 (work in progress),
January 2014. January 2014.
[I-D.ietf-rtcweb-security-arch] [I-D.ietf-rtcweb-security-arch]
Rescorla, E., "WebRTC Security Architecture", Rescorla, E., "WebRTC Security Architecture",
draft-ietf-rtcweb-security-arch-09 (work in progress), draft-ietf-rtcweb-security-arch-09 (work in progress),
February 2014. February 2014.
[I-D.ietf-tsvwg-rtcweb-qos]
Dhesikan, S., Druta, D., Jones, P., and J. Polk, "DSCP and
other packet markings for RTCWeb QoS",
draft-ietf-tsvwg-rtcweb-qos-00 (work in progress),
April 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", Encapsulation of SCTP Packets",
draft-ietf-tsvwg-sctp-dtls-encaps-03 (work in progress), draft-ietf-tsvwg-sctp-dtls-encaps-03 (work in progress),
February 2014. February 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, "A
New Data Chunk for Stream Control Transmission Protocol", New Data Chunk for Stream Control Transmission Protocol",
draft-ietf-tsvwg-sctp-ndata-00 (work in progress), draft-ietf-tsvwg-sctp-ndata-00 (work in progress),
skipping to change at page 9, line 42 skipping to change at page 11, line 13
RFC 6156, April 2011. RFC 6156, April 2011.
[RFC6544] Rosenberg, J., Keranen, A., Lowekamp, B., and A. Roach, [RFC6544] Rosenberg, J., Keranen, A., Lowekamp, B., and A. Roach,
"TCP Candidates with Interactive Connectivity "TCP Candidates with Interactive Connectivity
Establishment (ICE)", RFC 6544, March 2012. Establishment (ICE)", RFC 6544, March 2012.
[RFC6724] Thaler, D., Draves, R., Matsumoto, A., and T. Chown, [RFC6724] Thaler, D., Draves, R., Matsumoto, A., and T. Chown,
"Default Address Selection for Internet Protocol Version 6 "Default Address Selection for Internet Protocol Version 6
(IPv6)", RFC 6724, September 2012. (IPv6)", RFC 6724, September 2012.
7.2. 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-rtcweb-overview] [I-D.ietf-rtcweb-overview]
Alvestrand, H., "Overview: Real Time Protocols for Brower- Alvestrand, H., "Overview: Real Time Protocols for Brower-
based Applications", draft-ietf-rtcweb-overview-09 (work based Applications", draft-ietf-rtcweb-overview-09 (work
skipping to change at page 11, line 36 skipping to change at page 13, line 8
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 RTCWEB 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.
Author's Address Author's Address
Harald Alvestrand Harald Alvestrand
Google Google
Email: harald@alvestrand.no Email: harald@alvestrand.no
 End of changes. 21 change blocks. 
89 lines changed or deleted 168 lines changed or added

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