draft-ietf-rtcweb-transports-06.txt   draft-ietf-rtcweb-transports-07.txt 
Network Working Group H. Alvestrand Network Working Group H. Alvestrand
Internet-Draft Google Internet-Draft Google
Intended status: Standards Track August 11, 2014 Intended status: Standards Track October 22, 2014
Expires: February 12, 2015 Expires: April 25, 2015
Transports for WebRTC Transports for WebRTC
draft-ietf-rtcweb-transports-06 draft-ietf-rtcweb-transports-07
Abstract Abstract
This document describes the data transport protocols used by WebRTC, 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
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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 February 12, 2015. This Internet-Draft will expire on April 25, 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
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Table of Contents Table of Contents
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 . . . . . . . . . . . . . 6
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 . . . . . . . . . . . . . . . . . . 8 4.2. Local prioritization . . . . . . . . . . . . . . . . . . 8
5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 9 5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 9
6. Security Considerations . . . . . . . . . . . . . . . . . . . 9 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 . . . . . . . . . . . . . . . . . 12
Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 12 Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 12
A.1. Changes from -00 to -01 . . . . . . . . . . . . . . . . . 12 A.1. Changes from -00 to -01 . . . . . . . . . . . . . . . . . 12
A.2. Changes from -01 to -02 . . . . . . . . . . . . . . . . . 13 A.2. Changes from -01 to -02 . . . . . . . . . . . . . . . . . 13
A.3. Changes from -02 to -03 . . . . . . . . . . . . . . . . . 13 A.3. Changes from -02 to -03 . . . . . . . . . . . . . . . . . 13
A.4. Changes from -03 to -04 . . . . . . . . . . . . . . . . . 13 A.4. Changes from -03 to -04 . . . . . . . . . . . . . . . . . 14
A.5. Changes from -04 to -05 . . . . . . . . . . . . . . . . . 14 A.5. Changes from -04 to -05 . . . . . . . . . . . . . . . . . 14
A.6. Changes from -05 to -06 . . . . . . . . . . . . . . . . . 14 A.6. Changes from -05 to -06 . . . . . . . . . . . . . . . . . 14
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 14 A.7. Changes from -06 to -07 . . . . . . . . . . . . . . . . . 14
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 15
1. Introduction 1. Introduction
WebRTC 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.
WebRTC is described in the WebRTC overview document, WebRTC is described in the WebRTC overview document,
[I-D.ietf-rtcweb-overview], which also defines terminology used in [I-D.ietf-rtcweb-overview], which also defines terminology used in
this document. 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 WebRTC 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 This document describes requirements that apply to all WebRTC
devices. When there are requirements that apply only to WebRTC devices. When there are requirements that apply only to WebRTC User
browsers, this is called out by using the word "browser". Agents (also called browsers) , this is called out.
The form "WebRTC endpoint" is used as a synonym for "WebRTC device"
in contexts where other text talks about endpoints.
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 WebRTC protocols are available to the WebRTC devices:
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
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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 in a WebRTC browser MUST be able to utilize Web applications running in a WebRTC browser MUST be able to utilize
both IPv4 and IPv6 where available - that is, when two peers have both IPv4 and IPv6 where available - that is, when two peers have
only IPv4 connectivity to each other, or they have only IPv6 only IPv4 connectivity to each other, or they have only IPv6
connectivity to each other, applications running in the WebRTC connectivity to each other, applications running in the WebRTC
browser MUST be able to communicate. browser MUST be able to communicate.
WebRTC devices, when attached to networks with appropriate protocol
support MUST also be able to communicate using IPv6 and IPv4.
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
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 preference flag specified in [RFC5014]. by the IPV6_PREFER_SRC_TMP preference flag specified in [RFC5014].
However, this rule is not completely obvious in the ICE scope. This However, this rule is not completely obvious in the ICE scope. This
is therefore clarified as follows: is therefore clarified as follows:
When a client gathers all IPv6 addresses on a host, and both When a WebRTC endpoint 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. Middle box related functions 3.4. Middle box related functions
Except when called out, all requirements in this section apply to all
WebRTC devices.
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 is in traffic from the inside, but only from the outside if it is 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 WebRTC endpoints MUST support ICE [RFC5245]. The implementation MUST
ICE implementation, not ICE-Lite. A full ICE implementation allows be a full ICE implementation, not ICE-Lite. A full ICE
interworking with both ICE and ICE-Lite implementations when they are implementation allows interworking with both ICE and ICE-Lite
deployed appropriately. 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
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), WebRTC endpoints MUST support TURN [RFC5766].
WebRTC browsers MUST support configuration of STUN and TURN servers, WebRTC browsers MUST support configuration of STUN and TURN servers,
both from browser configuration and from an application. 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.
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their respective relay servers. their respective relay servers.
Third, using TCP only between the endpoint and its relay may result 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 in less issues with TCP in regards to real-time constraints, e.g. due
to head of line blocking. to head of line blocking.
ICE-TCP candidates [RFC6544] MUST 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 ICE-TCP connections are used, RTP framing according to [RFC4571]
be used, both for the RTP packets and for the DTLS packets used to MUST be used for all content that doesn't have its own framing
carry data channels. mechanism.
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 WebRTC with firewalls is In order to deal with the scenario in which the media must traverse a
contained in [I-D.hutton-rtcweb-nat-firewall-considerations]. This HTTP Proxy, WebRTC browser MUST support the HTTP CONNECT request
document makes no requirements on interacting with HTTP proxies or (Section 4.3.6 of [RFC7231]). WebRTC devices SHOULD support this
HTTP proxy configuration methods. request.
The WebRTC implementation MAY support accessing the Internet through The HTTP Proxy may require authentication and therefore, if HTTP
an HTTP proxy. If it does so, it MUST support the "connect" header CONNECT request is supported, proxy authentication as described in
as specified in [I-D.hutton-httpbis-connect-protocol]. Section 4.3.6 of [RFC7231] and [RFC7235] MUST also be supported.
In addition, the HTTP CONNECT MUST include an indication of the
protocol being used with the HTTP CONNECT initiated tunnel as
described in [I-D.ietf-httpbis-tunnel-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 WebRTC data channel For data transport over the WebRTC data channel
[I-D.ietf-rtcweb-data-channel], WebRTC implementations MUST support [I-D.ietf-rtcweb-data-channel], WebRTC endpoints MUST support SCTP
SCTP over DTLS over ICE. This encapsulation is specified in 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 WebRTC 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].
WebRTC implementations MUST support multiplexing of DTLS and RTP over WebRTC devices MUST support multiplexing of DTLS and RTP over the
the same port pair, as described in the DTLS_SRTP specification 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 Protocol identification MUST be supplied as part of the DTLS
handshake, as specified in [I-D.thomson-rtcweb-alpn]. handshake, as specified in [I-D.thomson-rtcweb-alpn].
4. Media Prioritization 4. Media Prioritization
The WebRTC prioritization model is that the application tells the The WebRTC prioritization model is that the application tells the
WebRTC implementation about the priority of media and data flows WebRTC browser about the priority of media and data flows through an
through an API. 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.
4.1. Usage of Quality of Service - DSCP and Multiplexing 4.1. Usage of Quality of Service - DSCP and Multiplexing
Implementations SHOULD attempt to set QoS on the packets sent, WebRTC endpoints SHOULD attempt to set QoS on the packets sent,
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 WebRTC endpoint 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 All packets carrying data from the SCTP association supporting the
data channels MUST use a single DSCP code point. data channels MUST use a single DSCP code point.
All packets on one TCP connection, no matter what it carries, MUST All packets on one TCP connection, no matter what it carries, MUST
use a single DSCP code point. use a single DSCP code point.
More advice on the use of DSCP code points with RTP is given in More advice on the use of DSCP code points with RTP is given in
[I-D.ietf-dart-dscp-rtp]. [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
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In each of the configurations mentioned, data channels may be carried In each of the configurations mentioned, data channels may be carried
in its own 5-tuple, or multiplexed together with one of the media in its own 5-tuple, or multiplexed together with one of the media
flows. flows.
More complex configurations, such as sending a high priority video More complex configurations, such as sending a high priority video
stream on one 5-tuple and sending all other video streams multiplexed stream on one 5-tuple and sending all other video streams multiplexed
together over another 5-tuple, can also be envisioned. More together over another 5-tuple, can also be envisioned. More
information on mapping media flows to 5-tuples can be found in information on mapping media flows to 5-tuples can be found in
[I-D.ietf-rtcweb-rtp-usage]. [I-D.ietf-rtcweb-rtp-usage].
A sending implementation MUST be able to support the following A sending WebRTC endpoint MUST be able to support the following
configurations: configurations:
o multiplex all media and data on a single 5-tuple (fully bundled) 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 o send each media stream on its own 5-tuple and data on its own
5-tuple (fully unbundled) 5-tuple (fully unbundled)
o bundle each media type (audio, video or data) into its own 5-tuple o bundle each media type (audio, video or data) into its own 5-tuple
(bundling by media type) (bundling by media type)
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.
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o send each media stream on its own 5-tuple and data on its own o send each media stream on its own 5-tuple and data on its own
5-tuple (fully unbundled) 5-tuple (fully unbundled)
o bundle each media type (audio, video or data) into its own 5-tuple o bundle each media type (audio, video or data) into its own 5-tuple
(bundling by media type) (bundling by media type)
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 WebRTC endpoint 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 WebRTC implementation has packets to send on multiple streams When an WebRTC endpoint has packets to send on multiple streams (with
(with each media stream and each data channel considered as one each media stream and each data channel considered as one "stream"
"stream" for this purpose) that are congestion-controlled under the for this purpose) that are congestion-controlled under the same
same congestion controller, the WebRTC implementation SHOULD cause congestion controller, the WebRTC endpoint SHOULD cause data to be
data to be emitted in such a way that each stream at each level of emitted in such a way that each stream at each level of priority is
priority is being given approximately twice the transmission capacity being given approximately twice the transmission capacity (measured
(measured in payload bytes) of the level below. 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.
For example: If there is a very high priority audio flow sending 100 For example: If there is a very high priority audio flow sending 100
byte packets, and a normal priority video flow sending 1000 byte byte packets, and a normal priority video flow sending 1000 byte
packets, and outgoing capacity exists for sending >5000 payload packets, and outgoing capacity exists for sending >5000 payload
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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] [I-D.ietf-httpbis-tunnel-protocol]
Hutton, A., Uberti, J., and M. Thomson, "HTTP Connect - Hutton, A., Uberti, J., and M. Thomson, "The Tunnel-
Tunnel Protocol For WebRTC", draft-hutton-httpbis-connect- Protocol HTTP Request Header Field", draft-ietf-httpbis-
protocol-00 (work in progress), June 2014. tunnel-protocol-00 (work in progress), August 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-07 Description Protocol (SDP)", draft-ietf-mmusic-sctp-sdp-07
(work in progress), July 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-11 (work in Channels", draft-ietf-rtcweb-data-channel-11 (work in
progress), July 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-16 (work in progress), July draft-ietf-rtcweb-rtp-usage-15 (work in progress), May
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-07 (work in progress), July 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-10 (work in progress), July 2014. rtcweb-security-arch-10 (work in progress), July 2014.
skipping to change at page 11, line 46 skipping to change at page 12, line 5
6156, April 2011. 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.
8.2. Informative References [RFC7231] Fielding, R. and J. Reschke, "Hypertext Transfer Protocol
(HTTP/1.1): Semantics and Content", RFC 7231, June 2014.
[I-D.hutton-rtcweb-nat-firewall-considerations] [RFC7235] Fielding, R. and J. Reschke, "Hypertext Transfer Protocol
Stach, T., Hutton, A., and J. Uberti, "RTCWEB (HTTP/1.1): Authentication", RFC 7235, June 2014.
Considerations for NATs, Firewalls and HTTP proxies",
draft-hutton-rtcweb-nat-firewall-considerations-03 (work 8.2. Informative References
in progress), January 2014.
[I-D.ietf-dart-dscp-rtp] [I-D.ietf-dart-dscp-rtp]
Black, D. and P. Jones, "Differentiated Services Black, D. and P. Jones, "Differentiated Services
(DiffServ) and Real-time Communication", draft-ietf-dart- (DiffServ) and Real-time Communication", draft-ietf-dart-
dscp-rtp-02 (work in progress), August 2014. dscp-rtp-08 (work in progress), October 2014.
[I-D.ietf-rtcweb-overview] [I-D.ietf-rtcweb-overview]
Alvestrand, H., "Overview: Real Time Protocols for Alvestrand, H., "Overview: Real Time Protocols for
Browser-based Applications", draft-ietf-rtcweb-overview-10 Browser-based Applications", draft-ietf-rtcweb-overview-10
(work in progress), June 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.
skipping to change at page 14, line 38 skipping to change at page 14, line 44
o Added reference to the httpbis "connect" protocol (being adopted o Added reference to the httpbis "connect" protocol (being adopted
by HTTPBIS) by HTTPBIS)
o Added reference to the ALPN header (being adopted by RTCWEB) o Added reference to the ALPN header (being adopted by RTCWEB)
o Added reference to the DART RTP document o Added reference to the DART RTP document
o Said explicitly that SCTP for data channels has a single DSCP o Said explicitly that SCTP for data channels has a single DSCP
codepoint codepoint
A.7. Changes from -06 to -07
o Updated terminology in accordance with -overview. Got rid of all
occurences of "WebRTC implementation".
o Modified description of ICE-TCP encapsulation in accordance with
list discussion.
o Added HTTP CONNECT requirement in accordance with list discussion.
Author's Address Author's Address
Harald Alvestrand Harald Alvestrand
Google Google
Email: harald@alvestrand.no Email: harald@alvestrand.no
 End of changes. 33 change blocks. 
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