Transports for RTCWEB
draft-ietf-rtcweb-transports-04
The information below is for an old version of the document.
| Document | Type |
This is an older version of an Internet-Draft that was ultimately published as RFC 8835.
|
|
|---|---|---|---|
| Author | Harald T. Alvestrand | ||
| Last updated | 2014-04-25 | ||
| RFC stream | Internet Engineering Task Force (IETF) | ||
| Formats | |||
| Reviews | |||
| Additional resources | Mailing list discussion | ||
| Stream | WG state | WG Document | |
| Document shepherd | Cullen Fluffy Jennings | ||
| IESG | IESG state | Became RFC 8835 (Proposed Standard) | |
| Consensus boilerplate | Unknown | ||
| Telechat date | (None) | ||
| Responsible AD | (None) | ||
| Send notices to | (None) |
draft-ietf-rtcweb-transports-04
Network Working Group H. Alvestrand
Internet-Draft Google
Intended status: Standards Track April 25, 2014
Expires: October 27, 2014
Transports for RTCWEB
draft-ietf-rtcweb-transports-04
Abstract
This document describes the data transport protocols used by RTCWEB,
including the protocols used for interaction with intermediate boxes
such as firewalls, relays and NAT boxes.
Status of this Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on October 27, 2014.
Copyright Notice
Copyright (c) 2014 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Requirements language . . . . . . . . . . . . . . . . . . . . 3
3. Transport and Middlebox specification . . . . . . . . . . . . 3
3.1. System-provided interfaces . . . . . . . . . . . . . . . . 3
3.2. Ability to use IPv4 and IPv6 . . . . . . . . . . . . . . . 4
3.3. Usage of temporary IPv6 addresses . . . . . . . . . . . . 4
3.4. Middle box related functions . . . . . . . . . . . . . . . 4
3.5. Transport protocols implemented . . . . . . . . . . . . . 5
4. Media Prioritization . . . . . . . . . . . . . . . . . . . . . 6
4.1. Usage of Quality of Service - DSCP and Multiplexing . . . 6
4.2. Local prioritization . . . . . . . . . . . . . . . . . . . 7
5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 8
6. Security Considerations . . . . . . . . . . . . . . . . . . . 8
7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 8
8. References . . . . . . . . . . . . . . . . . . . . . . . . . . 9
8.1. Normative References . . . . . . . . . . . . . . . . . . . 9
8.2. Informative References . . . . . . . . . . . . . . . . . . 11
Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 11
A.1. Changes from -00 to -01 . . . . . . . . . . . . . . . . . 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
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1. Introduction
The IETF RTCWEB effort, part of the WebRTC effort carried out in
cooperation between the IETF and the W3C, is aimed at specifying a
protocol suite that is useful for real time multimedia exchange
between browsers.
The overall effort is described in the RTCWEB overview document,
[I-D.ietf-rtcweb-overview]. This document focuses on the data
transport protocols that are used by conforming implementations.
This protocol suite is designed for WebRTC, and intends to satisfy
the security considerations described in the WebRTC security
documents, [I-D.ietf-rtcweb-security] and
[I-D.ietf-rtcweb-security-arch].
2. Requirements language
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [RFC2119].
3. Transport and Middlebox specification
3.1. System-provided interfaces
The protocol specifications used here assume that the following
protocols are available to the implementations of the RTCWEB
protocols:
o UDP. This is the protocol assumed by most protocol elements
described.
o TCP. This is used for HTTP/WebSockets, as well as for TURN/SSL
and ICE-TCP.
For both protocols, IPv4 and IPv6 support is assumed.
For UDP, this specification assumes the ability to set the DSCP code
point of the sockets opened on a per-packet basis, in order to
achieve the prioritizations described in [I-D.ietf-tsvwg-rtcweb-qos]
(see Section 4.1) when multiple media types are multiplexed. It does
not assume that the DSCP codepoints will be honored, and does assume
that they may be zeroed or changed, since this is a local
configuration issue.
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Platforms that do not give access to these interfaces will not be
able to support a conforming RTCWEB implementation.
This specification does not assume that the implementation will have
access to ICMP or raw IP.
3.2. Ability to use IPv4 and IPv6
Web applications running on top of the RTCWEB implementation MUST be
able to utilize both IPv4 and IPv6 where available - that is, when
two peers have only IPv4 connectivty to each other, or they have only
IPv6 connectivity to each other, applications running on top of the
RTCWEB implementation MUST be able to communicate.
When TURN is used, and the TURN server has IPv4 or IPv6 connectivity
to the peer or its TURN server, candidates of the appropriate types
MUST be supported. The "Happy Eyeballs" specification for ICE
[I-D.reddy-mmusic-ice-happy-eyeballs] SHOULD be supported.
3.3. Usage of temporary IPv6 addresses
The IPv6 default address selection specification [RFC6724] specifies
that temporary addresses [RFC4941] are to be preferred over permanent
addresses. This is a change from the rules specified by [RFC3484].
For applications that select a single address, this is usually done
by the IPV6_PREFER_SRC_TMP preference flag specified in [RFC5014].
However, this rule is not completely obvious in the ICE scope. This
is therefore clarified as follows:
When a client gathers all IPv6 addresses on a host, and both
temporary addresses and permanent addresses of the same scope are
present, the client SHOULD discard the permanent addresses before
forming pairs. This is consistent with the default policy described
in [RFC6724].
3.4. Middle box related functions
The primary mechanism to deal with middle boxes is ICE, which is an
appropriate way to deal with NAT boxes and firewalls that accept
traffic from the inside, but only from the outside if it's in
response to inside traffic (simple stateful firewalls).
ICE [RFC5245] MUST be supported. The implementation MUST be a full
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
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which perform endpoint-dependent mapping (as defined in [RFC5128]
section 2.4), TURN [RFC5766] MUST be supported.
Configuration of STUN and TURN servers, both from browser
configuration and from an applicaiton, MUST be supported.
In order to deal with firewalls that block all UDP traffic, TURN
using TCP between the client and the server MUST be supported, and
TURN using TLS over TCP between the client and the server MUST be
supported. See [RFC5766] section 2.1 for details.
In order to deal with situations where one party is on an IPv4
network and the other party is on an IPv6 network, TURN extensions
for IPv6 [RFC6156] MUST be supported.
TURN TCP candidates [RFC6062] MAY be supported.
However, such candidates are not seen as providing any significant
benefit. First, use of TURN TCP would only be relevant in cases
which both peers are required to use TCP to establish a
PeerConnection. Secondly, that use case is anyway supported by both
sides establishing UDP relay candidates using TURN over TCP to
connect to the relay server. Thirdly, using TCP only between the
endpoint and its relay may result in less issues with TCP in regards
to real-time constraints, e.g. due to head of line blocking.
ICE-TCP candidates [RFC6544] MUST be supported; this may allow
applications to communicate to peers with public IP addresses across
UDP-blocking firewalls without using a TURN server.
If TCP connections are used, RTP framing according to [RFC4571] MUST
be used, both for the RTP packets and for the DTLS packets used to
carry data channels.
The ALTERNATE-SERVER mechanism specified in [RFC5389] (STUN) section
11 (300 Try Alternate) MUST be supported.
Further discussion of the interaction of RTCWEB with firewalls is
contained in [I-D.hutton-rtcweb-nat-firewall-considerations]. This
document makes no requirements on interacting with HTTP proxies or
HTTP proxy configuration methods.
NOTE IN DRAFT: This may be added.
3.5. Transport protocols implemented
For transport of media, secure RTP is used. The details of the
profile of RTP used are described in "RTP Usage"
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[I-D.ietf-rtcweb-rtp-usage].
For data transport over the RTCWEB data channel
[I-D.ietf-rtcweb-data-channel], RTCWEB implementations MUST support
SCTP over DTLS over ICE. This encapsulation is specified in
[I-D.ietf-tsvwg-sctp-dtls-encaps]. Negotiation of this transport in
SDP is defined in [I-D.ietf-mmusic-sctp-sdp]. The SCTP extension for
NDATA, [I-D.ietf-tsvwg-sctp-ndata], MUST be supported.
The setup protocol for RTCWEB data channels is described in
[I-D.jesup-rtcweb-data-protocol].
RTCWEB implementations MUST support multiplexing of DTLS and RTP over
the same port pair, as described in the DTLS_SRTP specification
[RFC5764], section 5.1.2. All application layer protocol payloads
over this DTLS connection are SCTP packets.
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
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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
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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.
Note to RFC Editor: this section may be removed on publication as an
RFC.
6. Security Considerations
Security considerations are enumerated in [I-D.ietf-rtcweb-security].
7. Acknowledgements
This document is based on earlier versions embedded in
[I-D.ietf-rtcweb-overview], which were the results of contributions
from many RTCWEB WG members.
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Special thanks for reviews of earlier versions of this draft go to
Magnus Westerlund, Markus Isomaki and Dan Wing; the contributions
from Andrew Hutton also deserve special mention.
8. References
8.1. Normative References
[I-D.ietf-mmusic-sctp-sdp]
Loreto, S. and G. Camarillo, "Stream Control Transmission
Protocol (SCTP)-Based Media Transport in the Session
Description Protocol (SDP)", draft-ietf-mmusic-sctp-sdp-06
(work in progress), February 2014.
[I-D.ietf-rtcweb-data-channel]
Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data
Channels", draft-ietf-rtcweb-data-channel-08 (work in
progress), April 2014.
[I-D.ietf-rtcweb-rtp-usage]
Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
Communication (WebRTC): Media Transport and Use of RTP",
draft-ietf-rtcweb-rtp-usage-13 (work in progress),
April 2014.
[I-D.ietf-rtcweb-security]
Rescorla, E., "Security Considerations for WebRTC",
draft-ietf-rtcweb-security-06 (work in progress),
January 2014.
[I-D.ietf-rtcweb-security-arch]
Rescorla, E., "WebRTC Security Architecture",
draft-ietf-rtcweb-security-arch-09 (work in progress),
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]
Tuexen, M., Stewart, R., Jesup, R., and S. Loreto, "DTLS
Encapsulation of SCTP Packets",
draft-ietf-tsvwg-sctp-dtls-encaps-03 (work in progress),
February 2014.
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[I-D.ietf-tsvwg-sctp-ndata]
Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann, "A
New Data Chunk for Stream Control Transmission Protocol",
draft-ietf-tsvwg-sctp-ndata-00 (work in progress),
February 2014.
[I-D.reddy-mmusic-ice-happy-eyeballs]
Reddy, T., Patil, P., and P. Martinsen, "Happy Eyeballs
Extension for ICE",
draft-reddy-mmusic-ice-happy-eyeballs-06 (work in
progress), February 2014.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC4571] Lazzaro, J., "Framing Real-time Transport Protocol (RTP)
and RTP Control Protocol (RTCP) Packets over Connection-
Oriented Transport", RFC 4571, July 2006.
[RFC4941] Narten, T., Draves, R., and S. Krishnan, "Privacy
Extensions for Stateless Address Autoconfiguration in
IPv6", RFC 4941, September 2007.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245,
April 2010.
[RFC5389] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
"Session Traversal Utilities for NAT (STUN)", RFC 5389,
October 2008.
[RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for the Secure
Real-time Transport Protocol (SRTP)", RFC 5764, May 2010.
[RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using
Relays around NAT (TURN): Relay Extensions to Session
Traversal Utilities for NAT (STUN)", RFC 5766, April 2010.
[RFC6062] Perreault, S. and J. Rosenberg, "Traversal Using Relays
around NAT (TURN) Extensions for TCP Allocations",
RFC 6062, November 2010.
[RFC6156] Camarillo, G., Novo, O., and S. Perreault, "Traversal
Using Relays around NAT (TURN) Extension for IPv6",
RFC 6156, April 2011.
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[RFC6544] Rosenberg, J., Keranen, A., Lowekamp, B., and A. Roach,
"TCP Candidates with Interactive Connectivity
Establishment (ICE)", RFC 6544, March 2012.
[RFC6724] Thaler, D., Draves, R., Matsumoto, A., and T. Chown,
"Default Address Selection for Internet Protocol Version 6
(IPv6)", RFC 6724, September 2012.
8.2. Informative References
[I-D.hutton-rtcweb-nat-firewall-considerations]
Stach, T., Hutton, A., and J. Uberti, "RTCWEB
Considerations for NATs, Firewalls and HTTP proxies",
draft-hutton-rtcweb-nat-firewall-considerations-03 (work
in progress), January 2014.
[I-D.ietf-rtcweb-overview]
Alvestrand, H., "Overview: Real Time Protocols for Brower-
based Applications", draft-ietf-rtcweb-overview-09 (work
in progress), February 2014.
[I-D.jesup-rtcweb-data-protocol]
Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel
Protocol", draft-jesup-rtcweb-data-protocol-04 (work in
progress), February 2013.
[RFC3484] Draves, R., "Default Address Selection for Internet
Protocol version 6 (IPv6)", RFC 3484, February 2003.
[RFC5014] Nordmark, E., Chakrabarti, S., and J. Laganier, "IPv6
Socket API for Source Address Selection", RFC 5014,
September 2007.
[RFC5128] Srisuresh, P., Ford, B., and D. Kegel, "State of Peer-to-
Peer (P2P) Communication across Network Address
Translators (NATs)", RFC 5128, March 2008.
Appendix A. Change log
A.1. Changes from -00 to -01
o Clarified DSCP requirements, with reference to -qos-
o Clarified "symmetric NAT" -> "NATs which perform endpoint-
dependent mapping"
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o Made support of TURN over TCP mandatory
o Made support of TURN over TLS a MAY, and added open question
o Added an informative reference to -firewalls-
o Called out that we don't make requirements on HTTP proxy
interaction (yet
A.2. Changes from -01 to -02
o Required support for 300 Alternate Server from STUN.
o Separated the ICE-TCP candidate requirement from the TURN-TCP
requirement.
o Added new sections on using QoS functions, and on multiplexing
considerations.
o Removed all mention of RTP profiles. Those are the business of
the RTP usage draft, not this one.
o Required support for TURN IPv6 extensions.
o Removed reference to the TURN URI scheme, as it was unnecessary.
o Made an explicit statement that multiplexing (or not) is an
application matter.
.
A.3. Changes from -02 to -03
o Added required support for draft-ietf-tsvwg-sctp-ndata
o Removed discussion of multiplexing, since this is present in rtp-
usage.
o Added RFC 4571 reference for framing RTP packets over TCP.
o Downgraded TURN TCP candidates from SHOULD to MAY, and added more
language discussing TCP usage.
o Added language on IPv6 temporary addresses.
o Added language describing multiplexing choices.
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o Added a separate section detailing what it means when we say that
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
Harald Alvestrand
Google
Email: harald@alvestrand.no
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