wdiff draft-ietf-rtcweb-transports-02.txt draft-ietf-rtcweb-transports-03.txt

Network Working Group H. Alvestrand
Internet-Draft Google
Intended status: Standards Track January 22, March 31, 2014
Expires: July 26, October 2, 2014

Transports for RTCWEB
draft-ietf-rtcweb-transports-02
draft-ietf-rtcweb-transports-03

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.

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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. Usage of Quality of Service functions . Ability to use IPv4 and IPv6 . . . . . . . . . . 4
3.3. Support for multiplexing . . . . . 4
3.3. Usage of temporary IPv6 addresses . . . . . . . . . . . . 4
3.4. Usage of Quality of Service - DSCP and Multiplexing . . . 4
3.5. Middle box related functions . . . . . . . . . . . . . . . 4
3.5. 5
3.6. Transport protocols implemented . . . . . . . . . . . . . . 5 6
4. IANA Considerations . . . . . . . . . . . . . . . . . . . . . . 6 7
5. Security Considerations . . . . . . . . . . . . . . . . . . . . 6 7
6. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 6 7
7. References . . . . . . . . . . . . . . . . . . . . . . . . . . 6 7
7.1. Normative References . . . . . . . . . . . . . . . . . . . 6 7
7.2. Informative References . . . . . . . . . . . . . . . . . . 8 9
Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . . 8 10
A.1. Changes from -00 to -01 . . . . . . . . . . . . . . . . . . 8 10
A.2. Changes from -01 to -02 . . . . . . . . . . . . . . . . . 10
A.3. Changes from -02 to -03 . . 9 . . . . . . . . . . . . . . . 11
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 9 11

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 protocos 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; applications
MUST be able to utilize both IPv4 and IPv6 where available. 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.dhesikan-tsvwg-rtcweb-qos] (see Section 3.4) when multiple media
types are multiplexed. It does not assume that the DSCP codepoints
will be honored, and does assume that they may be zeroed or changed,
since this is a local configuration issue.

Platforms that do not give access to these interfaces will not be
able to support a conforming RTCWEB implementation.

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

3.2. Usage of Quality of Service functions

WebRTC implementations SHOULD attempt to set QoS on the packets sent,
according Ability to the guidelines in [I-D.dhesikan-tsvwg-rtcweb-qos]. It
is appropriate to depart from this recommendation when use IPv4 and IPv6

Web applications running on
platforms where QoS marking is not implemented.

3.3. Support for multiplexing top of the RTCWEB implementations implementation MUST support the ability be
able to send utilize both IPv4 and receive
multiple SSRCs 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 same transport, and
RTCWEB implementation MUST support the ability be able to
send communicate.

When TURN is used, and receive multiple SSRCs on multiple simultaneous transports,
including the ability TURN server has IPv4 or IPv6 connectivity
to send and receive audio and video on the same
transport. The choice peer or its TURN server, candidates of configuration is done at higher layers
(above transport), using mechanisms like BUNDLE
[I-D.ietf-mmusic-sdp-bundle-negotiation]. Further information on RTP
usage is found in [I-D.ietf-rtcweb-rtp-usage].

When different content types according to
[I-D.dhesikan-tsvwg-rtcweb-qos] are used on the same transport, appropriate per-packet DSCP marking SHOULD types
MUST be used.

DISCUSSION: Minimizing 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 number rules specified by [RFC3484].
For applications that select a single address, this is usually done
by the IPV6_PREFER_SRC_TMP 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 transports has advantages the same scope are
present, the client SHOULD discard the permanent addresses before
forming pairs. This is consistent with the default policy described
in
traversing NATs [RFC6724].

3.4. Usage of Quality of Service - DSCP and firewalls, due Multiplexing

WebRTC implementations SHOULD attempt to set QoS on the reduced chance of
negotiation failure. However, some network prioritization mechanisms
(in particular active queue management techniques and flow-
recognizing deep packet inspection boxes) will perform better 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 different characteristics are separated or without
differentiation into 6-tuples based on different
5-tuples. Since DSCP code points

In each of the optimum for this tradeoff is unknown, and configurations mentioned, data channels may be
variable, it is inappropriate to embed this choice carried
in its own 5-tuple, or multiplexed together with one of the protocol
layer, media
flows.

More complex configurations, such as sending a high priority video
stream on one 5-tuple and this 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 therefore left there a need to place the control of "group by media type,
with data multiplexed on the application.

3.4. 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
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.

In order to deal with situations where both parties are behind NATs
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] SHOULD MAY be supported; this allows
applications to achieve peer-to-peer communication when 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 parties peers are behind UDP-blocking firewalls required to use TCP to establish a
PeerConnection. Secondly, that use case is anyway supported by both
sides establishing UDP relay candidates using a single TURN over TCP to
connect to the relay server. (In
this case, one can also achieve communication Thirdly, using two TURN servers
that use TCP only between the server and the client,
endpoint and UDP between the
TURN servers.) 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] MAY 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.

3.5.

NOTE IN DRAFT: This may be added.

3.6. Transport protocols implemented

For transport of media, secure RTP is used. The details of the
profile of RTP used are described in "RTP Usage"
[I-D.ietf-rtcweb-rtp-usage].

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. IANA Considerations

This document makes no request of IANA.

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

5. Security Considerations

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

6. Acknowledgements

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

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

7. References

7.1. Normative References

[I-D.dhesikan-tsvwg-rtcweb-qos]
Dhesikan, S., Druta, D., Jones, P., and J. Polk, "DSCP and
other packet markings for RTCWeb QoS",
draft-dhesikan-tsvwg-rtcweb-qos-03
draft-dhesikan-tsvwg-rtcweb-qos-06 (work in progress),
December 2013.
March 2014.

[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-05 draft-ietf-mmusic-sctp-sdp-06
(work in progress), October 2013. February 2014.

[I-D.ietf-rtcweb-data-channel]
Jesup, R., Loreto, S., and M. Tuexen, "RTCWeb "WebRTC Data
Channels", draft-ietf-rtcweb-data-channel-06 draft-ietf-rtcweb-data-channel-07 (work in
progress), October 2013. February 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-11
draft-ietf-rtcweb-rtp-usage-12 (work in progress),
December 2013.
February 2014.

[I-D.ietf-rtcweb-security]
Rescorla, E., "Security Considerations for WebRTC",
draft-ietf-rtcweb-security-05
draft-ietf-rtcweb-security-06 (work in progress),
July 2013.
January 2014.

[I-D.ietf-rtcweb-security-arch]
Rescorla, E., "WebRTC Security Architecture",
draft-ietf-rtcweb-security-arch-07
draft-ietf-rtcweb-security-arch-09 (work in progress),
July 2013.
February 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-02
draft-ietf-tsvwg-sctp-dtls-encaps-03 (work in progress),
October 2013.
February 2014.

[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.

[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.

7.2. 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-02 (work
in progress), September 2013.

[I-D.ietf-mmusic-sdp-bundle-negotiation]
Holmberg, C., Alvestrand, H., and C. Jennings,
"Multiplexing Negotiation Using Session Description
Protocol (SDP) Port Numbers",
draft-ietf-mmusic-sdp-bundle-negotiation-05
draft-hutton-rtcweb-nat-firewall-considerations-03 (work
in progress), October 2013. January 2014.

[I-D.ietf-rtcweb-overview]
Alvestrand, H., "Overview: Real Time Protocols for Brower-
based Applications", draft-ietf-rtcweb-overview-08 draft-ietf-rtcweb-overview-09 (work
in progress), September 2013. 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"

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.

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

Author's Address

Harald Alvestrand
Google

Email: harald@alvestrand.no