[Docs] [txt|pdf|xml|html] [Tracker] [WG] [Email] [Diff1] [Diff2] [Nits]

Versions: 00 01 02 03 04 05 06 07 08 09 10 11 12 13 14 15 16 17 Draft is active
In: MissingRef
Network Working Group                                      H. Alvestrand
Internet-Draft                                                    Google
Intended status: Standards Track                          March 31, 2014
Expires: October 2, 2014


                         Transports for RTCWEB
                    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.

   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 2, 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.





Alvestrand               Expires October 2, 2014                [Page 1]


Internet-Draft              WebRTC Transports                 March 2014


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





























Alvestrand               Expires October 2, 2014                [Page 2]


Internet-Draft              WebRTC Transports                 March 2014


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.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.




Alvestrand               Expires October 2, 2014                [Page 3]


Internet-Draft              WebRTC Transports                 March 2014


   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 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.  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.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



Alvestrand               Expires October 2, 2014                [Page 4]


Internet-Draft              WebRTC Transports                 March 2014


   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
   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



Alvestrand               Expires October 2, 2014                [Page 5]


Internet-Draft              WebRTC Transports                 March 2014


   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] 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.

   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



Alvestrand               Expires October 2, 2014                [Page 6]


Internet-Draft              WebRTC Transports                 March 2014


   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-06 (work in progress),
              March 2014.

   [I-D.ietf-mmusic-sctp-sdp]
              Loreto, S. and G. Camarillo, "Stream Control Transmission
              Protocol (SCTP)-Based Media Transport in the Session



Alvestrand               Expires October 2, 2014                [Page 7]


Internet-Draft              WebRTC Transports                 March 2014


              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-07 (work in
              progress), 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-12 (work in progress),
              February 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-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.

   [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.



Alvestrand               Expires October 2, 2014                [Page 8]


Internet-Draft              WebRTC Transports                 March 2014


   [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-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



Alvestrand               Expires October 2, 2014                [Page 9]


Internet-Draft              WebRTC Transports                 March 2014


              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"

   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.





Alvestrand               Expires October 2, 2014               [Page 10]


Internet-Draft              WebRTC Transports                 March 2014


   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













Alvestrand               Expires October 2, 2014               [Page 11]


Html markup produced by rfcmarkup 1.129b, available from https://tools.ietf.org/tools/rfcmarkup/