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In: MissingRef
Network Working Group                                      H. Alvestrand
Internet-Draft                                                    Google
Intended status: Standards Track                        January 22, 2014
Expires: July 26, 2014


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

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
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   Internet-Drafts are draft documents valid for a maximum of six months
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   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 July 26, 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
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   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.  Usage of Quality of Service functions . . . . . . . . . . . 4
     3.3.  Support for multiplexing  . . . . . . . . . . . . . . . . . 4
     3.4.  Middle box related functions  . . . . . . . . . . . . . . . 4
     3.5.  Transport protocols implemented . . . . . . . . . . . . . . 5
   4.  IANA Considerations . . . . . . . . . . . . . . . . . . . . . . 6
   5.  Security Considerations . . . . . . . . . . . . . . . . . . . . 6
   6.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . . . 6
   7.  References  . . . . . . . . . . . . . . . . . . . . . . . . . . 6
     7.1.  Normative References  . . . . . . . . . . . . . . . . . . . 6
     7.2.  Informative References  . . . . . . . . . . . . . . . . . . 8
   Appendix A.  Change log . . . . . . . . . . . . . . . . . . . . . . 8
     A.1.  Changes from -00 to -01 . . . . . . . . . . . . . . . . . . 8
     A.2.  Changes from -01 to -02 . . . . . . . . . . . . . . . . . . 9
   Author's Address  . . . . . . . . . . . . . . . . . . . . . . . . . 9































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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 protocos 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.

   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] 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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   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 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.

3.3.  Support for multiplexing

   RTCWEB implementations MUST support the ability to send and receive
   multiple SSRCs on the same transport, and MUST support the ability to
   send and receive multiple SSRCs on multiple simultaneous transports,
   including the ability to send and receive audio and video on the same
   transport.  The choice 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 be used.

   DISCUSSION: Minimizing the number of transports has advantages in
   traversing NATs and firewalls, due to 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 when
   flows with different characteristics are separated on different
   5-tuples.  Since the optimum for this tradeoff is unknown, and may be
   variable, it is inappropriate to embed this choice in the protocol
   layer, and this is therefore left to the control of the application.

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.

   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.



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   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 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 be supported; this allows
   applications to achieve peer-to-peer communication when both parties
   are behind UDP-blocking firewalls using a single TURN server.  (In
   this case, one can also achieve communication using two TURN servers
   that use TCP between the server and the client, and UDP between the
   TURN servers.)

   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.

   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.  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 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.




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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 (work in progress),
              December 2013.

   [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
              (work in progress), October 2013.

   [I-D.ietf-rtcweb-data-channel]
              Jesup, R., Loreto, S., and M. Tuexen, "RTCWeb Data
              Channels", draft-ietf-rtcweb-data-channel-06 (work in
              progress), October 2013.

   [I-D.ietf-rtcweb-rtp-usage]
              Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
              Communication (WebRTC): Media Transport and Use of RTP",



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              draft-ietf-rtcweb-rtp-usage-11 (work in progress),
              December 2013.

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

   [I-D.ietf-rtcweb-security-arch]
              Rescorla, E., "WebRTC Security Architecture",
              draft-ietf-rtcweb-security-arch-07 (work in progress),
              July 2013.

   [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 (work in progress),
              October 2013.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [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,



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              "TCP Candidates with Interactive Connectivity
              Establishment (ICE)", RFC 6544, March 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 (work in
              progress), October 2013.

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

   [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.

   [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-




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   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.

   .


Author's Address

   Harald Alvestrand
   Google

   Email: harald@alvestrand.no



















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