One document matched: draft-ietf-rtcweb-transports-02.xml
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<rfc category="std" docName="draft-ietf-rtcweb-transports-02"
ipr="trust200902">
<front>
<title abbrev="WebRTC Transports">Transports for RTCWEB</title>
<author fullname="Harald Alvestrand" initials="H. T." surname="Alvestrand">
<organization>Google</organization>
<address>
<email>harald@alvestrand.no</email>
</address>
</author>
<date day="22" month="January" year="2014" />
<abstract>
<t>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.</t>
</abstract>
</front>
<middle>
<section title="Introduction">
<t>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.</t>
<t>The overall effort is described in the RTCWEB overview document,
<xref target="I-D.ietf-rtcweb-overview"></xref>. This document focuses
on the data transport protocos that are used by conforming
implementations.</t>
<t>This protocol suite is designed for WebRTC, and intends to satisfy
the security considerations described in the WebRTC security documents,
<xref target="I-D.ietf-rtcweb-security"></xref> and <xref
target="I-D.ietf-rtcweb-security-arch"></xref>.</t>
<t></t>
</section>
<section title="Requirements language">
<t>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 <xref
target="RFC2119">RFC 2119</xref>.</t>
</section>
<section anchor="app-transport"
title="Transport and Middlebox specification">
<t></t>
<section title="System-provided interfaces">
<t>The protocol specifications used here assume that the following
protocols are available to the implementations of the RTCWEB
protocols:</t>
<t><list style="symbols">
<t>UDP. This is the protocol assumed by most protocol elements
described.</t>
<t>TCP. This is used for HTTP/WebSockets, as well as for TURN/SSL
and ICE-TCP.</t>
</list></t>
<t>For both protocols, IPv4 and IPv6 support is assumed; applications
MUST be able to utilize both IPv4 and IPv6 where available.</t>
<t>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 <xref
target="I-D.dhesikan-tsvwg-rtcweb-qos"></xref> 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.</t>
<t>This specification does not assume that the implementation will
have access to ICMP or raw IP.</t>
</section>
<section title="Usage of Quality of Service functions">
<t>WebRTC implementations SHOULD attempt to set QoS on the packets
sent, according to the guidelines in <xref
target="I-D.dhesikan-tsvwg-rtcweb-qos"></xref>. It is appropriate to
depart from this recommendation when running on platforms where QoS
marking is not implemented.</t>
</section>
<section title="Support for multiplexing">
<t>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 <xref
target="I-D.ietf-mmusic-sdp-bundle-negotiation"></xref>. Further
information on RTP usage is found in <xref
target="I-D.ietf-rtcweb-rtp-usage"></xref>.</t>
<t>When different content types according to <xref
target="I-D.dhesikan-tsvwg-rtcweb-qos"> </xref> are used on the same
transport, appropriate per-packet DSCP marking SHOULD be used.</t>
<t>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.</t>
</section>
<section title="Middle box related functions">
<t>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).</t>
<t>ICE <xref target="RFC5245"></xref> MUST be supported. The
implementation MUST be a full ICE implementation, not ICE-Lite.</t>
<t>In order to deal with situations where both parties are behind NATs
which perform endpoint-dependent mapping (as defined in <xref
target="RFC5128"></xref> section 2.4), TURN <xref
target="RFC5766"></xref> MUST be supported.</t>
<t>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 <xref target="RFC5766"></xref> section 2.1 for details.</t>
<t>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 <xref target="RFC6156"></xref> MUST be supported.</t>
<t>TURN TCP candidates <xref target="RFC6062"></xref> 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.)</t>
<t>ICE-TCP candidates <xref target="RFC6544"></xref> MAY be supported;
this may allow applications to communicate to peers with public IP
addresses across UDP-blocking firewalls without using a TURN
server.</t>
<t>The ALTERNATE-SERVER mechanism specified in <xref
target="RFC5389"></xref> (STUN) section 11 (300 Try Alternate) MUST be
supported.</t>
<t>Further discussion of the interaction of RTCWEB with firewalls is
contained in <xref
target="I-D.hutton-rtcweb-nat-firewall-considerations"></xref>. This
document makes no requirements on interacting with HTTP proxies or
HTTP proxy configuration methods.</t>
</section>
<section title="Transport protocols implemented">
<t>For transport of media, secure RTP is used. The details of the
profile of RTP used are described in "RTP Usage" <xref
target="I-D.ietf-rtcweb-rtp-usage"></xref>.</t>
<t>For data transport over the RTCWEB data channel <xref
target="I-D.ietf-rtcweb-data-channel"></xref>, RTCWEB implementations
MUST support SCTP over DTLS over ICE. This encapsulation is specified
in <xref target="I-D.ietf-tsvwg-sctp-dtls-encaps"></xref>. Negotiation
of this transport in SDP is defined in <xref
target="I-D.ietf-mmusic-sctp-sdp"></xref>.</t>
<t>The setup protocol for RTCWEB data channels is described in <xref
target="I-D.jesup-rtcweb-data-protocol"></xref>.</t>
<t>RTCWEB implementations MUST support multiplexing of DTLS and RTP
over the same port pair, as described in the DTLS_SRTP specification
<xref target="RFC5764"></xref>, section 5.1.2. All application layer
protocol payloads over this DTLS connection are SCTP packets.</t>
</section>
</section>
<section anchor="IANA" title="IANA Considerations">
<t>This document makes no request of IANA.</t>
<t>Note to RFC Editor: this section may be removed on publication as an
RFC.</t>
</section>
<section anchor="Security" title="Security Considerations">
<t>Security considerations are enumerated in <xref
target="I-D.ietf-rtcweb-security"></xref>.</t>
</section>
<section anchor="Acknowledgements" title="Acknowledgements">
<t>This document is based on earlier versions embedded in <xref
target="I-D.ietf-rtcweb-overview"></xref>, which were the results of
contributions from many RTCWEB WG members.</t>
<t>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.</t>
</section>
</middle>
<back>
<references title="Normative References">
<?rfc include='reference.RFC.2119'?>
<?rfc include='reference.RFC.5245'?>
<?rfc include='reference.RFC.5389'?>
<?rfc include='reference.RFC.5764'?>
<?rfc include='reference.RFC.5766'?>
<?rfc include='reference.RFC.6062'?>
<?rfc include='reference.RFC.6156'?>
<?rfc include='reference.RFC.6544'?>
<?rfc include='reference.I-D.ietf-rtcweb-security'?>
<?rfc include='reference.I-D.ietf-rtcweb-rtp-usage'?>
<?rfc include='reference.I-D.ietf-rtcweb-data-channel'?>
<?rfc include='reference.I-D.ietf-rtcweb-security-arch'?>
<?rfc include='reference.I-D.dhesikan-tsvwg-rtcweb-qos'?>
<?rfc include='reference.I-D.ietf-tsvwg-sctp-dtls-encaps'?>
<?rfc include='reference.I-D.ietf-mmusic-sctp-sdp'?>
</references>
<references title="Informative References">
<?rfc include='reference.RFC.5128'?>
<?rfc include='reference.I-D.ietf-rtcweb-overview'?>
<?rfc include='reference.I-D.ietf-mmusic-sdp-bundle-negotiation'?>
<?rfc include='reference.I-D.jesup-rtcweb-data-protocol'?>
<?rfc include='reference.I-D.hutton-rtcweb-nat-firewall-considerations'?>
</references>
<section title="Change log">
<t></t>
<section title="Changes from -00 to -01">
<t><list style="symbols">
<t>Clarified DSCP requirements, with reference to -qos-</t>
<t>Clarified "symmetric NAT" -> "NATs which perform
endpoint-dependent mapping"</t>
<t>Made support of TURN over TCP mandatory</t>
<t>Made support of TURN over TLS a MAY, and added open
question</t>
<t>Added an informative reference to -firewalls-</t>
<t>Called out that we don't make requirements on HTTP proxy
interaction (yet</t>
</list></t>
</section>
<section title="Changes from -01 to -02">
<t><list style="symbols">
<t>Required support for 300 Alternate Server from STUN.</t>
<t>Separated the ICE-TCP candidate requirement from the TURN-TCP
requirement.</t>
<t>Added new sections on using QoS functions, and on multiplexing
considerations.</t>
<t>Removed all mention of RTP profiles. Those are the business of
the RTP usage draft, not this one.</t>
<t>Required support for TURN IPv6 extensions.</t>
<t>Removed reference to the TURN URI scheme, as it was
unnecessary.</t>
<t>Made an explicit statement that multiplexing (or not) is an
application matter.</t>
</list>.</t>
</section>
</section>
</back>
</rfc>
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