One document matched: draft-ietf-rtcweb-transports-04.xml
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<rfc category="std" docName="draft-ietf-rtcweb-transports-04"
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="25" month="April" 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 protocols 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.</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.ietf-tsvwg-rtcweb-qos"></xref> (see <xref
target="s-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>Platforms that do not give access to these interfaces will not be
able to support a conforming RTCWEB implementation.</t>
<t>This specification does not assume that the implementation will
have access to ICMP or raw IP.</t>
</section>
<section title="Ability to use IPv4 and IPv6">
<t>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.</t>
<t>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 <xref
target="I-D.reddy-mmusic-ice-happy-eyeballs"></xref> SHOULD be
supported.</t>
</section>
<section title="Usage of temporary IPv6 addresses">
<t>The IPv6 default address selection specification <xref
target="RFC6724"></xref> specifies that temporary addresses <xref
target="RFC4941"></xref> are to be preferred over permanent addresses.
This is a change from the rules specified by <xref
target="RFC3484"></xref>. For applications that select a single
address, this is usually done by the IPV6_PREFER_SRC_TMP preference
flag specified in <xref target="RFC5014"></xref>. However, this rule
is not completely obvious in the ICE scope. This is therefore
clarified as follows:</t>
<t>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
<xref target="RFC6724"></xref>.</t>
</section>
<section anchor="s-middlebox" 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; this
allows interworking with both ICE and ICE-Lite implementations when
they are deployed appropriately.</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>Configuration of STUN and TURN servers, both from browser
configuration and from an applicaiton, 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 over TCP 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> MAY be
supported.</t>
<t>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.</t>
<t>ICE-TCP candidates <xref target="RFC6544"></xref> MUST 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>If TCP connections are used, RTP framing according to <xref
target="RFC4571"></xref> MUST be used, both for the RTP packets and
for the DTLS packets used to carry data channels.</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>
<t>NOTE IN DRAFT: This may be added.</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>. The SCTP extension for
NDATA, <xref target="I-D.ietf-tsvwg-sctp-ndata"></xref>, MUST be
supported.</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 title="Media Prioritization">
<t>The RTCWEB prioritization model is that the application tells the
RTCWEB implementation about the priority of media and data flows through
an API.</t>
<t>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.</t>
<t>The priority settings affect two pieces of behavior: Packet markings
and packet send sequence decisions. Each is described in its own section
below.</t>
<section anchor="s-qos"
title="Usage of Quality of Service - DSCP and Multiplexing">
<t>WebRTC implementations SHOULD attempt to set QoS on the packets
sent, according to the guidelines in <xref
target="I-D.ietf-tsvwg-rtcweb-qos"></xref>. It is appropriate to
depart from this recommendation when running on platforms where QoS
marking is not implemented.</t>
<t>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?)</t>
<t>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:</t>
<t><list style="symbols">
<t>Each media stream carried on its own 5-tuple</t>
<t>Media streams grouped by media type into 5-tuples (such as
carrying all audio on one 5-tuple)</t>
<t>All media sent over a single 5-tuple, with or without
differentiation into 6-tuples based on DSCP code points</t>
</list>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.</t>
<t>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 <xref
target="I-D.ietf-rtcweb-rtp-usage"></xref>.</t>
<t>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.</t>
<t>Sending data over multiple 5-tuples is not supported.</t>
<t>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?</t>
<t>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.</t>
<t>A receiving implementation MUST be able to receive media and data
in all these configurations.</t>
</section>
<section title="Local prioritization">
<t>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 payload bytes) of the level below.</t>
<t>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.</t>
<t>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.</t>
<t>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.</t>
<t>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.</t>
<t>NOTE: The appropriate algorithm for deciding when to send SCTP data
vs media data is not described yet.</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.4571'?>
<?rfc include='reference.RFC.4941'?>
<?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.RFC.6724'?>
<?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.ietf-tsvwg-rtcweb-qos'?>
<?rfc include='reference.I-D.ietf-tsvwg-sctp-dtls-encaps'?>
<?rfc include='reference.I-D.ietf-mmusic-sctp-sdp'?>
<?rfc include='reference.I-D.ietf-tsvwg-sctp-ndata'?>
<?rfc include='reference.I-D.reddy-mmusic-ice-happy-eyeballs'?>
</references>
<references title="Informative References">
<?rfc include='reference.RFC.5128'?>
<?rfc include='reference.RFC.5014'?>
<?rfc include='reference.RFC.3484'?>
<?rfc include='reference.I-D.ietf-rtcweb-overview'?>
<?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 title="Changes from -02 to -03">
<t><list style="symbols">
<t>Added required support for draft-ietf-tsvwg-sctp-ndata</t>
<t>Removed discussion of multiplexing, since this is present in
rtp-usage.</t>
<t>Added RFC 4571 reference for framing RTP packets over TCP.</t>
<t>Downgraded TURN TCP candidates from SHOULD to MAY, and added
more language discussing TCP usage.</t>
<t>Added language on IPv6 temporary addresses.</t>
<t>Added language describing multiplexing choices.</t>
<t>Added a separate section detailing what it means when we say
that an RTCWEB implementation MUST support both IPv4 and IPv6.</t>
</list></t>
</section>
<section title="Changes from -03 to -04">
<t><list style="symbols">
<t>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.</t>
<t>ICE-TCP candidates was changed from MAY to MUST, in recognition
of the sense of the room at the London IETF.</t>
</list></t>
</section>
</section>
</back>
</rfc>
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