One document matched: draft-alvestrand-one-rtp-00.xml

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<rfc category="std" docName="draft-alvestrand-one-rtp-00" ipr="trust200902">
  <front>
    <title abbrev="Single RTP Session Grouping">SDP Grouping for Single RTP
    Sessions</title>

    <author fullname="Harald Tveit Alvestrand" initials="H. T."
            surname="Alvestrand">
      <organization>Google</organization>

      <address>
        <postal>
          <street>Kungsbron 2</street>

          <city>Stockholm</city>

          <region></region>

          <code>11122</code>

          <country>Sweden</country>
        </postal>

        <email>harald@alvestrand.no</email>
      </address>
    </author>

    <date day="11" month="August" year="2011" />

    <abstract>
      <t>This document describes an extension to the Session Description
      Protocol (SDP) to describe RTP sessions where media of multiple top
      level types, for example audio and video, are carried in the same RTP
      session.</t>

      <t>This document is presented to the RTCWEB, AVTCORE and MMUSIC WGs for
      consideration.</t>
    </abstract>

    <note 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>
    </note>
  </front>

  <middle>
    <section title="Introduction">
      <t>In the work with the RTCWEB specifications <xref
      target="I-D.ietf-rtcweb-overview"></xref>, a need was discovered for
      representing within the SDP framework an SDP session consisting of a
      single RTP session, where that single RTP session, mapped to a single
      transport flow, contained multiple top-level data types.</t>

      <t>This is advantageous for the use case where there is no desire for
      different treatment by the network of the different flows in the
      session, there exist other appropriate mechanisms (for instance based on
      SSRC) to identify the flows in the session to the applications, and
      where the handling of multiple RTP sessions would increase the work
      required to establish the session (for instance by requiring multiple
      ICE <xref target="RFC5245"></xref> negotiations, or handling of failure
      cases where one RTP session is established and another is not).</t>

      <t>This document describes how to represent such a session.</t>

      <t></t>
    </section>

    <section title="Requirements for a solution">
      <t>The requirements for our representation are:</t>

      <t><list style="symbols">
          <t>It should be possible to represent an SDP session consisting of a
          single RTP session, where that session carries both audio and
          video.</t>

          <t>If this description is presented in an Offer in the offer/answer
          model to an entity that does not understand it, the resulting Answer
          should contain a valid description of an SDP session consisting of
          one video RTP session and one audio RTP session.</t>
        </list></t>
    </section>

    <section title="SDP Grouping Framework Parameter">
      <t>This document defines a new semantics extension called TOGETHER
      within the SDP Grouping framework <xref target="RFC5888"></xref>.</t>

      <t>If this semantics extension is present in an SDP Session-level
      a=group: line, the semantics are that the two or more m lines are
      intended to be read as components of a single RTP session, creating a
      single SSRC numbering space that can contain components of all the types
      described in the referenced media sections.</t>

      <t>The following properties of the media sections are REQUIRED:</t>

      <t><list style="symbols">
          <t>The defined RTPMAP values of the section MUST NOT overlap</t>

          <t>The profile of the sections MUST be the same (e.g RTP/AVPF)</t>
        </list>The media sections MAY contain connection data (port numbers or
      ICE parameters), but some of these may be ignored in processing (see
      next section).</t>
    </section>

    <section title="Use in Offer/Answer">
      <t>This extension MAY be included in an Offer; if it is not included in
      an Offer, it MUST NOT be included in an answer. [Note: I believe this to
      be true for all extensions within the Grouping framework. Check.]</t>

      <t>If the responder understands the semantics of the TOGETHER extension,
      the parameters of the first section MUST be used to establish the RTP
      session, and the parameters for the other sections MUST be ignored.</t>

      <t>The following parameters are taken from the first section only:</t>

      <t><list style="symbols">
          <t>Port number from the m= line</t>

          <t>All media-level attributes defined in RFC 5245 section 15.1 -
          this includes "candidate", "remote-candidates", "ice-mismatch",
          "ice-ufrag", "ice-pwd"</t>
        </list>The bandwidth of the "m" line is treated specially: The values
      for all "m=" lines in the group are added together, and the resulting
      value is taken to be the negotiated bandwidth value for the RTP
      session.</t>

      <t>The expected behaviour when the extension is present in an offer and
      not understood is that the generated answer will not contain the
      "a=group:TOGETHER" line, and that each sections' parameters will be
      used.</t>
    </section>

    <section title="Interaction with other extensions">
      <t>If other extensions modify the bandwidth calculation algorithm, those
      extensions will have to take into consideration how bandwidth from
      multiple sections of the SDP description should be merged.</t>
    </section>

    <section title="RTCP bandwidth considerations">
      <t>A concern has been raised that when audio and video are combined, the
      bandwidth of RTCP reports required for an audio stream may exceed the
      bandwidth of the audio stream itself, which seems a bit bizarre. While
      not critical (overall RTCP bandwidth is still limited to 5% of the total
      bandwidth), this warrants a little more study.</t>

      <t>Considering a combined RTP session with one sender and one recipient,
      four 1-Mbit/sec video flows and four 100-Kbit/sec audio flows flowing in
      one direction.</t>

      <t>The total bandwidth is 4.4 Mbit/sec, so if the RTCP share of the
      bandwidth is 5% as recommended by RFC 3550 section 6.2, the RTCP
      bandwidth limit is 220 Kbits/sec. Eight SSRCs need to be reported
      on.</t>

      <t>[CHECK: Does this mean there are 9 reporters (8 senders and 1
      receiver), or 2 reporters (1 of each)? Assuming 9.]</t>

      <t>Each report sender will have 24.4 Kbits/second of RTCP bandwidth at
      its disposal. Assuming a packet size of 100 bytes (11 bytes per SSRC
      reported on), the maximum RTCP rate allowed is 30 RTCP packets per
      second, which is slightly slower than the typical audio heartbeat flow
      of 50 packets per second (20 ms interval).</t>

      <t>If this is deemed excessive, one can adopt the RTP/AVPF model of
      5-second regular RTCP reports with additional availability of
      "on-demand" RTCP packets. But the RTCP feedback interval also enters
      into congestion control algorithms, which may complicate the
      picture.</t>

      <t></t>
    </section>

    <section title="Examples">
      <t>The examples are taken from RFC 4317, "SDP Offer/Answer
      Examples".</t>

      <t></t>

      <figure>
        <preamble>Offer</preamble>

        <artwork><![CDATA[
      v=0
      o=alice 2890844526 2890844526 IN IP4 host.atlanta.example.com
      s=
      c=IN IP4 host.atlanta.example.com
      t=0 0
      a=group:TOGETHER foo bar
      m=audio 49170 RTP/AVP 0 8 97
      a=mid:foo
      b=AS:200
      a=rtpmap:0 PCMU/8000
      a=rtpmap:8 PCMA/8000
      a=rtpmap:97 iLBC/8000
      m=video 51372 RTP/AVP 31 32
      a=mid:bar
      b=AS:1000
      a=rtpmap:31 H261/90000
      a=rtpmap:32 MPV/90000]]></artwork>
      </figure>

      <t></t>

      <t>This is a request to have both audio and video sent over port 49170.
      If this can't be done, audio will be sent over port 49170, and video
      will be sent on port 51372. The total bandwidth, if combined, is 1200
      Kbits/second; if separated, 200 Kbits goes to audio and 1000 Kbits goes
      to video.</t>

      <figure>
        <preamble>Answer, from an entity that understands TOGETHER</preamble>

        <artwork><![CDATA[
      v=0
      o=bob 2808844564 2808844564 IN IP4 host.biloxi.example.com
      s=
      c=IN IP4 host.biloxi.example.com
      t=0 0
      a=group:TOGETHER foo bar
      m=audio 49174 RTP/AVP 0
      a=mid:foo
      b=AS:200
      a=rtpmap:0 PCMU/8000
      m=video 49170 RTP/AVP 32
      a=mid:bar
      b=AS:1000
      a=rtpmap:32 MPV/90000]]></artwork>
      </figure>

      <figure>
        <preamble>Answer, from an entity that understands grouping, but does
        not understand TOGETHER</preamble>

        <artwork><![CDATA[      
      v=0
      o=bob 2808844564 2808844564 IN IP4 host.biloxi.example.com
      s=
      c=IN IP4 host.biloxi.example.com
      t=0 0
      m=audio 49174 RTP/AVP 0
      a=mid:foo
      a=rtpmap:0 PCMU/8000
      b=AS:200
      m=video 49170 RTP/AVP 32
      a=mid:bar
      a=rtpmap:32 MPV/90000
      b=AS:1000]]></artwork>
      </figure>
    </section>

    <section anchor="IANA" title="IANA Considerations">
      <t>This document requests IANA to register the new SDP Grouping semantic
      extension called TOGETHER.</t>

      <t></t>
    </section>

    <section anchor="Security" title="Security Considerations">
      <t>No new security issues have been raised specifically for this
      extension.</t>

      <t>Third-party interceptors that sniff negotiation but do not understand
      the extension may end up listening to the wrong port number for some of
      the media flows. This is not deemed greatly harmful.</t>
    </section>

    <section anchor="Acknowledgements" title="Acknowledgements">
      <t>This draft is based on a discussion between a number of participants
      at the Quebec City IETF, July 2011, about the issue of multiplexing
      audio and video on a single network transport using RTP.</t>
    </section>
  </middle>

  <back>
    <references title="Normative References">
      <?rfc include="reference.RFC.2119"?>

      <?rfc include='reference.RFC.3550'?>

      <?rfc include='reference.RFC.5245'?>

      <?rfc include='reference.RFC.5888'?>
    </references>

    <references title="Informative References">
      <?rfc include='reference.I-D.ietf-rtcweb-overview'?>
    </references>
  </back>
</rfc>