One document matched: draft-cbran-rtcweb-codec-01.xml

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<rfc category="std" docName="draft-cbran-rtcweb-codec-01"
     ipr="noDerivativesTrust200902">
  <front>
    <title abbrev="Abbreviated-Title">WebRTC Codec and Media Processing
    Requirements</title>

    <author fullname="Cary Bran" initials="C." surname="Bran">
      <organization>Plantronics</organization>

      <address>
        <postal>
          <street>345 Encinial Street</street>

          <city>Santa Cruz</city>

          <region>CA</region>

          <code>95060</code>

          <country>USA</country>
        </postal>

        <phone>+1 206 661-2398</phone>

        <email>cary.bran@plantronics.com</email>
      </address>
    </author>

    <author fullname="Cullen Jennings" initials="C." surname="Jennings">
      <organization>Cisco</organization>

      <address>
        <postal>
          <street>170 West Tasman Drive</street>

          <city>San Jose</city>

          <region>CA</region>

          <code>95134</code>

          <country>USA</country>
        </postal>

        <phone>+1 408 421-9990</phone>

        <email>fluffy@cisco.com</email>
      </address>
    </author>

    <date day="29" month="October" year="2011" />

    <abstract>
      <t>This document outlines the codec and media processing requirements
      for WebRTC client application and endpoint devices.</t>
    </abstract>
  </front>

  <middle>
    <section title="Introduction">
      <t>An integral part of the success and adoption of the Web Real Time
      Communications (WebRTC) will be the voice and video interoperability
      between WebRTC applications. This specification will outline the media
      processing and codec requirements for WebRTC client implementations.</t>
    </section>

    <section title="Terminology">
      <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 title="Codec Requirements">
      <t>This section covers the audio and video codec requirements for WebRTC
      client applications. To ensure a baseline level of interoperability
      between WebRTC clients, a minimum set of required codecs are specified
      below. While this section specifies the codecs that will be mandated for
      all WebRTC client implementations, it leaves the question of supporting
      additional codecs to the will of the implementer.</t>

      <section title="Audio Codec Requirements">
        <t>WebRTC clients are REQUIRED to implement the following audio
        codecs.</t>

        <t><list style="symbols">
            <t>PCMA/PCMU - 1 channel with a rate of 8000 Hz and a ptime of 20
            - see section 4.5.14 of <xref target="RFC3551"></xref></t>

            <t>Telephone Event - <xref target="RFC4734"></xref></t>

            <t>Opus [draft-ietf-codec-opus]</t>
          </list></t>
      </section>

      <section title="Video Codec Requirements">
        <t>If the MPEG-LA issues an intent to offer H.264 baseline profile on
        a royalty free basis for use in browsers before March 15, 2012, then
        the REQUIRED video codecs will be H.264 baseline. If this does not
        happen by that the date, then the REQUIRED video codec will be VP8
        <xref target="I-D.webm"></xref>.</t>

        <t>The following feature list applies to all required video
        codecs.</t>

        <t>Required video codecs:</t>

        <t><list style="symbols">
            <t>MUST support at least 10 frames per second (fps) and SHOULD
            support 30 fps</t>

            <t>If VP8, then MUST support a the bilinear and none
            reconstruction filters</t>

            <t>OPTIONALLY offer support for additional color spaces</t>

            <t>MUST support a minimum resolution of 320X240</t>

            <t>SHOULD support resolutions of 1280x720, 720x480, 1024x768,
            800x600, 640x480, 640 x 360 , 320x240</t>
          </list></t>
      </section>
    </section>

    <section title="WebRTC Client Requirements">
      <t>It is plausible that the dominant near to mid-term WebRTC usage model
      will be people using the interactive audio and video capabilities to
      communicate with each other via web browsers running on a notebook
      computer that has built-in microphone and speakers. The
      notebook-as-communication-device paradigm presents challenging echo
      cancellation and audio gain problems, the specific remedy of which will
      not be mandated here. However, while no specific algorithm or standard
      will be required by WebRTC compatible clients, functionality such as
      automatic gain control, echo cancellation, headset detection and passing
      call control events to connected devices will improve the user
      experience and should be implemented by the endpoint device.</t>

      <t>To address the problems outlined above, suitable implementations of
      the functionality listed below SHOULD be available within an RTC-Web
      endpoint device.</t>

      <t><list style="symbols">
          <t>Automatic gain control</t>

          <t>Ability to override automatic gain control to manually set gain
          </t>

          <t>Auto-adjustments to gain control and echo cancellation algorithms
          based on if a headset or internal speakers/microphone is being
          used</t>

          <t>Echo cancellation, including acoustic echo cancellation</t>

          <t>Headset detection</t>

          <t>Call control event notification to connected devices such as
          headsets</t>
        </list></t>
    </section>

    <section title="Legacy VoIP Interoperability">
      <t>The codec requirements above will ensure, at a minimum, voice
      interoperability capabilities between WebRTC client applications and
      legacy phone systems.</t>

      <t>Video interoperability will be dependent upon the MPEG-LA decision
      regarding H.264 baseline.</t>
    </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>The codec requirements have no additional security considerations
      other than those captured in <xref
      target="I-D.ekr-security-considerations-for-rtc-web"></xref>.</t>
    </section>

    <section anchor="Acknowledgements" title="Acknowledgements">
      <t>This draft incorporates ideas and text from various other drafts. In
      particularly we would like to acknowledge, and say thanks for, work we
      incorporated from Harald Alvestrand.</t>
    </section>
  </middle>

  <back>
    <references title="Normative References">
      <?rfc include="http://xml.resource.org/public/rfc/bibxml/reference.RFC.2119.xml"?>

      <?rfc include='http://xml.resource.org/public/rfc/bibxml/reference.RFC.3551.xml'?>

      <?rfc include='http://xml.resource.org/public/rfc/bibxml/reference.RFC.4734.xml'?>

      <reference anchor="I-D.ekr-security-considerations-for-rtc-web">
        <front>
          <title abbrev="RTC-Web Security">Security Considerations for
          RTC-Web</title>

          <author fullname="Eric Rescorla" initials="E.K." surname="Rescorla">
            <organization>RTFM, Inc.</organization>

            <address>
              <postal>
                <street>2064 Edgewood Drive</street>

                <city>Palo Alto</city>

                <region>CA</region>

                <code>94303</code>

                <country>USA</country>
              </postal>

              <phone>+1 650 678 2350</phone>

              <email>ekr@rtfm.com</email>
            </address>
          </author>

          <date day="30" month="May" year="2011" />

          <area>RAI</area>

          <workgroup>RTC-Web</workgroup>

          <abstract>
            <t>The Real-Time Communications on the Web (RTC-Web) working group
            is tasked with standardizing protocols for real-time
            communications between Web browsers. The two major use cases for
            RTC-Web technology are real-time audio and/or video calls and
            direct data transfer. Unlike most conventional real-time systems
            (e.g., SIP-based soft phones) RTC-Web communications are directly
            controlled by some Web server, which poses new security
            challenges. For instance, a Web browser might expose a JavaScript
            API which allows a server to place a video call. Unrestricted
            access to such an API would allow any site which a user visited to
            "bug" a user's computer, capturing any activity which passed in
            front of their camera. This document defines the RTC-Web threat
            model and defines an architecture which provides security within
            that threat model.</t>
          </abstract>

          <note title="Legal">
            <t>THIS DOCUMENT AND THE INFORMATION CONTAINED THEREIN ARE
            PROVIDED ON AN “AS IS” BASIS AND THE CONTRIBUTOR, THE
            ORGANIZATION HE/SHE REPRESENTS OR IS SPONSORED BY (IF ANY), THE
            INTERNET SOCIETY, THE IETF TRUST, AND THE INTERNET ENGINEERING
            TASK FORCE, DISCLAIM ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
            BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
            THEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
            MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.</t>
          </note>
        </front>
      </reference>

      <reference anchor="I-D.webm">
        <front>
          <title>VP8 Data Format and Decoding Guide</title>

          <author fullname="Google, Inc." surname="Google, Inc.">
            <organization></organization>
          </author>

          <date month="July" year="2010" />

          <note title="Link to WebM Project">
            <t>http://www.webmproject.org</t>
          </note>
        </front>
      </reference>
    </references>
  </back>
</rfc>