<?xml version="1.0" encoding="UTF-8"?>
<reference anchor="I-D.ietf-rtcweb-rtp-usage" target="https://datatracker.ietf.org/doc/html/draft-ietf-rtcweb-rtp-usage-26">
   <front>
      <title>Media Transport and Use of RTP in WebRTC</title>
      <author initials="C." surname="Perkins" fullname="Colin Perkins">
         <organization>University of Glasgow</organization>
      </author>
      <author initials="M." surname="Westerlund" fullname="Magnus Westerlund">
         <organization>Ericsson</organization>
      </author>
      <author initials="J." surname="Ott" fullname="Joerg Ott">
         <organization>Aalto University</organization>
      </author>
      <date month="March" day="17" year="2016" />
      <abstract>
	 <t>The framework for Web Real-Time Communication (WebRTC) provides support for direct interactive rich communication using audio, video, text, collaboration, games, etc.  between two peers&#x27; web browsers.  This memo describes the media transport aspects of the WebRTC framework.  It specifies how the Real-time Transport Protocol (RTP) is used in the WebRTC context and gives requirements for which RTP features, profiles, and extensions need to be supported.
	 </t>
      </abstract>
   </front>
   <seriesInfo name="Internet-Draft" value="draft-ietf-rtcweb-rtp-usage-26" />
   
</reference>
