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<reference anchor="I-D.murillo-whip" target="https://datatracker.ietf.org/doc/html/draft-murillo-whip-00">
   <front>
      <title>WebRTC-HTTP ingestion protocol (WHIP)</title>
      <author initials="S. G." surname="Murillo" fullname="Sergio Garcia Murillo">
         <organization>CoSMo Software</organization>
      </author>
      <author initials="A." surname="Gouaillard" fullname="Dr. Alex Gouaillard">
         <organization>CoSMo Software</organization>
      </author>
      <date month="September" day="9" year="2020" />
      <abstract>
	 <t>   While WebRTC has been very sucessfull in a wide range of scenarios,
   its adption in the broadcasting/streaming industry is lagging behind.
   Currently there is no standard protocol (like SIP or RTSP) designed
   for ingesting media in a streaming service, and content providers
   still rely heavily on protocols like RTMP for it.

   These protocols are much older than webrtc and lack by default some
   important security and resilience features provided by webrtc with
   minimal delay.

   The media codecs used in older protocols do not always match those
   being used in WebRTC, mandating transcoding on the ingest node,
   introducing delay and degrading media quality.  This transcoding step
   is always present in traditionnal streaming to support e.g.  ABR, and
   comes at no cost.  However webrtc implements client-side ABR, also
   called Network-Aware Encoding by e.g.  Huavision, by means of
   simulcast and SVC codecs, which otherwise alleviate the need for
   server-side transcoding.  Content protection and Privacy Enhancement
   can be achieve with End-to-End Encryption, which preclude any server-
   side media processing.

   This document proposes a simple HTTP based protocol that will allow
   WebRTC endpoings to ingest content into streaming servics and/or CDNs
   to fill this gap and facilitate deployment.

	 </t>
      </abstract>
   </front>
   <seriesInfo name="Internet-Draft" value="draft-murillo-whip-00" />
   
</reference>
