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To: Ben Campbell <ben@nostrum.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Cc: Alex Gouaillard <dralex@millicast.com>
References: <085505e3-9899-f817-b6df-db8022d52e26@gmail.com>
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From: Adam Roach <adam@nostrum.com>
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Date: Mon, 30 Nov 2020 11:56:09 -0600
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Subject: Re: [dispatch] WHIP - WebRTC HTTP ingestion protocol
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Having read through the proposal and watched the corresponding segment 
of the DISPATCH meeting from two weeks ago, I think that having 
something like this standardized would be very useful, both for the 
handful of broadcasting tools that exist as well as for the various 
WebRTC broadcast networks that are being deployed.

So I definitely think we should pursue this work in the IETF.

It's a little less clear to me whether we want to slot this into 
something like MMUSIC or to create a short-lived working group. I'm 
leaning towards the second, for a couple of reasons. First, I found 
Ted's comments at the DISPATCH meeting to be kind of illuminating as far 
as the kinds of additional work that might be necessary (and I have my 
own list of things that might need some refinement, like negotiation of 
authentication mechanisms); and that nudges the effort into a "maybe a 
little larger than just one work item in a long-standing working group." 
More importantly, this work straddles the real-time media and web 
disciplines, and I'd like to make sure we have people from the latter 
group putting eyes on it. I seriously doubt we can get HTTP experts 
showing up at MMUSIC, but there's at least a fair chance that we might 
get them interested in a lower-volume mailing list such as a dedicated 
WHIP WG.

I think this mirrors, for the most part, the tentative conclusion that 
came out of the DISPATCH discussion earlier this month. If we decide to 
take this path of creating a short-lived WG, I'm happy to help craft a 
charter and plan to be actively involved in the technical work.

/a


On 10/28/2020 3:10 PM, Ben Campbell wrote:
> Hi Everyone,
>
> Did anyone have feedback or other thoughts on Sergio’s proposal?
>
> Thanks!
>
> Ben.
>
>> On Sep 30, 2020, at 5:24 AM, Sergio Garcia Murillo 
>> <sergio.garcia.murillo@gmail.com 
>> <mailto:sergio.garcia.murillo@gmail.com>> wrote:
>>
>> Hi all!
>>
>>
>> While WebRTC has been very successful in a wide range of scenarios, 
>> its adaption 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 traditional streaming to support e.g. ABR, and 
>> comes at no cost. However webrtc implements client-side ABR, 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.
>>
>> We have been working on a proposal for a simple HTTP based protocol 
>> that will allow WebRTC endpoints to ingest content into streaming 
>> services and/or CDNs to fill this gap and facilitate deployment:
>>
>>   * https://tools.ietf.org/html/draft-murillo-whip-00
>>   * https://github.com/murillo128/webrtc-http-ingest-protocol/
>>
>>
>> We have already implemented it on Janus and Medooze media servers:
>>
>>   * https://www.meetecho.com/blog/whip-janus/
>>   * https://medium.com/@medooze/whip-webrtc-meets-the-broadcasting-world-86772eba8ae7
>>
>>
>> And added support into a WebRTC version of OBS studio:
>>
>>   * https://github.com/CoSMoSoftware/OBS-studio-webrtc/releases/tag/m84v23.2-RC2
>>
>>
>> We also plan to have an interop session on the next IETF hackhaton, 
>> that will allow to check the interoperability between different 
>> WebRTC implementations.
>>
>>
>> What would be the best way of moving this forward? Obviously, any 
>> feedback will be very welcome.
>>
>>
>> Best regards
>>
>> Sergio
>>
>>
>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org <mailto:dispatch@ietf.org>
>> https://www.ietf.org/mailman/listinfo/dispatch
>
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch



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    <p>Having read through the proposal and watched the corresponding
      segment of the DISPATCH meeting from two weeks ago, I think that
      having something like this standardized would be very useful, both
      for the handful of broadcasting tools that exist as well as for
      the various WebRTC broadcast networks that are being deployed.</p>
    <p>So I definitely think we should pursue this work in the IETF.</p>
    <p>It's a little less clear to me whether we want to slot this into
      something like MMUSIC or to create a short-lived working group.
      I'm leaning towards the second, for a couple of reasons. First, I
      found Ted's comments at the DISPATCH meeting to be kind of
      illuminating as far as the kinds of additional work that might be
      necessary (and I have my own list of things that might need some
      refinement, like negotiation of authentication mechanisms); and
      that nudges the effort into a "maybe a little larger than just one
      work item in a long-standing working group." More importantly,
      this work straddles the real-time media and web disciplines, and
      I'd like to make sure we have people from the latter group putting
      eyes on it. I seriously doubt we can get HTTP experts showing up
      at MMUSIC, but there's at least a fair chance that we might get
      them interested in a lower-volume mailing list such as a dedicated
      WHIP WG.</p>
    <p>I think this mirrors, for the most part, the tentative conclusion
      that came out of the DISPATCH discussion earlier this month. If we
      decide to take this path of creating a short-lived WG, I'm happy
      to help craft a charter and plan to be actively involved in the
      technical work.</p>
    <p>/a<br>
    </p>
    <div class="moz-cite-prefix"><br>
    </div>
    <div class="moz-cite-prefix">On 10/28/2020 3:10 PM, Ben Campbell
      wrote:<br>
    </div>
    <blockquote type="cite"
      cite="mid:0E35658D-D768-4DF2-BF0F-95D6F5889B4F@nostrum.com">
      <meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
      Hi Everyone,
      <div class=""><br class="">
      </div>
      <div class="">Did anyone have feedback or other thoughts on
        Sergio’s proposal?</div>
      <div class=""><br class="">
      </div>
      <div class="">Thanks!</div>
      <div class=""><br class="">
      </div>
      <div class="">Ben.<br class="">
        <div><br class="">
          <blockquote type="cite" class="">
            <div class="">On Sep 30, 2020, at 5:24 AM, Sergio Garcia
              Murillo &lt;<a
                href="mailto:sergio.garcia.murillo@gmail.com" class=""
                moz-do-not-send="true">sergio.garcia.murillo@gmail.com</a>&gt;
              wrote:</div>
            <br class="Apple-interchange-newline">
            <div class="">
              <meta http-equiv="content-type" content="text/html;
                charset=UTF-8" class="">
              <div class="">
                <p class="">Hi all!</p>
                <p class=""><br class="">
                </p>
                <p class="">While WebRTC has been very successful in a
                  wide range of scenarios, its adaption 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.<br class="">
                  <br class="">
                  These protocols are much older than WebRTC and lack by
                  default some important security and resilience
                  features provided by WebRTC with minimal delay.<br
                    class="">
                  <br class="">
                  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 traditional streaming to support
                  e.g. ABR, and comes at no cost. However webrtc
                  implements client-side ABR, 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.<br class="">
                  <br class="">
                  We have been working on a proposal for a simple HTTP
                  based protocol that will allow WebRTC endpoints to
                  ingest content into streaming services and/or CDNs to
                  fill this gap and facilitate deployment:</p>
                <ul class="">
                  <li class=""><a class="moz-txt-link-freetext"
                      href="https://tools.ietf.org/html/draft-murillo-whip-00"
                      moz-do-not-send="true">https://tools.ietf.org/html/draft-murillo-whip-00</a></li>
                  <li class=""><a class="moz-txt-link-freetext"
                      href="https://github.com/murillo128/webrtc-http-ingest-protocol/"
                      moz-do-not-send="true">https://github.com/murillo128/webrtc-http-ingest-protocol/</a></li>
                </ul>
                <p class=""><br class="">
                </p>
                <p class="">We have already implemented it on Janus and
                  Medooze media servers:</p>
                <ul class="">
                  <li class=""><a class="moz-txt-link-freetext"
                      href="https://www.meetecho.com/blog/whip-janus/"
                      moz-do-not-send="true">https://www.meetecho.com/blog/whip-janus/</a></li>
                  <li class=""><a class="moz-txt-link-freetext"
href="https://medium.com/@medooze/whip-webrtc-meets-the-broadcasting-world-86772eba8ae7"
                      moz-do-not-send="true">https://medium.com/@medooze/whip-webrtc-meets-the-broadcasting-world-86772eba8ae7</a></li>
                </ul>
                <p class=""><br class="">
                </p>
                <p class="">And added support into a WebRTC version of
                  OBS studio:</p>
                <ul class="">
                  <li class=""><a class="moz-txt-link-freetext"
href="https://github.com/CoSMoSoftware/OBS-studio-webrtc/releases/tag/m84v23.2-RC2"
                      moz-do-not-send="true">https://github.com/CoSMoSoftware/OBS-studio-webrtc/releases/tag/m84v23.2-RC2</a></li>
                </ul>
                <p class=""><br class="">
                </p>
                <p class="">We also plan to have an interop session on
                  the next IETF hackhaton, that will allow to check the
                  interoperability between different WebRTC
                  implementations.<br class="">
                </p>
                <p class=""><br class="">
                </p>
                <p class="">What would be the best way of moving this
                  forward? Obviously, any feedback will be very welcome.<br
                    class="">
                </p>
                <p class=""><br class="">
                </p>
                <p class="">Best regards</p>
                <p class="">Sergio<br class="">
                </p>
                <p class=""><br class="">
                </p>
                <p class=""><br class="">
                </p>
              </div>
              _______________________________________________<br
                class="">
              dispatch mailing list<br class="">
              <a href="mailto:dispatch@ietf.org" class=""
                moz-do-not-send="true">dispatch@ietf.org</a><br class="">
              <a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf.org/mailman/listinfo/dispatch</a><br class="">
            </div>
          </blockquote>
        </div>
        <br class="">
      </div>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <pre class="moz-quote-pre" wrap="">_______________________________________________
dispatch mailing list
<a class="moz-txt-link-abbreviated" href="mailto:dispatch@ietf.org">dispatch@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf.org/mailman/listinfo/dispatch</a>
</pre>
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