Re: [dispatch] WHIP

My feedback:

1) this is a good idea - we need something like this. Full webRTC O/A is too complex for simple ingestion usecases.
2) A URI is an excellent hook to hang this on.
3) I don't think it goes far enough.

Almost all of the offer/answer content is redundant, because most of it is _known_ by both sides already.
The ingestion services typically specify what codecs/bitrates/framerates they will accept.
They don't output any media.

I claim we can simplify the O/A to something that looks more like RTSP's SDP or even simpler.

- So I support this, but would want to discuss simplification.

Tim.

> On 28 Oct 2020, at 20:10, Ben Campbell <ben@nostrum.com> 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://tools.ietf.org/html/draft-murillo-whip-00>
>> https://github.com/murillo128/webrtc-http-ingest-protocol/ <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://www.meetecho.com/blog/whip-janus/>
>> https://medium.com/@medooze/whip-webrtc-meets-the-broadcasting-world-86772eba8ae7 <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 <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
>> 
>> 
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Re: [dispatch] WHIP - WebRTC HTTP ingestion protocol