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Re: [Wish] WHIP should also allow *media* to be sent over the HTTPS connection

Tim Panton <tim@pi.pe> Sun, 14 March 2021 13:57 UTC

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From: Tim Panton <tim@pi.pe>
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Date: Sun, 14 Mar 2021 13:57:01 +0000
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To: Ross Finlayson <finlayson@live555.com>
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Subject: Re: [Wish] WHIP should also allow *media* to be sent over the HTTPS connection
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Sent from my iPad

> On 13 Mar 2021, at 03:59, Ross Finlayson <finlayson@live555.com> wrote:
> 
> 
> 
>> On Mar 12, 2021, at 7:41 PM, Alexandre GOUAILLARD <agouaillard@gmail.com> wrote:
>> 
>> Media management is entirely delegated to the RTCWEB specifications. In particular, the use of ICE, and the use of DTLS-SRTP are mandatory. For clarification, the use of JSEP, SDP, SDP O/A are also all mandatory.
> 
> Alex,
> 
> I think you may be missing my point.  A WebRTC stream that happens to negotiate COMEDIA transport (with RFC 4571 framing of RTP/RTCP) is still, in my mind, a WebRTC stream.  It would be negotiated the same way as any other WebRTC stream (using JSEP, SDP, O/A).  But the stream itself wouldn’t use datagrams, and thus wouldn’t use ICE (because it wouldn’t need it).  That’s why I described this as being ‘WHIP-light’ - a subset of the WHIP protocol just for this simple class of streams.
> 
> Perhaps I have a more expansive view of WebRTC than everyone else, but I consider what I’m describing as being a WebRTC stream; albeit a special case.


Would it run DTLS over comedia ?
If not how would the SRTP keys be generated ?
how would bandwidth estimation /congestion control work ?
Without ice how does continuing consent work?

Unless I am mistaken it will take quite a lot of work to engineer this into the webRTC stack on a browser.

I am also not convinced that the results are much different from piping webAudio into a websocket. 

T.
> 
>    Ross.
> 
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