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Re: [rtcweb] JSEP question: How to set up simulcast for server-originated calls?

Iñaki Baz Castillo <ibc@aliax.net> Wed, 07 November 2018 12:00 UTC

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From: Iñaki Baz Castillo <ibc@aliax.net>
Date: Wed, 07 Nov 2018 13:00:42 +0100
Message-ID: <CALiegfkC3kTOmwfV4sRgJys5+_uYo0G_-R5CBdmaLtz6Hvx+KA@mail.gmail.com>
To: Stefan Håkansson LK <stefan.lk.hakansson@ericsson.com>
Cc: Bernard Aboba <bernard.aboba@gmail.com>, Cullen Jennings <fluffy@iii.ca>, rtcweb@ietf.org
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Subject: Re: [rtcweb] JSEP question: How to set up simulcast for server-originated calls?
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On Wed, 7 Nov 2018 at 12:48, Stefan Håkansson LK
<stefan.lk.hakansson@ericsson.com> wrote:
> An non inspected SDP offer/answer can influence _how_ it is sent, but I
> think that is per design - no point in encoding with an encoder that
> the remote can't decode etc.

I can buy that argument when it comes to just that: codec and RTCP
interoperability. Ok, WebRTC 1.0 assumes that app developers cannot
correlate capabilities in JS so the browser does it auto-magically.

But here we are talking about simulcast, which consumes resources and
requires more network uplink. And I want to control what my app sends
(and its max bitrate) because sending 3 streams instead of just 1
means more CPU usage, battery and bandwidth (money).

A received UFO should not change that.


-- 
Iñaki Baz Castillo
<ibc@aliax.net>

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