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Re: [rtcweb] Default proto transport in JSEP

Iñaki Baz Castillo <ibc@aliax.net> Mon, 19 November 2018 20:31 UTC

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From: Iñaki Baz Castillo <ibc@aliax.net>
Date: Mon, 19 Nov 2018 21:31:13 +0100
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To: Roman Shpount <roman@telurix.com>
Cc: rtcweb@ietf.org, mmusic@ietf.org
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Subject: Re: [rtcweb] Default proto transport in JSEP
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On Mon, 19 Nov 2018 at 21:21, Roman Shpount <roman@telurix.com> wrote:
> ICE specifications (both RFC 5245 and ice-sip-sdp) are saying that these fields are not cosmetic. All I am asking is to put values in these fields that indicate that they are cosmetic instead of creating some rules on how these fields should change which would cause potential interop issues.

I suggest:

m=KIND 9 ICE [codec PTs]


> BTW, there are plenty of VoIP phones which use ICE and support Opus (or G.711). They will also produce ICE mismatch since they implement RFC 5245. So, these end points are WebRTC compatible except for this specific issue.

If those phones already implement ICE (and let's assume also
DTLS-SRTP), how will a wrong or "invalid" c= or m= content affect
them? Browsers won't complain no matter what the SDP re-offer contains
in those lines, so the phone can put whatever it wishes. And the phone
should not read them in received SDP re-offers, so I don't fully
understand what the problem is.


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

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