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Re: [MMUSIC] [rtcweb] Default proto transport in JSEP

Roman Shpount <roman@telurix.com> Mon, 19 November 2018 20:21 UTC

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From: Roman Shpount <roman@telurix.com>
Date: Mon, 19 Nov 2018 15:20:53 -0500
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To: Iñaki Baz Castillo <ibc@aliax.net>
Cc: RTCWeb IETF <rtcweb@ietf.org>, mmusic WG <mmusic@ietf.org>
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Subject: Re: [MMUSIC] [rtcweb] Default proto transport in JSEP
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On Mon, Nov 19, 2018 at 3:09 PM Iñaki Baz Castillo <ibc@aliax.net> wrote:

> Which interop? If a peer is WebRTC compatible then it MUST use ICE, so
> whatever is written in the c= line and whatever the m= protoo is, it
> won't need to read it. And if the other party is "legacy" and does not
> implement WebRTC, then they both just can not communicate.
>
> So this is just cosmetic and has zero value. And there are endless
> discussions about this topic for the last 3-4 years just because those
> fields exist in the SDP.
>
> 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.

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.

Regards,
_____________
Roman Shpount

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