Re: [rtcweb] Proposal to break the ICE impasse

[Christer Holmberg <[e-mail]>]

Hi,



> I'm not yet persuaded this is needed. The alleged need here is that there are some ICE-implementing endpoints which will choke if
> they see a profile that doesn't match any actual candidate. I recognize that this is required by 5245, but that doesn't mean anyone
> ever did it. Can you please point me to a client which would interoperate with a WebRTC endpoint with this change that would not
> if you just always sent the same profile as JSEP currently requires.

I don't think it is ok to *specify* that discarding a MUST is ok as long as nobody can show an implementation that would break by doing so.

Having said that, in order to prevent an RTCWEB shutdown I am generally ok with Adam's approach. I would like my pull request comments to be addressed, though, that is related to separation between the JSEP API and an application using it: an application should be allowed to act according to 5245/draft-ice-sdp and update the c/m line if it wishes to, but due to the way the JSEP API works such applications might sometimes always include the same value in the c/m- line.

I also think it shall be explicitly written that JSEP does not update 5245/draft-ice-sdp, but rather deviates when it comes to the c/m- line.

Regards,

Christer




On Thu, Jan 24, 2019 at 11:12 AM Adam Roach <[e-mail]<mailto:[e-mail]>> wrote:
Based on conversations in MMUSIC, as well as several offline
conversations with interested parties, I've put together a proposed
change to JSEP that, if accepted, will allow publication of the Cluster
238 documents to move forward.

Note that this new text has no impact on existing implementations (at
least, as far as I am able to discern), which do not currently have the
capability of producing media sections consisting of exclusively TCP
candidates. From that perspective, the change makes existing
implementations no less compliant with JSEP than they were before.

What this change does provide is both on-paper and in-the-future
compatibility between WebRTC implementations once they finalize TCP
candidate handling (and candidate handling in general for mid-session
offers).

   https://github.com/rtcweb-wg/jsep/pull/862/files

The key insight here is that JSEP's use of ICE completely discards any
meaning associated with the transport parameter, while SIP's use of ICE
does not. The trivial change that I propose, which bears only on future
WebRTC implementations -- that is, which has no as-built specification
to point to -- allows JSEP to continue to ignore the value of the
transport parameter, while specifying that it says the right thing for
SIP implementations to function properly.

/a

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