[Asterisk-code-review] chan sip.c: Start ICE negotiation when response is sent or r... (asterisk[11])

[Eugene Voityuk]

Eugene Voityuk has posted comments on this change.

Change subject: chan_sip.c: Start ICE negotiation when response is sent or received.
......................................................................


Patch Set 1:

Richard Mudgett: Everyone in the thread had an issue, that i am purposing solution for, no audio after call answered after 7 secs of ringing. Video in asterisk 11 + chan_sip + WebRTC is not working at all, it simply can't parse Invite SDP which includes two packs of candidates (one for audio and one for video). Even with my patch, and even with patch for VP8 pass-through, video won't work, as asterisk thinks that invite for secured video was sent without encryption details (as it can't parse one SDP with encryption details for video and for audio ). About text  i can't tell you anything as didn't ever used it. But obviously it should also be fixed somehow. You simply can't start ICE negotiation when any of asterisk/client don't have at least one pair of local and one remote candidate.  ICE stands that the best practice for that is using rel100, which chan_sip also don't support, so only one reason left is starting ICE negotiation only after sending or receiving response, hoping that remote party received you candidates.

-- 
To view, visit https://gerrit.asterisk.org/1751
To unsubscribe, visit https://gerrit.asterisk.org/settings

Gerrit-MessageType: comment
Gerrit-Change-Id: I55a632bde9e9827871b09141d82747e08379a8ca
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 11
Gerrit-Owner: Eugene Voityuk <eugene at thirdlane.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Eugene Voityuk <eugene at thirdlane.com>
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Matt Jordan <mjordan at digium.com>
Gerrit-Reviewer: Richard Mudgett <rmudgett at digium.com>
Gerrit-HasComments: No