[Asterisk-code-review] res_rtp_asterisk: Send correct sender SSRC when p2p bridge in use (asterisk[13])
Joshua Colp
asteriskteam at digium.com
Tue Mar 10 07:33:16 CDT 2020
Joshua Colp has posted comments on this change. ( https://gerrit.asterisk.org/c/asterisk/+/13859 )
Change subject: res_rtp_asterisk: Send correct sender SSRC when p2p bridge in use
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Patch Set 2:
> Patch Set 2: Code-Review+1
>
> I think this has a possibility of breaking WebRTC and requires further testing. Specifically in WebRTC SSRC is exchanged in the SDP between both sides, and if video is in use with bundle then that SSRC has meaning since it allows you to determine when you receive a packet what stream it is in regards to. If this scenario is present and local bridging is done, then this would cause
Ignore my comment. I meant to remove it after further investigation.
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Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Change-Id: I39f923bde28ebb4f0fddc926b92494aed294a478
Gerrit-Change-Number: 13859
Gerrit-PatchSet: 2
Gerrit-Owner: Torrey Searle <tsearle at gmail.com>
Gerrit-Reviewer: Friendly Automation
Gerrit-Reviewer: Joshua Colp <jcolp at sangoma.com>
Gerrit-Reviewer: Sean Bright <sean.bright at gmail.com>
Gerrit-Comment-Date: Tue, 10 Mar 2020 12:33:16 +0000
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