[Asterisk-code-review] res pjsip: Re-use IP version of signaling (SIP) for media (R... (asterisk[13])

Joshua Colp asteriskteam at digium.com
Tue Sep 27 09:56:14 CDT 2016


Joshua Colp has abandoned this change.

Change subject: res_pjsip: Re-use IP version of signaling (SIP) for media (RTP).
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Abandoned

I'm abandoning this review in favor of the dual stack one I published. While it has a problem with RTP/RTCP specifically the local address it was tested at SIPit and has been confirmed to work, even with the DNS failover support in Asterisk 14 (which can cause a message to start out as IPv6 and then actually get sent as IPv4). It also allows ICE with both IPv4 and IPv6 candidates to work. Comments on it would be welcome there.

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Gerrit-MessageType: abandon
Gerrit-Change-Id: I01a85a8c6723fcc12e86139f80e090e2078d04bb
Gerrit-PatchSet: 4
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: Alexander Traud <pabstraud at compuserve.com>
Gerrit-Reviewer: Alexander Traud <pabstraud at compuserve.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Mark Michelson <mmichelson at digium.com>



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