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

[Joshua Colp]

Joshua Colp has posted comments on this change.

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


Patch Set 1:

That is precisely what it did. If enabled you could establish an RTP session with IPv6, while the SIP signaling used an IPv4 transport, since they were separate. You've now changed this to be automatic with a fallback to the configuration option (which isn't bad). Since the automatic support takes precedence that means that it is a behavior change. If my SIP is using IPv4 then my RTP is going to use IPv4, even if I've turned rtp_ipv6 on. The number of people who would do that are small, but it's a behavior change nonetheless with no ability to change it.

-- 
To view, visit https://gerrit.asterisk.org/3663
To unsubscribe, visit https://gerrit.asterisk.org/settings

Gerrit-MessageType: comment
Gerrit-Change-Id: I01a85a8c6723fcc12e86139f80e090e2078d04bb
Gerrit-PatchSet: 1
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: Joshua Colp <jcolp at digium.com>
Gerrit-HasComments: No