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

George Joseph asteriskteam at digium.com
Mon Sep 12 08:56:23 CDT 2016


Hello Anonymous Coward #1000019, Joshua Colp,

I'd like you to reexamine a change.  Please visit

    https://gerrit.asterisk.org/3663

to look at the new patch set (#3).

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

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

Previously, the parameter rtp_ipv6 had to be configured for each endpoint in the
file pjsip.conf, in advance, before a SIP client connected to Asterisk. That
defeated the idea of an IPv4/IPv6 Dual Stack server, for which the IP versions
of the client is *not* known in advance. This was no issue for the channel
driver chan_sip and therefore this was changed in res_pjsip only.

ASTERISK-26309 #close

Change-Id: I01a85a8c6723fcc12e86139f80e090e2078d04bb
---
M CHANGES
M configs/samples/pjsip.conf.sample
M contrib/scripts/sip_to_pjsip/sip_to_pjsip.py
M res/res_pjsip_sdp_rtp.c
M res/res_pjsip_session.c
M res/res_pjsip_t38.c
6 files changed, 85 insertions(+), 32 deletions(-)


  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/63/3663/3
-- 
To view, visit https://gerrit.asterisk.org/3663
To unsubscribe, visit https://gerrit.asterisk.org/settings

Gerrit-MessageType: newpatchset
Gerrit-Change-Id: I01a85a8c6723fcc12e86139f80e090e2078d04bb
Gerrit-PatchSet: 3
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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