[Asterisk-code-review] res pjsip: Re-use IP version of signaling (SIP) for media (R... (asterisk[13])
Alexander Traud
asteriskteam at digium.com
Tue Sep 6 05:01:42 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 (#2).
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/63/3663/2
--
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: 2
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-Reviewer: Mark Michelson <mmichelson at digium.com>
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