[Asterisk-code-review] res pjsip session/BUNDLE: Handle no audio codecs on endpoint (asterisk[15])

[George Joseph]

Hello Jenkins2, Joshua Colp, 

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

    https://gerrit.asterisk.org/6555

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

Change subject: res_pjsip_session/BUNDLE:  Handle no audio codecs on endpoint
......................................................................

res_pjsip_session/BUNDLE:  Handle no audio codecs on endpoint

When an INVITE came in with both audio and video streams but there
were no audio codecs defined for the endpoint, we weren't declining
the audio stream.  Since it's usually the first/transport stream,
when the video stream was processed and tried to use the transport,
it was empty and caused a crash.  We now decline the the stream if
there are no matching codecs so when the video stream is processed,
it's now the first/transport stream and processes normally.

Change-Id: Ic854eda54c95031e66b076ecfae3041d34daa692
---
M res/res_pjsip_session.c
1 file changed, 22 insertions(+), 5 deletions(-)


  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/55/6555/3
-- 
To view, visit https://gerrit.asterisk.org/6555
To unsubscribe, visit https://gerrit.asterisk.org/settings

Gerrit-Project: asterisk
Gerrit-Branch: 15
Gerrit-MessageType: newpatchset
Gerrit-Change-Id: Ic854eda54c95031e66b076ecfae3041d34daa692
Gerrit-Change-Number: 6555
Gerrit-PatchSet: 3
Gerrit-Owner: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Jenkins2
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-code-review/attachments/20170922/dfb1fc1b/attachment.html>