[asterisk-dev] [Code Review] 4000: res_pjsip_sdp_rtp.c: Fix native formats containing formats that were not negotiated.

[rmudgett]

-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/4000/
-----------------------------------------------------------

(Updated Sept. 19, 2014, 12:08 p.m.)


Status
------

This change has been marked as submitted.


Review request for Asterisk Developers.


Changes
-------

Committed in revision 423561


Bugs: AFS-162
    https://issues.asterisk.org/jira/browse/AFS-162


Repository: Asterisk


Description
-------

Outgoing PJSIP calls can result in non-negotiated formats listed in the channel's native formats if video formats are listed in the endpoint's configuration.  The resulting call could then use a non-negotiated format resulting in one way audio.

* Simplified the update of session->req_caps in set_caps().  Why do something in five steps when only one is needed?


Diffs
-----

  /branches/13/res/res_pjsip_sdp_rtp.c 423446 
  /branches/13/channels/chan_pjsip.c 423446 

Diff: https://reviewboard.asterisk.org/r/4000/diff/


Testing
-------

Configured PJSIP endpoints with: allow=!all,h264,g722,h263,ulaw,h263p,alaw
Called from D40 with g722 among other formats enabled to a Polycom that negotiates ulaw.
Before the patch, Asterisk would send g722 frames to the Polycom.  The resulting call had one way audio because the Polycom does not understand g722.
After the patch, Asterisk sends ulaw frames to the Polycom.


Thanks,

rmudgett

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20140919/5ae943f2/attachment.html>