[asterisk-dev] [Code Review] 3571: res_pjsip_session: Fix leaking video RTP ports.

[rmudgett]

-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3571/
-----------------------------------------------------------

Review request for Asterisk Developers.


Bugs: ASTERISK-23721
    https://issues.asterisk.org/jira/browse/ASTERISK-23721


Repository: Asterisk


Description
-------

Simply enabling PJSIP to negotiage a video codec (e.g., h264) would leak video RTP ports if the codec were not negotiated by an incoming call.

* Made add_sdp_streams() associate the handler with the media stream if the handler handled the media stream.  Otherwise, when the ast_sip_session_media object was destroyed it didn't know how to clean up the RTP resources.

* Fixed sdp_requires_deferral() associating the handler with the media stream when deciding if the SDP processing needs to be deferred for T.38.  Like the leaked video RTP ports, the T.38 handler needs to clean up allocated resources when deciding if SDP processing needs to be deffered.

* Cleaned up some dead code in handle_incoming_sdp() and sdp_requires_deferral().


Diffs
-----

  /branches/12/res/res_pjsip_t38.c 414555 
  /branches/12/res/res_pjsip_session.c 414555 
  /branches/12/include/asterisk/res_pjsip_session.h 414555 

Diff: https://reviewboard.asterisk.org/r/3571/diff/


Testing
-------

With the patch the video RTP ports are no longer leaked.


Thanks,

rmudgett

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20140527/b306039f/attachment.html>