-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3571/
-----------------------------------------------------------
(Updated May 28, 2014, 11:54 a.m.)
Status
------
This change has been marked as submitted.
Review request for Asterisk Developers.
Changes
-------
Committed in revision 414749
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
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-dev