diff options
| author | Mark Michelson <mmichelson@digium.com> | 2015-01-28 17:42:48 +0000 |
|---|---|---|
| committer | Mark Michelson <mmichelson@digium.com> | 2015-01-28 17:42:48 +0000 |
| commit | b3ff43a4e8f0616440a2820f03c930ffe2008bc4 (patch) | |
| tree | d70500dd229b1c36fb354d6283dc553d6fbeca20 /res/res_pjsip_sdp_rtp.c | |
| parent | 3cccfac399e2330b5a4084b806db8f2f18ebf0f2 (diff) | |
Fix file descriptor leak in RTP code.
SIP requests that offered codecs incompatible with configured values
could result in the allocation of RTP and RTCP ports that would not get
reclaimed later.
ASTERISK-24666 #close
Reported by Y Ateya
Review: https://reviewboard.asterisk.org/r/4323
AST-2015-001
........
Merged revisions 431300 from http://svn.asterisk.org/svn/asterisk/branches/12
........
Merged revisions 431303 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'res/res_pjsip_sdp_rtp.c')
| -rw-r--r-- | res/res_pjsip_sdp_rtp.c | 1 |
1 files changed, 1 insertions, 0 deletions
diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c index a3ccd19d4..bc8c748eb 100644 --- a/res/res_pjsip_sdp_rtp.c +++ b/res/res_pjsip_sdp_rtp.c @@ -1243,6 +1243,7 @@ static void stream_destroy(struct ast_sip_session_media *session_media) ast_rtp_instance_stop(session_media->rtp); ast_rtp_instance_destroy(session_media->rtp); } + session_media->rtp = NULL; } /*! \brief SDP handler for 'audio' media stream */ |