diff options
| author | mmichelson <mmichelson@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-06-23 15:08:27 +0000 |
|---|---|---|
| committer | mmichelson <mmichelson@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-06-23 15:08:27 +0000 |
| commit | bffcd79a2b620a8b30f80174b677be9e4dbabbb7 (patch) | |
| tree | 2bc4a555d6db4d592134350585fa93b2fcea72b9 /channels | |
| parent | 3c0fb2cd2ea2dae7f95da88267fcaf01e4a5202e (diff) | |
Fix potential memory leak in chan_sip when video rtp is not allocated properly.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@202572 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels')
| -rw-r--r-- | channels/chan_sip.c | 4 |
1 files changed, 4 insertions, 0 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index a1e627931..99b3d4f3c 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -4600,6 +4600,10 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si if (!p->rtp || (ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && !p->vrtp)) { ast_log(LOG_WARNING, "Unable to create RTP audio %s session: %s\n", ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) ? "and video" : "", strerror(errno)); + /* If rtp was successfully allocated, but vrtp was not, then we need to be sure to free rtp here */ + if (p->rtp) { + ast_rtp_destroy(p->rtp); + } ast_mutex_destroy(&p->lock); if (p->chanvars) { ast_variables_destroy(p->chanvars); |