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| author | mmichelson <mmichelson@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-06-23 15:16:06 +0000 |
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| committer | mmichelson <mmichelson@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-06-23 15:16:06 +0000 |
| commit | f1e5c6a675a2e4adfe6030f86b7d1cea06dca3b0 (patch) | |
| tree | 44cae28ecf7055b270e6e18edc7f0baac86f0198 | |
| parent | 7c29020182660b6a2872e3f9aaa27a383773f744 (diff) | |
Recorded merge of revisions 202574 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r202574 | mmichelson | 2009-06-23 10:11:47 -0500 (Tue, 23 Jun 2009) | 8 lines
Blocked revisions 202572 via svnmerge
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r202572 | mmichelson | 2009-06-23 10:08:27 -0500 (Tue, 23 Jun 2009) | 3 lines
Fix potential memory leak in chan_sip when video rtp is not allocated properly.
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@202577 f38db490-d61c-443f-a65b-d21fe96a405b
| -rw-r--r-- | channels/chan_sip.c | 6 |
1 files changed, 6 insertions, 0 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 38e447720..fa5374318 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -6097,6 +6097,12 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si ast_log(LOG_WARNING, "Unable to create RTP audio %s%ssession: %s\n", ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) ? "and video " : "", ast_test_flag(&p->flags[1], SIP_PAGE2_TEXTSUPPORT) ? "and text " : "", strerror(errno)); + if (p->rtp) { + ast_rtp_destroy(p->rtp); + } + if (p->vrtp) { + ast_rtp_destroy(p->vrtp); + } ast_mutex_destroy(&p->pvt_lock); if (p->chanvars) { ast_variables_destroy(p->chanvars); |