[svn-commits] mmichelson: branch 1.6.0 r202577 - /branches/1.6.0/channels/chan_sip.c

[SVN commits to the Digium repositories]

Author: mmichelson
Date: Tue Jun 23 10:16:06 2009
New Revision: 202577

URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=202577
Log:
Recorded merge of revisions 202574 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

................
  r202574 | mmichelson | 2009-06-23 10:11:47 -0500 (Tue, 23 Jun 2009) | 8 lines
  
  Blocked revisions 202572 via svnmerge
  
  ........
    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.
  ........
................

Modified:
    branches/1.6.0/channels/chan_sip.c

Modified: branches/1.6.0/channels/chan_sip.c
URL: http://svn.asterisk.org/svn-view/asterisk/branches/1.6.0/channels/chan_sip.c?view=diff&rev=202577&r1=202576&r2=202577
==============================================================================
--- branches/1.6.0/channels/chan_sip.c (original)
+++ branches/1.6.0/channels/chan_sip.c Tue Jun 23 10:16:06 2009
@@ -6097,6 +6097,12 @@
  			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);