[asterisk-commits] oej: branch 1.4 r64974 -
/branches/1.4/channels/chan_sip.c
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Fri May 18 03:37:45 MST 2007
Author: oej
Date: Fri May 18 05:37:44 2007
New Revision: 64974
URL: http://svn.digium.com/view/asterisk?view=rev&rev=64974
Log:
Issue 9487 - stop media flows at hangup of call
Modified:
branches/1.4/channels/chan_sip.c
Modified: branches/1.4/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=64974&r1=64973&r2=64974
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Fri May 18 05:37:44 2007
@@ -1291,6 +1291,7 @@
char **m_buf, size_t *m_size, char **a_buf, size_t *a_size,
int debug);
static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p);
+static void stop_media_flows(struct sip_pvt *p);
/*--- Authentication stuff */
static int reply_digest(struct sip_pvt *p, struct sip_request *req, char *header, int sipmethod, char *digest, int digest_len);
@@ -3367,6 +3368,8 @@
if (option_debug > 3)
ast_log(LOG_DEBUG, "Hanging up channel in state %s (not UP)\n", ast_state2str(ast->_state));
}
+
+ stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
/* Disconnect */
if (p->vad)
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