[svn-commits] rizzo: branch 1.4 r44665 -
/branches/1.4/channels/chan_sip.c
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Sat Oct 7 04:37:31 MST 2006
Author: rizzo
Date: Sat Oct 7 06:37:30 2006
New Revision: 44665
URL: http://svn.digium.com/view/asterisk?rev=44665&view=rev
Log:
merge revision 44664 - put common code in a function to avoid repetitions.
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?rev=44665&r1=44664&r2=44665&view=diff
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Sat Oct 7 06:37:30 2006
@@ -11791,6 +11791,18 @@
return 1;
}
+/*! \brief Immediately stop RTP, VRTP and UDPTL as applicable */
+static void stop_data_flows(struct sip_pvt *p)
+{
+ /* Immediately stop RTP, VRTP and UDPTL as applicable */
+ if (p->rtp)
+ ast_rtp_stop(p->rtp);
+ if (p->vrtp)
+ ast_rtp_stop(p->vrtp);
+ if (p->udptl)
+ ast_udptl_stop(p->udptl);
+}
+
/*! \brief Handle SIP response in dialogue */
/* XXX only called by handle_request */
static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int ignore, int seqno)
@@ -11980,18 +11992,9 @@
if ((option_verbose > 2) && (resp != 487))
ast_verbose(VERBOSE_PREFIX_3 "Got SIP response %d \"%s\" back from %s\n", resp, rest, ast_inet_ntoa(p->sa.sin_addr));
ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
- if (p->rtp) {
- /* Immediately stop RTP */
- ast_rtp_stop(p->rtp);
- }
- if (p->vrtp) {
- /* Immediately stop VRTP */
- ast_rtp_stop(p->vrtp);
- }
- if (p->udptl) {
- /* Immediately stop UDPTL */
- ast_udptl_stop(p->udptl);
- }
+
+ stop_data_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
+
/* XXX Locking issues?? XXX */
switch(resp) {
case 300: /* Multiple Choices */
@@ -13721,18 +13724,8 @@
ast_log(LOG_DEBUG, "Got CANCEL on an answered call. Ignoring... \n");
return 0;
}
- if (p->rtp) {
- /* Immediately stop RTP */
- ast_rtp_stop(p->rtp);
- }
- if (p->vrtp) {
- /* Immediately stop VRTP */
- ast_rtp_stop(p->vrtp);
- }
- if (p->udptl) {
- /* Immediately stop UDPTL */
- ast_udptl_stop(p->udptl);
- }
+ stop_data_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
+
if (p->owner)
ast_queue_hangup(p->owner);
else
@@ -13753,7 +13746,6 @@
struct ast_channel *c=NULL;
int res;
struct ast_channel *bridged_to;
- char *audioqos = NULL, *videoqos = NULL;
if (p->pendinginvite && !ast_test_flag(&p->flags[0], SIP_OUTGOING) && !ast_test_flag(req, SIP_PKT_IGNORE))
transmit_response_reliable(p, "487 Request Terminated", &p->initreq);
@@ -13762,35 +13754,27 @@
check_via(p, req);
ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
- if (p->rtp)
- audioqos = ast_rtp_get_quality(p->rtp);
- if (p->vrtp)
- videoqos = ast_rtp_get_quality(p->vrtp);
-
/* Get RTCP quality before end of call */
- if (recordhistory) {
- if (p->rtp)
- append_history(p, "RTCPaudio", "Quality:%s", audioqos);
- if (p->vrtp)
- append_history(p, "RTCPvideo", "Quality:%s", videoqos);
- }
-
+ if (recordhistory || p->owner) {
+ char *audioqos, *videoqos;
if (p->rtp) {
+ audioqos = ast_rtp_get_quality(p->rtp);
+ if (recordhistory)
+ append_history(p, "RTCPaudio", "Quality:%s", audioqos);
if (p->owner)
pbx_builtin_setvar_helper(p->owner, "RTPAUDIOQOS", audioqos);
- /* Immediately stop RTP */
- ast_rtp_stop(p->rtp);
}
if (p->vrtp) {
+ videoqos = ast_rtp_get_quality(p->vrtp);
+ if (recordhistory)
+ append_history(p, "RTCPvideo", "Quality:%s", videoqos);
if (p->owner)
pbx_builtin_setvar_helper(p->owner, "RTPVIDEOQOS", videoqos);
- /* Immediately stop VRTP */
- ast_rtp_stop(p->vrtp);
- }
- if (p->udptl) {
- /* Immediately stop UDPTL */
- ast_udptl_stop(p->udptl);
- }
+ }
+ }
+
+ stop_data_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
+
if (!ast_strlen_zero(get_header(req, "Also"))) {
ast_log(LOG_NOTICE, "Client '%s' using deprecated BYE/Also transfer method. Ask vendor to support REFER instead\n",
ast_inet_ntoa(p->recv.sin_addr));
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