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merge revision 44664 - put common code in a function to avoid repetitions.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44665 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Luigi Rizzo
2006-10-07 11:37:30 +00:00
parent 169342bc0d
commit f480d547cc
1 changed files with 28 additions and 44 deletions
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72
channels/chan_sip.c
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@@ -11791,6 +11791,18 @@ static int handle_response_peerpoke(struct sip_pvt *p, int resp, struct sip_requ
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 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_
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 @@ static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req)
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 @@ static int handle_request_bye(struct sip_pvt *p, struct sip_request *req)
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 @@ static int handle_request_bye(struct sip_pvt *p, struct sip_request *req)
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));