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Store RTCP reports in channel variables and SIP history 

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@33374 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Olle Johansson 2006-06-09 21:28:52 +00:00
parent 6d4ab87321
commit d2fa9efdad
1 changed files with 25 additions and 4 deletions
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29
channels/chan_sip.c
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@ -2966,16 +2966,26 @@ static int sip_hangup(struct ast_channel *ast)
}
} else { /* Call is in UP state, send BYE */
if (!p->pendinginvite) {
char *audioqos = "";
char *videoqos = "";
if (p->rtp)
audioqos = ast_rtp_get_quality(p->rtp);
if (p->vrtp)
videoqos = ast_rtp_get_quality(p->vrtp);
/* Send a hangup */
transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
/* Get RTCP quality before end of call */
if (recordhistory) {
if (p->rtp)
append_history(p, "RTCPaudio", "Quality:%s", ast_rtp_get_quality(p->rtp));
append_history(p, "RTCPaudio", "Quality:%s", audioqos);
if (p->vrtp)
append_history(p, "RTCPvideo", "Quality:%s", ast_rtp_get_quality(p->vrtp));
append_history(p, "RTCPvideo", "Quality:%s", videoqos);
}
if (p->rtp)
pbx_builtin_setvar_helper(p->owner, "RTPAUDIOQOS", audioqos);
if (p->vrtp)
pbx_builtin_setvar_helper(p->owner, "RTPVIDEOQOS", videoqos);
} else {
/* Note we will need a BYE when this all settles out
but we can't send one while we have "INVITE" outstanding. */
@ -12629,6 +12639,7 @@ static int handle_request_bye(struct sip_pvt *p, struct sip_request *req)
int res;
struct ast_channel *bridged_to;
char iabuf[INET_ADDRSTRLEN];
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);
@ -12637,18 +12648,28 @@ 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", ast_rtp_get_quality(p->rtp));
append_history(p, "RTCPaudio", "Quality:%s", audioqos);
if (p->vrtp)
append_history(p, "RTCPvideo", "Quality:%s", ast_rtp_get_quality(p->vrtp));
append_history(p, "RTCPvideo", "Quality:%s", videoqos);
}
if (p->rtp) {
if (p->owner)
pbx_builtin_setvar_helper(p->owner, "RTPAUDIOQOS", audioqos);
/* Immediately stop RTP */
ast_rtp_stop(p->rtp);
}
if (p->vrtp) {
if (p->owner)
pbx_builtin_setvar_helper(p->owner, "RTPVIDEOQOS", videoqos);
/* Immediately stop VRTP */
ast_rtp_stop(p->vrtp);
}