diff options
| author | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-06-09 21:28:52 +0000 |
|---|---|---|
| committer | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-06-09 21:28:52 +0000 |
| commit | 3c809bb1a34adc3df0f8d401d13bf1627f9fd9a0 (patch) | |
| tree | 643ca9ba40175ae8a45a353ba92d2e63b869c550 /channels | |
| parent | bf45417f7149b5c08b40a09d47271b410d9c2d90 (diff) | |
Store RTCP reports in channel variables and SIP history
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@33374 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels')
| -rw-r--r-- | channels/chan_sip.c | 29 |
1 files changed, 25 insertions, 4 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 1930e3cd2..9d73f372c 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -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); } |