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

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 1930e3cd21b6961a39140f86182e6b6cc3e1cb1a..9d73f372cf3428fed8e7c6ca5321ec4c807aea16 100644 (file)
--- 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);
        }