put common code in a function to avoid repetitions.
authorLuigi Rizzo <rizzo@icir.org>
Sat, 7 Oct 2006 11:28:38 +0000 (11:28 +0000)
committerLuigi Rizzo <rizzo@icir.org>
Sat, 7 Oct 2006 11:28:38 +0000 (11:28 +0000)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44664 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

index 629cf77..0f4a1e5 100644 (file)
@@ -11766,6 +11766,18 @@ 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)
@@ -11955,18 +11967,9 @@ 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 */
@@ -13696,18 +13699,8 @@ 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
@@ -13728,7 +13721,6 @@ 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);
@@ -13737,35 +13729,27 @@ 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);
+               }
+               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);
+               }
        }
 
-       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);
-       }
-       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));