Improve debugging of RTP ports (bug #2131, heavily modified)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@3548 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 24777946ea3faa02c9d7a94ee9dc91d295300880..756e11a9be596f23fede841c12e6b6fe9a0961ec 100755 (executable)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -2682,17 +2682,25 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
        /* RTP addresses and ports for audio and video */
        sin.sin_family = AF_INET;
        memcpy(&sin.sin_addr, hp->h_addr, sizeof(sin.sin_addr));
+
        /* Setup audio port number */
        sin.sin_port = htons(portno);
-       if (p->rtp && sin.sin_port)
+       if (p->rtp && sin.sin_port) {
                ast_rtp_set_peer(p->rtp, &sin);
+               if (debug) {
+                       ast_verbose("Peer audio RTP is at port %s:%d\n", ast_inet_ntoa(iabuf,sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port));
+                       ast_log(LOG_DEBUG,"Peer audio RTP is at port %s:%d\n",ast_inet_ntoa(iabuf, sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port));
+               }
+       }
        /* Setup video port number */
        sin.sin_port = htons(vportno);
-       if (p->vrtp && sin.sin_port)
+       if (p->vrtp && sin.sin_port) {
                ast_rtp_set_peer(p->vrtp, &sin);
-
-       if (sipdebug)
-               ast_verbose("Peer RTP is at port %s:%d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port));
+               if (debug) {
+                       ast_verbose("Peer video RTP is at port %s:%d\n", ast_inet_ntoa(iabuf,sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port));
+                       ast_log(LOG_DEBUG,"Peer video RTP is at port %s:%d\n",ast_inet_ntoa(iabuf, sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port));
+               }
+       }
 
        /* Next, scan through each "a=rtpmap:" line, noting each
         * specified RTP payload type (with corresponding MIME subtype):