[Asterisk-cvs] asterisk/channels chan_sip.c,1.464,1.465

[markster at lists.digium.com]

Update of /usr/cvsroot/asterisk/channels
In directory localhost.localdomain:/tmp/cvs-serv3242/channels

Modified Files:
	chan_sip.c 
Log Message:
Improve debugging of RTP ports (bug #2131, heavily modified)


Index: chan_sip.c
===================================================================
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.464
retrieving revision 1.465
diff -u -d -r1.464 -r1.465
--- chan_sip.c	29 Jul 2004 18:58:57 -0000	1.464
+++ chan_sip.c	31 Jul 2004 02:31:24 -0000	1.465
@@ -2682,17 +2682,25 @@
 	/* 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):