Index: rtp.c
===================================================================
RCS file: /usr/cvsroot/asterisk/rtp.c,v
retrieving revision 1.129
diff -u -r1.129 rtp.c
--- rtp.c	15 May 2005 04:06:16 -0000	1.129
+++ rtp.c	16 May 2005 10:24:01 -0000
@@ -212,6 +212,9 @@
 	return f;
 }
 
+/* process_rfc2833: Process RTP DTMF and events according to RFC 2833:
+	"RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals"
+*/
 static struct ast_frame *process_rfc2833(struct ast_rtp *rtp, unsigned char *data, int len)
 {
 	unsigned int event;
@@ -226,9 +229,8 @@
 	event_end >>= 24;
 	duration = ntohl(*((unsigned int *)(data)));
 	duration &= 0xFFFF;
-#if 0
-	printf("Event: %08x (len = %d)\n", event, len);
-#endif
+	if (rtpdebug)
+		ast_log(LOG_DEBUG, "- RTP 2833 Event: %08x (len = %d)\n", event, len);
 	if (event < 10) {
 		resp = '0' + event;
 	} else if (event < 11) {
@@ -237,23 +239,19 @@
 		resp = '#';
 	} else if (event < 16) {
 		resp = 'A' + (event - 12);
-	} else if (event < 17) {
-		resp = 'X';
+	} else if (event < 17) {	/* Event 16: Hook flash */
+		resp = 'X';	
 	}
 	if (rtp->resp && (rtp->resp != resp)) {
 		f = send_dtmf(rtp);
-	}
-	else if(event_end & 0x80)
-	{
+	} else if(event_end & 0x80) {
 		if (rtp->resp) {
 			f = send_dtmf(rtp);
 			rtp->resp = 0;
 		}
 		resp = 0;
 		duration = 0;
-	}
-	else if(rtp->dtmfduration && (duration < rtp->dtmfduration))
-	{
+	} else if(rtp->dtmfduration && (duration < rtp->dtmfduration)) {
 		f = send_dtmf(rtp);
 	}
 	if (!(event_end & 0x80))
@@ -263,17 +261,20 @@
 	return f;
 }
 
+/*--- process_rfc3389: Process Comfort Noice RTP. 
+	This is incomplete at the moment.
+*/
 static struct ast_frame *process_rfc3389(struct ast_rtp *rtp, unsigned char *data, int len)
 {
 	struct ast_frame *f = NULL;
 	/* Convert comfort noise into audio with various codecs.  Unfortunately this doesn't
 	   totally help us out becuase we don't have an engine to keep it going and we are not
 	   guaranteed to have it every 20ms or anything */
-#if 1
-	printf("RFC3389: %d bytes, level %d...\n", len, rtp->lastrxformat);
-#endif	
+	if (rtpdebug)
+		ast_log(LOG_DEBUG, "- RTP 3389 Comfort noice event: Level %d (len = %d)\n", rtp->lastrxformat, len);
 	if (!(rtp->flags & FLAG_3389_WARNING)) {
-		ast_log(LOG_NOTICE, "RFC3389 support incomplete.  Turn off on client if possible\n");
+		char iabuf[INET_ADDRSTRLEN];
+		ast_log(LOG_NOTICE, "Comfort noice support incomplete in Asterisk (RFC 3389).  Please turn off on client if possible. Client IP: %s\n", ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr));
 		rtp->flags |= FLAG_3389_WARNING;
 	}
 	/* Must have at least one byte */
@@ -902,7 +903,7 @@
 		if (x > rtpend)
 			x = (rtpstart + 1) & ~1;
 		if (x == startplace) {
-			ast_log(LOG_ERROR, "No RTP ports remaining\n");
+			ast_log(LOG_ERROR, "No RTP ports remaining. Can't setup media stream for this call.\n");
 			close(rtp->s);
 			if (rtp->rtcp) {
 				close(rtp->rtcp->s);
@@ -1413,6 +1414,9 @@
 	return NULL;
 }
 
+/* ast_rtp_bridge: Bridge calls. If possible and allowed, initiate
+	re-invite so the peers exchange media directly outside 
+	of Asterisk. */
 int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc)
 {
 	struct ast_frame *f;
@@ -1485,14 +1489,16 @@
 	else
 		codec1 = 0;
 	if (pr0->get_codec && pr1->get_codec) {
-		/* Hey, we can't do reinvite if both parties speak diffrent codecs */
+		/* Hey, we can't do reinvite if both parties speak different codecs */
 		if (!(codec0 & codec1)) {
-			ast_log(LOG_WARNING, "codec0 = %d is not codec1 = %d, cannot native bridge.\n",codec0,codec1);
+			if (option_debug)
+				ast_log(LOG_DEBUG, "Channel codec0 = %d is not codec1 = %d, cannot native bridge in RTP.\n", codec0, codec1);
 			ast_mutex_unlock(&c0->lock);
 			ast_mutex_unlock(&c1->lock);
 			return -2;
 		}
 	}
+	/* Ok, we should be able to redirect the media. Start with one channel */
 	if (pr0->set_rtp_peer(c0, p1, vp1, codec1)) 
 		ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", c0->name, c1->name);
 	else {
@@ -1501,6 +1507,7 @@
 		if (vp1)
 			ast_rtp_get_peer(vp1, &vac1);
 	}
+	/* Then test the other channel */
 	if (pr1->set_rtp_peer(c1, p0, vp0, codec0))
 		ast_log(LOG_WARNING, "Channel '%s' failed to talk back to '%s'\n", c1->name, c0->name);
 	else {
@@ -1577,7 +1584,7 @@
 		if (!who) {
 			if (option_debug)
 				ast_log(LOG_DEBUG, "Ooh, empty read...\n");
-			/* check for hagnup / whentohangup */
+			/* check for hangup / whentohangup */
 			if (ast_check_hangup(c0) || ast_check_hangup(c1))
 				break;
 			continue;
@@ -1729,7 +1736,7 @@
 		ast_config_destroy(cfg);
 	}
 	if (rtpstart >= rtpend) {
-		ast_log(LOG_WARNING, "Unreasonable values for RTP start in rtp.conf/end\n");
+		ast_log(LOG_WARNING, "Unreasonable values for RTP start/end port in rtp.conf\n");
 		rtpstart = 5000;
 		rtpend = 31000;
 	}
Index: configs/rtp.conf.sample
===================================================================
RCS file: /usr/cvsroot/asterisk/configs/rtp.conf.sample,v
retrieving revision 1.2
diff -u -r1.2 rtp.conf.sample
--- configs/rtp.conf.sample	17 Jul 2004 21:08:12 -0000	1.2
+++ configs/rtp.conf.sample	16 May 2005 10:24:01 -0000
@@ -5,6 +5,8 @@
 ;
 ; RTP start and RTP end configure start and end addresses
 ;
+; Defaults are rtpstart=5000 and rtpend=31000
+;
 rtpstart=10000
 rtpend=20000
 ;