[asterisk-commits] mmichelson: branch 1.6.1 r254541 - /branches/1.6.1/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: mmichelson
Date: Thu Mar 25 12:10:59 2010
New Revision: 254541

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=254541
Log:
Fix potential crashes from trying to reference non-existent RTP streams.


Modified:
    branches/1.6.1/channels/chan_sip.c

Modified: branches/1.6.1/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.1/channels/chan_sip.c?view=diff&rev=254541&r1=254540&r2=254541
==============================================================================
--- branches/1.6.1/channels/chan_sip.c (original)
+++ branches/1.6.1/channels/chan_sip.c Thu Mar 25 12:10:59 2010
@@ -20023,19 +20023,26 @@
 		ast_copy_string(buf, (p->t38.state == T38_DISABLED) ? "0" : "1", buflen);
 	} else if (!strcasecmp(args.param, "rtpdest")) {
 		struct sockaddr_in sin;
+		struct ast_rtp *stream;
 
 		if (ast_strlen_zero(args.type))
 			args.type = "audio";
 
-		if (!strcasecmp(args.type, "audio"))
-			ast_rtp_get_peer(p->rtp, &sin);
-		else if (!strcasecmp(args.type, "video"))
-			ast_rtp_get_peer(p->vrtp, &sin);
-		else if (!strcasecmp(args.type, "text"))
-			ast_rtp_get_peer(p->trtp, &sin);
-		else
+		if (!strcasecmp(args.type, "audio")) {
+			stream = p->rtp;
+		} else if (!strcasecmp(args.type, "video")) {
+			stream = p->vrtp;
+		} else if (!strcasecmp(args.type, "text")) {
+			stream = p->trtp;
+		} else {
 			return -1;
-
+		}
+
+		if (!stream) {
+			return -1;
+		}
+
+		ast_rtp_get_peer(stream, &sin);
 		snprintf(buf, buflen, "%s:%d", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
 	} else if (!strcasecmp(args.param, "rtpqos")) {
 		struct ast_rtp_quality qos;