Add rtpdest option to SIP CHANNEL() dialplan function to return the IP address...

Add rtpdest option to SIP CHANNEL() dialplan function to return the IP address and port that RTP (be it audio/video/text) is going to.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@71988 65c4cc65-6c06-0410-ace0-fbb531ad65f3

CHANGES

@@ -59,6 +59,7 @@ SIP changes
      required due to the restructuring of how MWI is handled.  See the descriptions 
      in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf 
      for more information.
+  * Added rtpdest option to CHANNEL() dialplan function.
 
 IAX2 changes
 ------------

channels/chan_sip.c

@@ -15001,9 +15001,9 @@ static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req)
 static int acf_channel_read(struct ast_channel *chan, const char *funcname, char *preparse, char *buf, size_t buflen)
 {
-	struct ast_rtp_quality qos;
 	struct sip_pvt *p = chan->tech_pvt;
 	char *all = "", *parse = ast_strdupa(preparse);
+	int res = 0;
 	AST_DECLARE_APP_ARGS(args,
 		AST_APP_ARG(param);
 		AST_APP_ARG(type);
@@ -15017,51 +15017,68 @@ static int acf_channel_read(struct ast_channel *chan, const char *funcname, char
 		return 0;
 	}
-	if (ast_strlen_zero(args.param) || strcasecmp(args.param, "rtpqos"))
-		return -1;
-	/* Default arguments of audio,all */
-	if (ast_strlen_zero(args.type))
-		args.type = "audio";
-	if (ast_strlen_zero(args.field))
-		args.field = "all";
-	memset(buf, 0, buflen);
-	memset(&qos, 0, sizeof(qos));
-
-	if (strcasecmp(args.type, "AUDIO") == 0) {
-		all = ast_rtp_get_quality(p->rtp, &qos);
-	} else if (strcasecmp(args.type, "VIDEO") == 0) {
-		all = ast_rtp_get_quality(p->vrtp, &qos);
-	} else if (strcasecmp(args.type, "TEXT") == 0) {
-		all = ast_rtp_get_quality(p->trtp, &qos);
-	}
-
-	if (strcasecmp(args.field, "local_ssrc") == 0)
-		snprintf(buf, buflen, "%u", qos.local_ssrc);
-	else if (strcasecmp(args.field, "local_lostpackets") == 0)
-		snprintf(buf, buflen, "%u", qos.local_lostpackets);
-	else if (strcasecmp(args.field, "local_jitter") == 0)
-		snprintf(buf, buflen, "%.0lf", qos.local_jitter * 1000.0);
-	else if (strcasecmp(args.field, "local_count") == 0)
-		snprintf(buf, buflen, "%u", qos.local_count);
-	else if (strcasecmp(args.field, "remote_ssrc") == 0)
-		snprintf(buf, buflen, "%u", qos.remote_ssrc);
-	else if (strcasecmp(args.field, "remote_lostpackets") == 0)
-		snprintf(buf, buflen, "%u", qos.remote_lostpackets);
-	else if (strcasecmp(args.field, "remote_jitter") == 0)
-		snprintf(buf, buflen, "%.0lf", qos.remote_jitter * 1000.0);
-	else if (strcasecmp(args.field, "remote_count") == 0)
-		snprintf(buf, buflen, "%u", qos.remote_count);
-	else if (strcasecmp(args.field, "rtt") == 0)
-		snprintf(buf, buflen, "%.0lf", qos.rtt * 1000.0);
-	else if (strcasecmp(args.field, "all") == 0)
-		ast_copy_string(buf, all, buflen);
-	else {
-		ast_log(LOG_WARNING, "Unrecognized argument '%s' to %s\n", preparse, funcname);
-		return -1;
+	memset(buf, 0, buflen);
+	if (!strcasecmp(args.param, "rtpdest")) {
+		struct sockaddr_in sin;
+		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);
+
+		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;
+		
+		memset(&qos, 0, sizeof(qos));
+
+		if (ast_strlen_zero(args.type))
+			args.type = "audio";
+		if (ast_strlen_zero(args.field))
+			args.field = "all";
+		
+		if (strcasecmp(args.type, "AUDIO") == 0) {
+			all = ast_rtp_get_quality(p->rtp, &qos);
+		} else if (strcasecmp(args.type, "VIDEO") == 0) {
+			all = ast_rtp_get_quality(p->vrtp, &qos);
+		} else if (strcasecmp(args.type, "TEXT") == 0) {
+			all = ast_rtp_get_quality(p->trtp, &qos);
+		}
+		
+		if (strcasecmp(args.field, "local_ssrc") == 0)
+			snprintf(buf, buflen, "%u", qos.local_ssrc);
+		else if (strcasecmp(args.field, "local_lostpackets") == 0)
+			snprintf(buf, buflen, "%u", qos.local_lostpackets);
+		else if (strcasecmp(args.field, "local_jitter") == 0)
+			snprintf(buf, buflen, "%.0lf", qos.local_jitter * 1000.0);
+		else if (strcasecmp(args.field, "local_count") == 0)
+			snprintf(buf, buflen, "%u", qos.local_count);
+		else if (strcasecmp(args.field, "remote_ssrc") == 0)
+			snprintf(buf, buflen, "%u", qos.remote_ssrc);
+		else if (strcasecmp(args.field, "remote_lostpackets") == 0)
+			snprintf(buf, buflen, "%u", qos.remote_lostpackets);
+		else if (strcasecmp(args.field, "remote_jitter") == 0)
+			snprintf(buf, buflen, "%.0lf", qos.remote_jitter * 1000.0);
+		else if (strcasecmp(args.field, "remote_count") == 0)
+			snprintf(buf, buflen, "%u", qos.remote_count);
+		else if (strcasecmp(args.field, "rtt") == 0)
+			snprintf(buf, buflen, "%.0lf", qos.rtt * 1000.0);
+		else if (strcasecmp(args.field, "all") == 0)
+			ast_copy_string(buf, all, buflen);
+		else {
+			ast_log(LOG_WARNING, "Unrecognized argument '%s' to %s\n", preparse, funcname);
+			return -1;
+		}
+	} else {
+		res = -1;
 	}
-	return 0;
+	return res;
 }
 /*! \brief Handle incoming BYE request */

funcs/func_channel.c

@@ -181,6 +181,11 @@ static struct ast_custom_function channel_function = {
 		"    remote_count          Number of transmitted packets\n"
 		"    rtt                   Round trip time\n"
 		"    all                   All statistics (in a form suited to logging, but not for parsing)\n"
+	        "R/O    rtpdest            Get remote RTP destination information\n"
+	        "       This option takes one additional argument:\n"
+                "  Argument 1:\n"
+                "    audio                 Get audio destination\n"
+                "    video                 Get video destination\n"
 		"\n"
 		"chan_iax2 provides the following additional options:\n"
 		"R/W    osptoken           Get or set the OSP token information for a call\n"