Merged revisions 59207 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r59207 | russell | 2007-03-26 12:45:55 -0500 (Mon, 26 Mar 2007) | 7 lines

The AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in some
because they get set in sip_hangup.  So, there are common situations where
the variables will not be available in the dialplan at all.  So, this patch
provides an alternate method for getting to this information by introducing
AUDIORTPQOS and VIDEORTPQOS dialplan functions.
(issue #9370, patch by Corydon76, with some testing by blitzrage)

........


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@59208 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 117 insertions, 5 deletions

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 07ba786be..b054465a2 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -3686,11 +3686,11 @@ static int sip_hangup(struct ast_channel *ast)
 				char *videoqos = "";
 				char *textqos = "";
 				if (p->rtp)
-					audioqos = ast_rtp_get_quality(p->rtp);
+					audioqos = ast_rtp_get_quality(p->rtp, NULL);
 				if (p->vrtp)
-					videoqos = ast_rtp_get_quality(p->vrtp);
+					videoqos = ast_rtp_get_quality(p->vrtp, NULL);
 				if (p->trtp)
-					textqos = ast_rtp_get_quality(p->trtp);
+					textqos = ast_rtp_get_quality(p->trtp, NULL);
 				/* Send a hangup */
 				transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
 
@@ -14800,6 +14800,55 @@ static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req)
 	}
 }
 
+static int acf_audiortpqos_read(struct ast_channel *chan, char *funcname, char *args, char *buf, size_t buflen)
+{
+	struct ast_rtp_quality qos;
+	struct sip_pvt *p = chan->tech_pvt;
+	char *all = "";
+
+	/* Sanity check */
+	if (chan->tech != &sip_tech && chan->tech != &sip_tech_info) {
+		ast_log(LOG_ERROR, "Cannot call %s on a non-SIP channel\n", funcname);
+	}
+
+	memset(buf, 0, buflen);
+	memset(&qos, 0, sizeof(qos));
+
+	if (strcmp(funcname, "AUDIORTPQOS") == 0) {
+		all = ast_rtp_get_quality(p->rtp, &qos);
+	} else if (strcmp(funcname, "VIDEORTPQOS") == 0) {
+		all = ast_rtp_get_quality(p->vrtp, &qos);
+	} else if (strcmp(funcname, "TEXTRTPQOS") == 0) {
+		all = ast_rtp_get_quality(p->trtp, &qos);
+	}
+
+	if (strcasecmp(args, "local_ssrc") == 0)
+		snprintf(buf, buflen, "%u", qos.local_ssrc);
+	else if (strcasecmp(args, "local_lostpackets") == 0)
+		snprintf(buf, buflen, "%u", qos.local_lostpackets);
+	else if (strcasecmp(args, "local_jitter") == 0)
+		snprintf(buf, buflen, "%.0lf", qos.local_jitter * 1000.0);
+	else if (strcasecmp(args, "local_count") == 0)
+		snprintf(buf, buflen, "%u", qos.local_count);
+	else if (strcasecmp(args, "remote_ssrc") == 0)
+		snprintf(buf, buflen, "%u", qos.remote_ssrc);
+	else if (strcasecmp(args, "remote_lostpackets") == 0)
+		snprintf(buf, buflen, "%u", qos.remote_lostpackets);
+	else if (strcasecmp(args, "remote_jitter") == 0)
+		snprintf(buf, buflen, "%.0lf", qos.remote_jitter * 1000.0);
+	else if (strcasecmp(args, "remote_count") == 0)
+		snprintf(buf, buflen, "%u", qos.remote_count);
+	else if (strcasecmp(args, "rtt") == 0)
+		snprintf(buf, buflen, "%.0lf", qos.rtt * 1000.0);
+	else if (strcasecmp(args, "all") == 0)
+		ast_copy_string(buf, all, buflen);
+	else {
+		ast_log(LOG_WARNING, "Unrecognized argument '%s' to %s\n", args, funcname);
+		return -1;
+	}
+	return 0;
+}
+
 /*! \brief Handle incoming BYE request */
 static int handle_request_bye(struct sip_pvt *p, struct sip_request *req)
 {
@@ -14823,14 +14872,14 @@ static int handle_request_bye(struct sip_pvt *p, struct sip_request *req)
 	if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY) || p->owner) {
 		char *audioqos, *videoqos, *textqos;
 		if (p->rtp) {
-			audioqos = ast_rtp_get_quality(p->rtp);
+			audioqos = ast_rtp_get_quality(p->rtp, NULL);
 			if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
 				append_history(p, "RTCPaudio", "Quality:%s", audioqos);
 			if (p->owner)
 				pbx_builtin_setvar_helper(p->owner, "RTPAUDIOQOS", audioqos);
 		}
 		if (p->vrtp) {
-			videoqos = ast_rtp_get_quality(p->vrtp);
+			videoqos = ast_rtp_get_quality(p->vrtp, NULL);
 			if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
 				append_history(p, "RTCPvideo", "Quality:%s", videoqos);
 			if (p->owner)
@@ -18081,6 +18130,63 @@ static struct ast_cli_entry cli_sip[] = {
 	sip_reload_usage },
 };
 
+struct ast_custom_function acf_audiortpqos = {
+	.name = "AUDIORTPQOS",
+	.synopsis = "Retrieve statistics about an RTP audio stream",
+	.desc =
+"The following statistics may be retrieved:\n"
+"  local_ssrc         - Local SSRC (stream ID)\n"
+"  local_lostpackets  - Local lost packets\n"
+"  local_jitter       - Local calculated jitter\n"
+"  local_count        - Number of received packets\n"
+"  remote_ssrc        - Remote SSRC (stream ID)\n"
+"  remote_lostpackets - Remote lost packets\n"
+"  remote_jitter      - Remote reported jitter\n"
+"  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)",
+	.syntax = "AUDIORTPQOS(<field>)",
+	.read = acf_audiortpqos_read,
+};
+
+struct ast_custom_function acf_videortpqos = {
+	.name = "VIDEORTPQOS",
+	.synopsis = "Retrieve statistics about an RTP audio stream",
+	.desc =
+"The following statistics may be retrieved:\n"
+"  local_ssrc         - Local SSRC (stream ID)\n"
+"  local_lostpackets  - Local lost packets\n"
+"  local_jitter       - Local calculated jitter\n"
+"  local_count        - Number of received packets\n"
+"  remote_ssrc        - Remote SSRC (stream ID)\n"
+"  remote_lostpackets - Remote lost packets\n"
+"  remote_jitter      - Remote reported jitter\n"
+"  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)",
+	.syntax = "AUDIORTPQOS(<field>)",
+	.read = acf_audiortpqos_read,
+};
+
+struct ast_custom_function acf_textrtpqos = {
+	.name = "TEXTRTPQOS",
+	.synopsis = "Retrieve statistics about an RTP text stream",
+	.desc =
+"The following statistics may be retrieved:\n"
+"  local_ssrc         - Local SSRC (stream ID)\n"
+"  local_lostpackets  - Local lost packets\n"
+"  local_jitter       - Local calculated jitter\n"
+"  local_count        - Number of received packets\n"
+"  remote_ssrc        - Remote SSRC (stream ID)\n"
+"  remote_lostpackets - Remote lost packets\n"
+"  remote_jitter      - Remote reported jitter\n"
+"  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)",
+	.syntax = "TEXTRTPQOS(<field>)",
+	.read = acf_audiortpqos_read,
+};
+
 /*! \brief PBX load module - initialization */
 static int load_module(void)
 {
@@ -18130,6 +18236,9 @@ static int load_module(void)
 	ast_custom_function_register(&sippeer_function);
 	ast_custom_function_register(&sipchaninfo_function);
 	ast_custom_function_register(&checksipdomain_function);
+	ast_custom_function_register(&acf_audiortpqos);
+	ast_custom_function_register(&acf_videortpqos);
+	ast_custom_function_register(&acf_textrtpqos);
 
 	/* Register manager commands */
 	ast_manager_register2("SIPpeers", EVENT_FLAG_SYSTEM, manager_sip_show_peers,
@@ -18159,6 +18268,9 @@ static int unload_module(void)
 	ast_custom_function_unregister(&sippeer_function);
 	ast_custom_function_unregister(&sip_header_function);
 	ast_custom_function_unregister(&checksipdomain_function);
+	ast_custom_function_unregister(&acf_audiortpqos);
+	ast_custom_function_unregister(&acf_videortpqos);
+	ast_custom_function_unregister(&acf_textrtpqos);
 
 	/* Unregister dial plan applications */
 	ast_unregister_application(app_dtmfmode);