[asterisk-commits] russell: branch 1.4 r59207 - in /branches/1.4:
channels/ include/asterisk/ main/
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Mon Mar 26 10:45:56 MST 2007
Author: russell
Date: Mon Mar 26 12:45:55 2007
New Revision: 59207
URL: http://svn.digium.com/view/asterisk?view=rev&rev=59207
Log:
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)
Modified:
branches/1.4/channels/chan_sip.c
branches/1.4/include/asterisk/rtp.h
branches/1.4/main/rtp.c
Modified: branches/1.4/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=59207&r1=59206&r2=59207
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Mon Mar 26 12:45:55 2007
@@ -3392,9 +3392,9 @@
char *audioqos = "";
char *videoqos = "";
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);
/* Send a hangup */
transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
@@ -14127,6 +14127,53 @@
}
}
+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);
+ }
+
+ 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)
{
@@ -14148,14 +14195,14 @@
if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY) || p->owner) {
char *audioqos, *videoqos;
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)
@@ -17321,6 +17368,44 @@
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,
+};
+
/*! \brief PBX load module - initialization */
static int load_module(void)
{
@@ -17370,6 +17455,8 @@
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);
/* Register manager commands */
ast_manager_register2("SIPpeers", EVENT_FLAG_SYSTEM, manager_sip_show_peers,
@@ -17399,6 +17486,8 @@
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);
/* Unregister dial plan applications */
ast_unregister_application(app_dtmfmode);
Modified: branches/1.4/include/asterisk/rtp.h
URL: http://svn.digium.com/view/asterisk/branches/1.4/include/asterisk/rtp.h?view=diff&rev=59207&r1=59206&r2=59207
==============================================================================
--- branches/1.4/include/asterisk/rtp.h (original)
+++ branches/1.4/include/asterisk/rtp.h Mon Mar 26 12:45:55 2007
@@ -77,6 +77,18 @@
AST_LIST_ENTRY(ast_rtp_protocol) list;
};
+struct ast_rtp_quality {
+ unsigned int local_ssrc; /* Our SSRC */
+ unsigned int local_lostpackets; /* Our lost packets */
+ double local_jitter; /* Our calculated jitter */
+ unsigned int local_count; /* Number of received packets */
+ unsigned int remote_ssrc; /* Their SSRC */
+ unsigned int remote_lostpackets; /* Their lost packets */
+ double remote_jitter; /* Their reported jitter */
+ unsigned int remote_count; /* Number of transmitted packets */
+ double rtt; /* Round trip time */
+};
+
#define FLAG_3389_WARNING (1 << 0)
@@ -206,7 +218,7 @@
void ast_rtp_stop(struct ast_rtp *rtp);
/*! \brief Return RTCP quality string */
-char *ast_rtp_get_quality(struct ast_rtp *rtp);
+char *ast_rtp_get_quality(struct ast_rtp *rtp, struct ast_rtp_quality *qual);
/*! \brief Send an H.261 fast update request. Some devices need this rather than the XML message in SIP */
int ast_rtcp_send_h261fur(void *data);
Modified: branches/1.4/main/rtp.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/main/rtp.c?view=diff&rev=59207&r1=59206&r2=59207
==============================================================================
--- branches/1.4/main/rtp.c (original)
+++ branches/1.4/main/rtp.c Mon Mar 26 12:45:55 2007
@@ -2024,7 +2024,7 @@
rtp->rxseqno = 0;
}
-char *ast_rtp_get_quality(struct ast_rtp *rtp)
+char *ast_rtp_get_quality(struct ast_rtp *rtp, struct ast_rtp_quality *qual)
{
/*
*ssrc our ssrc
@@ -2035,8 +2035,20 @@
*txjitter reported jitter of the other end
*txcount transmitted packets
*rlp remote lost packets
+ *rtt round trip time
*/
-
+
+ if (qual) {
+ qual->local_ssrc = rtp->ssrc;
+ qual->local_lostpackets = rtp->rtcp->expected_prior - rtp->rtcp->received_prior;
+ qual->local_jitter = rtp->rxjitter;
+ qual->local_count = rtp->rxcount;
+ qual->remote_ssrc = rtp->themssrc;
+ qual->remote_lostpackets = rtp->rtcp->reported_lost;
+ qual->remote_jitter = rtp->rtcp->reported_jitter / 65536.0;
+ qual->remote_count = rtp->txcount;
+ qual->rtt = rtp->rtcp->rtt;
+ }
snprintf(rtp->rtcp->quality, sizeof(rtp->rtcp->quality), "ssrc=%u;themssrc=%u;lp=%u;rxjitter=%f;rxcount=%u;txjitter=%f;txcount=%u;rlp=%u;rtt=%f", rtp->ssrc, rtp->themssrc, rtp->rtcp->expected_prior - rtp->rtcp->received_prior, rtp->rxjitter, rtp->rxcount, (double)rtp->rtcp->reported_jitter/65536., rtp->txcount, rtp->rtcp->reported_lost, rtp->rtcp->rtt);
return rtp->rtcp->quality;
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