[svn-commits] file: branch file/rtp_engine r131167 - /team/file/rtp_engine/channels/chan_sip.c
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Jul 15 19:54:11 CDT 2008
Author: file
Date: Tue Jul 15 19:54:11 2008
New Revision: 131167
URL: http://svn.digium.com/view/asterisk?view=rev&rev=131167
Log:
Add RTCP stats back into chan_sip.
Modified:
team/file/rtp_engine/channels/chan_sip.c
Modified: team/file/rtp_engine/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/file/rtp_engine/channels/chan_sip.c?view=diff&rev=131167&r1=131166&r2=131167
==============================================================================
--- team/file/rtp_engine/channels/chan_sip.c (original)
+++ team/file/rtp_engine/channels/chan_sip.c Tue Jul 15 19:54:11 2008
@@ -726,6 +726,7 @@
static char default_mohsuggest[MAX_MUSICCLASS]; /*!< Global setting for moh class to suggest when putting
* a bridged channel on hold */
static char default_parkinglot[AST_MAX_CONTEXT]; /*!< Parkinglot */
+static char default_engine[256]; /*!< Default RTP engine */
static int default_maxcallbitrate; /*!< Maximum bitrate for call */
static struct ast_codec_pref default_prefs; /*!< Default codec prefs */
@@ -1276,6 +1277,7 @@
AST_STRING_FIELD(rpid_from); /*!< Our RPID From header */
AST_STRING_FIELD(url); /*!< URL to be sent with next message to peer */
AST_STRING_FIELD(parkinglot); /*!< Parkinglot */
+ AST_STRING_FIELD(engine); /*!< RTP engine to use */
);
struct sip_socket socket; /*!< The socket used for this dialog */
unsigned int ocseq; /*!< Current outgoing seqno */
@@ -1503,6 +1505,7 @@
char mohinterpret[MAX_MUSICCLASS];/*!< Music on Hold class */
char mohsuggest[MAX_MUSICCLASS];/*!< Music on Hold class */
char parkinglot[AST_MAX_CONTEXT];/*!< Parkinglot */
+ char engine[256]; /*!< RTP Engine to use */
char useragent[256]; /*!< User agent in SIP request (saved from registration) */
struct ast_codec_pref prefs; /*!< codec prefs */
int lastmsgssent;
@@ -2167,6 +2170,9 @@
static enum st_mode st_get_mode(struct sip_pvt *);
static struct sip_st_dlg* sip_st_alloc(struct sip_pvt *const p);
+/*------- RTP Glue functions -------- */
+static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_rtp_instance *vinstance, struct ast_rtp_instance *tinstance, int codecs, int nat_active);
+
/*! \brief Definition of this channel for PBX channel registration */
static const struct ast_channel_tech sip_tech = {
.type = "SIP",
@@ -4183,6 +4189,7 @@
ast_string_field_set(dialog, fullcontact, peer->fullcontact);
ast_string_field_set(dialog, context, peer->context);
ast_string_field_set(dialog, parkinglot, peer->parkinglot);
+ ast_string_field_set(dialog, engine, peer->engine);
dialog->outboundproxy = obproxy_get(dialog, peer);
dialog->callgroup = peer->callgroup;
dialog->pickupgroup = peer->pickupgroup;
@@ -4859,7 +4866,7 @@
struct sip_pvt *p = ast->tech_pvt;
int needcancel = FALSE;
int needdestroy = 0;
-// struct ast_channel *oldowner = ast;
+ struct ast_channel *oldowner = ast;
if (!p) {
ast_debug(1, "Asked to hangup channel that was not connected\n");
@@ -4983,7 +4990,59 @@
}
if (!p->pendinginvite) {
-// struct ast_channel *bridge = ast_bridged_channel(oldowner);
+ char quality_buf[AST_MAX_USER_FIELD], *quality;
+
+ if (p->rtp && (quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
+ if (p->do_history) {
+ append_history(p, "RTCPaudio", "Quality:%s", quality);
+ }
+ if (oldowner) {
+ struct ast_channel *bridge = ast_bridged_channel(oldowner);
+
+ pbx_builtin_setvar_helper(oldowner, "RTPAUDIOQOS", quality);
+
+ if (bridge) {
+ pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSBRIDGED", quality);
+ }
+
+ if ((quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER, quality_buf, sizeof(quality_buf)))) {
+ pbx_builtin_setvar_helper(oldowner, "RTPAUDIOQOSJITTER", quality);
+ if (bridge) {
+ pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSJITTERBRIDGED", quality);
+ }
+ }
+ if ((quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS, quality_buf, sizeof(quality_buf)))) {
+ pbx_builtin_setvar_helper(oldowner, "RTPAUDIOQOSLOSS", quality);
+ if (bridge) {
+ pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSLOSSBRIDGED", quality);
+ }
+ }
+ if ((quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT, quality_buf, sizeof(quality_buf)))) {
+ pbx_builtin_setvar_helper(oldowner, "RTPAUDIOQOSRTT", quality);
+ if (bridge) {
+ pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSRTTBRIDGED", quality);
+ }
+ }
+ }
+ }
+ if (p->do_history || oldowner) {
+ if (p->vrtp && (quality = ast_rtp_instance_get_quality(p->vrtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
+ if (p->do_history) {
+ append_history(p, "RTCPvideo", "Quality:%s", quality);
+ }
+ if (oldowner) {
+ pbx_builtin_setvar_helper(oldowner, "RTPVIDEOQOS", quality);
+ }
+ }
+ if (p->trtp && (quality = ast_rtp_instance_get_quality(p->trtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
+ if (p->do_history) {
+ append_history(p, "RTCPtext", "Quality:%s", quality);
+ }
+ if (oldowner) {
+ pbx_builtin_setvar_helper(oldowner, "RTCPTEXTQOS", quality);
+ }
+ }
+ }
/* Send a hangup */
transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
@@ -5185,7 +5244,7 @@
redirect of both channels). Note that a channel can not be masqueraded *into*
a native bridge. So there is no danger that this breaks a native bridge that
should stay up. */
-// sip_set_rtp_peer(newchan, NULL, NULL, 0, 0, 0);
+ sip_set_rtp_peer(newchan, NULL, NULL, 0, 0, 0);
ret = 0;
}
ast_debug(3, "SIP Fixup: New owner for dialogue %s: %s (Old parent: %s)\n", p->callid, p->owner->name, oldchan->name);
@@ -6014,7 +6073,7 @@
}
ast_string_field_set(p, context, default_context);
ast_string_field_set(p, parkinglot, default_parkinglot);
-
+ ast_string_field_set(p, engine, default_engine);
/* Add to active dialog list */
@@ -11661,6 +11720,7 @@
ast_string_field_set(p, mohinterpret, peer->mohinterpret);
ast_string_field_set(p, mohsuggest, peer->mohsuggest);
ast_string_field_set(p, parkinglot, peer->parkinglot);
+ ast_string_field_set(p, engine, peer->engine);
if (peer->callingpres) /* Peer calling pres setting will override RPID */
p->callingpres = peer->callingpres;
if (peer->maxms && peer->lastms)
@@ -13295,12 +13355,11 @@
#define FORMAT2 "%-15.15s %-11.11s %-8.8s %-10.10s %-10.10s (%-2.2s) %-6.6s %-10.10s %-10.10s ( %%) %-6.6s\n"
#define FORMAT "%-15.15s %-11.11s %-8.8s %-10.10u%-1.1s %-10.10u (%-2.2u%%) %-6.6u %-10.10u%-1.1s %-10.10u (%-2.2u%%) %-6.6u\n"
struct sip_pvt *cur = __cur;
-// unsigned int rxcount;
-// unsigned int txcount;
-// char durbuf[10];
- // int duration;
- // int durh, durm, durs;
- //struct ast_channel *c = cur->owner;
+ struct ast_rtp_instance_stats stats;
+ char durbuf[10];
+ int duration;
+ int durh, durm, durs;
+ struct ast_channel *c = cur->owner;
struct __show_chan_arg *arg = __arg;
int fd = arg->fd;
@@ -13314,9 +13373,7 @@
return 0; /* don't care, we scan all channels */
}
-/*
- rxcount = ast_rtp_get_qosvalue(cur->rtp, AST_RTP_RXCOUNT);
- txcount = ast_rtp_get_qosvalue(cur->rtp, AST_RTP_TXCOUNT);
+ ast_rtp_instance_get_stats(cur->rtp, &stats, AST_RTP_INSTANCE_STAT_ALL);
if (c && c->cdr && !ast_tvzero(c->cdr->start)) {
duration = (int)(ast_tvdiff_ms(ast_tvnow(), c->cdr->start) / 1000);
@@ -13332,19 +13389,19 @@
ast_inet_ntoa(cur->sa.sin_addr),
cur->callid,
durbuf,
- rxcount > (unsigned int) 100000 ? (unsigned int) (rxcount)/(unsigned int) 1000 : rxcount,
- rxcount > (unsigned int) 100000 ? "K":" ",
- ast_rtp_get_qosvalue(cur->rtp, AST_RTP_RXPLOSS),
- rxcount > ast_rtp_get_qosvalue(cur->rtp, AST_RTP_RXPLOSS) ? (unsigned int) (ast_rtp_get_qosvalue(cur->rtp, AST_RTP_RXPLOSS) / rxcount * 100) : 0,
- ast_rtp_get_qosvalue(cur->rtp, AST_RTP_RXJITTER),
- txcount > (unsigned int) 100000 ? (unsigned int) (txcount)/(unsigned int) 1000 : txcount,
- txcount > (unsigned int) 100000 ? "K":" ",
- ast_rtp_get_qosvalue(cur->rtp, AST_RTP_TXPLOSS),
- txcount > ast_rtp_get_qosvalue(cur->rtp, AST_RTP_TXPLOSS) ? (unsigned int) (ast_rtp_get_qosvalue(cur->rtp, AST_RTP_TXPLOSS)/ txcount * 100) : 0,
- ast_rtp_get_qosvalue(cur->rtp, AST_RTP_TXJITTER)
+ stats.rxcount > (unsigned int) 100000 ? (unsigned int) (stats.rxcount)/(unsigned int) 1000 : stats.rxcount,
+ stats.rxcount > (unsigned int) 100000 ? "K":" ",
+ stats.rxploss,
+ stats.rxcount > stats.rxploss ? (stats.rxploss / stats.rxcount * 100) : 0,
+ stats.rxjitter,
+ stats.txcount > (unsigned int) 100000 ? (unsigned int) (stats.txcount)/(unsigned int) 1000 : stats.txcount,
+ stats.txcount > (unsigned int) 100000 ? "K":" ",
+ stats.txploss,
+ stats.txcount > stats.txploss ? (stats.txploss / stats.txcount * 100) : 0,
+ stats.txjitter
);
arg->numchans++;
-*/
+
return 0; /* don't care, we scan all channels */
}
@@ -17824,7 +17881,7 @@
static int acf_channel_read(struct ast_channel *chan, const char *funcname, char *preparse, char *buf, size_t buflen)
{
struct sip_pvt *p = chan->tech_pvt;
- char /**all = "",*/ *parse = ast_strdupa(preparse);
+ char *parse = ast_strdupa(preparse);
int res = 0;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(param);
@@ -17876,54 +17933,61 @@
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;
- struct ast_rtp *rtp = p->rtp;
-
- memset(&qos, 0, sizeof(qos));
-
- if (ast_strlen_zero(args.type))
+ struct ast_rtp_instance *rtp = NULL;
+
+ 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(rtp = p->rtp, &qos, RTPQOS_SUMMARY);
- } else if (strcasecmp(args.type, "VIDEO") == 0) {
- all = ast_rtp_get_quality(rtp = p->vrtp, &qos, RTPQOS_SUMMARY);
- } else if (strcasecmp(args.type, "TEXT") == 0) {
- all = ast_rtp_get_quality(rtp = p->trtp, &qos, RTPQOS_SUMMARY);
+ }
+
+ if (!strcasecmp(args.type, "audio")) {
+ rtp = p->rtp;
+ } else if (!strcasecmp(args.type, "video")) {
+ rtp = p->vrtp;
+ } else if (!strcasecmp(args.type, "text")) {
+ rtp = p->trtp;
} else {
return -1;
}
-
- 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, "%.0f", 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, "%.0f", 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, "%.0f", qos.rtt * 1000.0);
- else if (strcasecmp(args.field, "all") == 0)
- ast_copy_string(buf, all, buflen);
- else if (!ast_rtp_get_qos(rtp, args.field, buf, buflen))
- ;
- else {
- ast_log(LOG_WARNING, "Unrecognized argument '%s' to %s\n", preparse, funcname);
- return -1;
- }
-*/
+
+ if (ast_strlen_zero(args.field) || !strcasecmp(args.field, "all")) {
+ char quality_buf[AST_MAX_USER_FIELD], *quality;
+
+ if (!(quality = ast_rtp_instance_get_quality(rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
+ return -1;
+ }
+
+ ast_copy_string(buf, quality_buf, buflen);
+ return res;
+ } else {
+ struct ast_rtp_instance_stats stats;
+
+ if (ast_rtp_instance_get_stats(rtp, &stats, AST_RTP_INSTANCE_STAT_ALL)) {
+ return -1;
+ }
+
+ if (!strcasecmp(args.field, "local_ssrc")) {
+ snprintf(buf, buflen, "%u", stats.local_ssrc);
+ } else if (!strcasecmp(args.field, "local_lostpackets")) {
+ snprintf(buf, buflen, "%u", stats.rxploss);
+ } else if (!strcasecmp(args.field, "local_jitter")) {
+ snprintf(buf, buflen, "%u", stats.rxjitter);
+ } else if (!strcasecmp(args.field, "local_count")) {
+ snprintf(buf, buflen, "%u", stats.rxcount);
+ } else if (!strcasecmp(args.field, "remote_ssrc")) {
+ snprintf(buf, buflen, "%u", stats.remote_ssrc);
+ } else if (!strcasecmp(args.field, "remote_lostpackets")) {
+ snprintf(buf, buflen, "%u", stats.txploss);
+ } else if (!strcasecmp(args.field, "remote_jitter")) {
+ snprintf(buf, buflen, "%u", stats.txjitter);
+ } else if (!strcasecmp(args.field, "remote_count")) {
+ snprintf(buf, buflen, "%u", stats.txcount);
+ } else if (!strcasecmp(args.field, "rtt")) {
+ snprintf(buf, buflen, "%u", stats.rtt);
+ } else {
+ ast_log(LOG_WARNING, "Unrecognized argument '%s' to %s\n", preparse, funcname);
+ return -1;
+ }
+ }
} else {
res = -1;
}
@@ -17951,56 +18015,59 @@
sip_alreadygone(p);
/* Get RTCP quality before end of call */
-/*
if (p->do_history || p->owner) {
- struct ast_channel *bridge = ast_bridged_channel(p->owner);
- char *videoqos, *textqos;
-
- if (p->rtp) {
+ char quality_buf[AST_MAX_USER_FIELD], *quality;
+
+ if (p->rtp && (quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
+ struct ast_channel *bridge = ast_bridged_channel(p->owner);
+
if (p->do_history) {
- char *audioqos,
- *audioqos_jitter,
- *audioqos_loss,
- *audioqos_rtt;
-
- audioqos = ast_rtp_get_quality(p->rtp, NULL, RTPQOS_SUMMARY);
- audioqos_jitter = ast_rtp_get_quality(p->rtp, NULL, RTPQOS_JITTER);
- audioqos_loss = ast_rtp_get_quality(p->rtp, NULL, RTPQOS_LOSS);
- audioqos_rtt = ast_rtp_get_quality(p->rtp, NULL, RTPQOS_RTT);
-
- append_history(p, "RTCPaudio", "Quality:%s", audioqos);
- append_history(p, "RTCPaudioJitter", "Quality:%s", audioqos_jitter);
- append_history(p, "RTCPaudioLoss", "Quality:%s", audioqos_loss);
- append_history(p, "RTCPaudioRTT", "Quality:%s", audioqos_rtt);
+ append_history(p, "RTCPaudio", "Quality:%s", quality);
+ }
+
+ pbx_builtin_setvar_helper(p->owner, "RTPAUDIOQOS", quality);
+
+ if (bridge) {
+ pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSBRIDGED", quality);
}
- ast_rtp_set_vars(p->owner, p->rtp);
- }
-
- if (bridge) {
- struct sip_pvt *q = bridge->tech_pvt;
-
- if (IS_SIP_TECH(bridge->tech) && q->rtp)
- ast_rtp_set_vars(bridge, q->rtp);
- }
-
- if (p->vrtp) {
- videoqos = ast_rtp_get_quality(p->vrtp, NULL, RTPQOS_SUMMARY);
- if (p->do_history)
- append_history(p, "RTCPvideo", "Quality:%s", videoqos);
- if (p->owner)
- pbx_builtin_setvar_helper(p->owner, "RTPVIDEOQOS", videoqos);
- }
-
- if (p->trtp) {
- textqos = ast_rtp_get_quality(p->trtp, NULL, RTPQOS_SUMMARY);
- if (p->do_history)
- append_history(p, "RTCPtext", "Quality:%s", textqos);
- if (p->owner)
- pbx_builtin_setvar_helper(p->owner, "RTPTEXTQOS", textqos);
- }
- }
-*/
+ if ((quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER, quality_buf, sizeof(quality_buf)))) {
+ pbx_builtin_setvar_helper(p->owner, "RTPAUDIOQOSJITTER", quality);
+ if (bridge) {
+ pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSJITTERBRIDGED", quality);
+ }
+ }
+ if ((quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS, quality_buf, sizeof(quality_buf)))) {
+ pbx_builtin_setvar_helper(p->owner, "RTPAUDIOQOSLOSS", quality);
+ if (bridge) {
+ pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSLOSSBRIDGED", quality);
+ }
+ }
+ if ((quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT, quality_buf, sizeof(quality_buf)))) {
+ pbx_builtin_setvar_helper(p->owner, "RTPAUDIOQOSRTT", quality);
+ if (bridge) {
+ pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSRTTBRIDGED", quality);
+ }
+ }
+ }
+
+ if (p->vrtp && (quality = ast_rtp_instance_get_quality(p->vrtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
+ if (p->do_history) {
+ append_history(p, "RTCPvideo", "Quality:%s", quality);
+ }
+ if (p->owner) {
+ pbx_builtin_setvar_helper(p->owner, "RTPVIDEOQOS", quality);
+ }
+ }
+ if (p->trtp && (quality = ast_rtp_instance_get_quality(p->trtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
+ if (p->do_history) {
+ append_history(p, "RTCPtext", "Quality:%s", quality);
+ }
+ if (p->owner) {
+ pbx_builtin_setvar_helper(p->owner, "RTPTEXTQOS", quality);
+ }
+ }
+ }
stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
stop_session_timer(p); /* Stop Session-Timer */
@@ -20477,6 +20544,8 @@
ast_copy_string(peer->mohsuggest, v->value, sizeof(peer->mohsuggest));
} else if (!strcasecmp(v->name, "parkinglot")) {
ast_copy_string(peer->parkinglot, v->value, sizeof(peer->parkinglot));
+ } else if (!strcasecmp(v->name, "rtp_engine")) {
+ ast_copy_string(peer->engine, v->value, sizeof(peer->engine));
} else if (!strcasecmp(v->name, "mailbox")) {
add_peer_mailboxes(peer, v->value);
} else if (!strcasecmp(v->name, "hasvoicemail")) {
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