[svn-commits] oej: branch oej/pinefrog-1.4 r286265 - in /team/oej/pinefrog-1.4: ./ channels...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Sat Sep 11 09:45:56 CDT 2010
Author: oej
Date: Sat Sep 11 09:45:48 2010
New Revision: 286265
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=286265
Log:
Development should move forward. Adding some support for translation time in the reports
Modified:
team/oej/pinefrog-1.4/ (props changed)
team/oej/pinefrog-1.4/channels/chan_sip.c
team/oej/pinefrog-1.4/include/asterisk/rtp.h
team/oej/pinefrog-1.4/main/rtp.c
Propchange: team/oej/pinefrog-1.4/
------------------------------------------------------------------------------
automerge = http://www.codename-pineapple.org/
Modified: team/oej/pinefrog-1.4/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinefrog-1.4/channels/chan_sip.c?view=diff&rev=286265&r1=286264&r2=286265
==============================================================================
--- team/oej/pinefrog-1.4/channels/chan_sip.c (original)
+++ team/oej/pinefrog-1.4/channels/chan_sip.c Sat Sep 11 09:45:48 2010
@@ -13570,11 +13570,13 @@
}
/*! \brief send manager report of RTCP
- reporttype = 1 means endof-call report
reporttype = 0 means report during call (if configured)
- reporttype = 2 means report at end of call leg (like transfer)
+ reporttype = 1 means endof-call (hangup) report
+ reporttype = 2 means report at closure of SIP pvt (used only for realtime reports. We might get
+ an incoming final RTCP after we send BYE */
+ reporttype = 10 means report at end of call leg (like transfer)
*/
-static void sip_rtcp_report(struct sip_pvt *p, struct ast_rtp *rtp, const char *mediatype, int reporttype)
+static void sip_rtcp_report(struct sip_pvt *p, struct ast_rtp *rtp, enum media_type type, int reporttype)
{
struct ast_rtp_quality qual;
char *rtpqstring = NULL;
@@ -13591,28 +13593,30 @@
ast_rtcp_set_bridged(rtp, bridgepeer->name, bridgepeer->uniqueid);
ast_log(LOG_DEBUG, "---- Setting bridged peer name to %s\n", bridgepeer->name);
}
- /* Try to find out if there's transcoding */
- readtrans = p->owner->readtrans != NULL;
- writetrans = p->owner->writetrans != NULL;
- if (option_debug > 1) {
- /* This is just exploring the way into translator system.
- if we have a translator, the bridge delay is increased, which affects the QoS
- of the call. If I can get info from the translation matrix too, that would
- be great!
- */
- if (option_debug) {
+
+ /* Try to find out if there's active transcoding */
+ /* Currently, the only media stream that has translation is the audio stream. At some point
+ we might have transcoding for other types of media. */
+ if (type == SDP_AUDIO) {
+ /* if we have a translator, the bridge delay is increased, which affects the QoS of the call. */
+ readtrans = p->owner->readtrans != NULL;
+ writetrans = p->owner->writetrans != NULL;
+ ast_rtcp_settranslator(rtp, readtrans ? p->owner->readtrans->t->name : NULL, readtrans ? p->owner->readtrans->t->cost : 0,
+ writetrans ? p->owner->writetrans->t->name : NULL, writetrans ? p->owner->writetrans->t->cost : 0);
+
+ if (option_debug > 1) {
if (readtrans && p->owner->readtrans->t) {
- ast_log(LOG_DEBUG, "--- Read translator: %s Cost %d\n", p->owner->readtrans->t->name, p->owner->readtrans->t->cost);
+ ast_log(LOG_DEBUG, "--- Audio Read translator: %s Cost %d\n", p->owner->readtrans->t->name, p->owner->readtrans->t->cost);
}
if (writetrans && p->owner->writetrans->t) {
- ast_log(LOG_DEBUG, "--- Write translator: %s Cost %d\n", p->owner->writetrans->t->name, p->owner->writetrans->t->cost);
+ ast_log(LOG_DEBUG, "--- Audio Write translator: %s Cost %d\n", p->owner->writetrans->t->name, p->owner->writetrans->t->cost);
}
}
}
}
- if (global_rtcpevents) {
+ if (global_rtcpevents && reporttype != 2) {
rtpqstring = ast_rtp_get_quality(rtp, &qual);
/*
If numberofreports == 0 we have no incoming RTCP active, thus we can't
@@ -13643,8 +13647,10 @@
"RTPInLocalPlPercent: %5.2f\r\n"
"RTPOutPacketLoss: %d\r\n"
"RTPOutPlPercent: %5.2f\r\n"
- "ChanTranslatRead: %s\r\n"
- "ChanTranslatWrite: %s\r\n"
+ "TranslateRead: %s\r\n"
+ "TranslateReadCost: %d\r\n"
+ "TranslateWrite: %s\r\n"
+ "TranslateWriteCost: %d\r\n"
"\r\n",
p->owner ? p->owner->name : "",
p->owner ? p->owner->uniqueid : "",
@@ -13655,7 +13661,7 @@
duration,
p->callid,
ast_inet_ntoa(qual.them.sin_addr),
- mediatype,
+ type == SDP_AUDIO ? "audio" : (type == SDP_VIDEO ? "video" : "fax") ,
ast_getformatname(qual.lasttxformat),
ast_getformatname(qual.lastrxformat),
qual.local_ssrc,
@@ -13673,9 +13679,9 @@
divided with our counter of sent packets
*/
(qual.local_count + qual.remote_lostpackets) > 0 ? (double) qual.remote_lostpackets / qual.local_count * 100 : 0,
- readtrans ? "yes" : "no",
- writetrans ? "yes" : "no"
- );
+ qual.readtranslator, qual.readcost,
+ qual.writetranslator, qual.writecost
+ );
}
/* CDR records are not reliable when it comes to near-death-of-channel events, so we need to store the RTCP
@@ -13684,12 +13690,12 @@
the quality report structure in the PVT and let the function that kills the pvt store the stuff in the
monitor thread instead.
*/
- if (reporttype == 1 && qosrealtime) {
- if (mediatype[0] == 'a') { /* Audio */
+ if (reporttype == 2 && qosrealtime) {
+ if (type == SDP_AUDIO) { /* Audio */
p->audioqual = ast_calloc(sizeof(struct ast_rtp_quality), 1);
(* p->audioqual) = qual;
p->audioqual->end = ast_tvnow();
- } else if (mediatype[0] == 'v') { /* Video */
+ } else if (type == SDP_VIDEO) { /* Video */
p->videoqual = ast_calloc(sizeof(struct ast_rtp_quality), 1);
(* p->videoqual) = qual;
p->videoqual->end = ast_tvnow();
@@ -13704,6 +13710,11 @@
unsigned int duration; /* Duration in secs */
char buf_duration[10], buf_lssrc[30], buf_rssrc[30], buf_rtt[30];
char localjitter[10], remotejitter[10];
+ char buf_readcost[5], buf_writecost[5];
+
+ /* Update the reports before writing */
+ sip_rtcp_report(p, p->rtp, SDP_AUDIO, 2);
+ sip_rtcp_report(p, p->vrtp, SDP_VIDEO, 2);
/* Since the CDR is already gone, we need to calculate our own duration.
The CDR duration is the definitive resource for billing, this is
@@ -13719,6 +13730,8 @@
sprintf(buf_rssrc, "%u", qual->remote_ssrc);
sprintf(buf_rtt, "%f", qual->rtt);
sprintf(buf_duration, "%u", duration);
+ sprintf(buf_readcost, "%d", qual->readcost);
+ sprintf(buf_writecost, "%d", qual->writecost);
ast_store_realtime("rtpqos",
"channel", dialog->owner ? dialog->owner->name : "",
"uniqueid", dialog->owner ? dialog->owner->uniqueid : "",
@@ -13733,6 +13746,10 @@
"receiveformat", ast_getformatname(qual->lastrxformat),
"rtcpstatus", qual->numberofreports == 0 ? "Inactive" : "Active",
"duration", buf_duration,
+ "writetranslator", qual->writetranslator,
+ "writecost", buf_writecost,
+ "readtranslator", qual->readtranslator,
+ "readcost", buf_readcost,
NULL);
}
#endif
@@ -13744,10 +13761,10 @@
struct sip_pvt *dialog = (struct sip_pvt *) data;
if (dialog->rtp && ast_rtp_isactive(dialog->rtp)) {
- sip_rtcp_report(dialog, dialog->rtp, "audio", FALSE);
+ sip_rtcp_report(dialog, dialog->rtp, SDP_AUDIO, FALSE);
}
if (dialog->vrtp && ast_rtp_isactive(dialog->vrtp)) {
- sip_rtcp_report(dialog, dialog->vrtp, "video", FALSE);
+ sip_rtcp_report(dialog, dialog->vrtp, SDP_VIDEO, FALSE);
}
return global_rtcptimer;
}
@@ -13775,11 +13792,11 @@
/* Immediately stop RTP, VRTP and UDPTL as applicable */
if (p->rtp && ast_rtp_isactive(p->rtp)) {
ast_rtp_stop(p->rtp);
- sip_rtcp_report(p, p->rtp, "audio", TRUE);
+ sip_rtcp_report(p, p->rtp, SDP_AUDIO, TRUE);
}
if (p->vrtp && ast_rtp_isactive(p->vrtp)) {
ast_rtp_stop(p->vrtp);
- sip_rtcp_report(p, p->vrtp, "video", TRUE);
+ sip_rtcp_report(p, p->vrtp, SDP_VIDEO, TRUE);
}
if (p->udptl)
ast_udptl_stop(p->udptl);
Modified: team/oej/pinefrog-1.4/include/asterisk/rtp.h
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinefrog-1.4/include/asterisk/rtp.h?view=diff&rev=286265&r1=286264&r2=286265
==============================================================================
--- team/oej/pinefrog-1.4/include/asterisk/rtp.h (original)
+++ team/oej/pinefrog-1.4/include/asterisk/rtp.h Sat Sep 11 09:45:48 2010
@@ -102,8 +102,10 @@
struct sockaddr_in them; /*!< The Ip address used for media by remote end */
struct timeval start; /*!< When the call started */
struct timeval end; /*!< When the call ended */
- char writetranslate[80];
- char readtranslate[80];
+ char writetranslator[80]; /*!< Translator used when writing */
+ char readtranslator[80]; /*!< Translator providing frames when reading */
+ int writecost; /*!< Cost in milliseconds for encoding/decoding 1 second of outbound media */
+ int readcost; /*!< Cost in milliseconds for encoding/decoding 1 second of inbound media */
};
@@ -197,6 +199,9 @@
int ast_rtp_settos(struct ast_rtp *rtp, int tos);
void ast_rtcp_setcname(struct ast_rtp *rtp, const char *cname, size_t length);
+
+/*! \brief Set the transcoding variables for the QoS reports */
+void ast_rtcp_settranslator(struct ast_rtp *rtp, const char *readtranslator, const int readcost, const char *writetranslator, const int writecost);
/*! \brief set the name of the bridged channel
Modified: team/oej/pinefrog-1.4/main/rtp.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinefrog-1.4/main/rtp.c?view=diff&rev=286265&r1=286264&r2=286265
==============================================================================
--- team/oej/pinefrog-1.4/main/rtp.c (original)
+++ team/oej/pinefrog-1.4/main/rtp.c Sat Sep 11 09:45:48 2010
@@ -281,8 +281,8 @@
char bridgedchan[AST_MAX_EXTENSION]; /*!< Bridged channel name */
char bridgeduniqueid[AST_MAX_EXTENSION]; /*!< Bridged channel uniqueid */
char readtranslator[80]; /* Translation done on reading audio from PBX */
+ char writetranslator[80]; /* Translation done on writing audio to PBX - bridged channel */
int readcost; /* Delay in milliseconds for translation of 1 second of audio */
- char writetranslator[80]; /* Translation done on writing audio to PBX - bridged channel */
int writecost; /* Delay in milliseconds for translation of 1 second of audio */
};
@@ -2403,6 +2403,19 @@
ast_log(LOG_DEBUG, "--- Copied CNAME %s to RTCP structure (length %d)\n", cname, (int) length);
}
+/*! \brief Set the transcoding variables for the QoS reports */
+void ast_rtcp_settranslator(struct ast_rtp *rtp, const char *readtranslator, const int readcost, const char *writetranslator, const int writecost)
+{
+ if (!rtp || !rtp->rtcp) {
+ return;
+ }
+ ast_copy_string(rtp->rtcp->readtranslator, readtranslator, sizeof(rtp->rtcp->readtranslator));
+ ast_copy_string(rtp->rtcp->writetranslator, writetranslator, sizeof(rtp->rtcp->writetranslator));
+ rtp->rtcp->readcost = readcost;
+ rtp->rtcp->writecost = writecost;
+
+}
+
/*! \brief set the name of the bridged channel
At the time when we write the report there might not be a bridge, so we need
@@ -2566,7 +2579,6 @@
*txcount transmitted packets
*rlp remote lost packets
*rtt round trip time
-
*/
if (qual && rtp) {
@@ -2596,6 +2608,14 @@
}
if (!ast_strlen_zero(rtp->rtcp->bridgeduniqueid)) {
ast_copy_string(qual->bridgeduniqueid, rtp->rtcp->bridgeduniqueid, sizeof(qual->bridgeduniqueid));
+ }
+ qual->readcost = rtp->rtcp->readcost;
+ qual->writecost = rtp->rtcp->writecost;
+ if (!ast_strlen_zero(rtp->rtcp->readtranslator)) {
+ ast_copy_string(qual->readtranslator, rtp->rtcp->readtranslator, sizeof(qual->readtranslator));
+ }
+ if (!ast_strlen_zero(rtp->rtcp->writetranslator)) {
+ ast_copy_string(qual->writetranslator, rtp->rtcp->writetranslator, sizeof(qual->writetranslator));
}
}
}
More information about the svn-commits
mailing list