[svn-commits] oej: branch oej/pinefrog-1.4 r252225 - in /team/oej/pinefrog-1.4: ./ channels...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Sat Mar 13 02:01:24 CST 2010
Author: oej
Date: Sat Mar 13 02:01:18 2010
New Revision: 252225
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=252225
Log:
- Moving storage of RTCP data in realtime out of the call thread
- Adding information about transcoding to QoS data structure (transcoding adds delay,
and Asterisk has an estimate of that delay)
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=252225&r1=252224&r2=252225
==============================================================================
--- team/oej/pinefrog-1.4/channels/chan_sip.c (original)
+++ team/oej/pinefrog-1.4/channels/chan_sip.c Sat Mar 13 02:01:18 2010
@@ -1052,6 +1052,7 @@
struct sip_invite_param *options; /*!< Options for INVITE */
int autoframing;
int hangupcause; /*!< Storage of hangupcause copied from our owner before we disconnect from the AST channel (only used at hangup) */
+ struct ast_rtp_quality *qual; /*!< The latest quality report, for realtime storage */
/*! When receiving an SDP offer, it is important to take note of what media types were offered.
* By doing this, even if we don't want to answer a particular media stream with something meaningful, we can
* still put an m= line in our answer with the port set to 0.
@@ -2603,9 +2604,18 @@
if (option_debug > 2)
ast_log(LOG_DEBUG, "Destroying SIP peer %s\n", peer->name);
+ if (peer->qual) {
+ /* We have a quality report to write to realtime before we leave this world. */
+ qos_write_realtime(peer->qual);
+ free(peer->qual);
+ peer->qual = NULL;
+ }
+
/* Delete it, it needs to disappear */
- if (peer->call)
+ if (peer->call) {
sip_destroy(peer->call);
+ peer->call = NULL;
+ }
if (peer->mwipvt) /* We have an active subscription, delete it */
sip_destroy(peer->mwipvt);
@@ -13435,25 +13445,47 @@
}
/*! \brief send manager report of RTCP
- endreport means endof-call report
+ 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)
*/
-static void sip_rtcp_report(struct sip_pvt *p, struct ast_rtp *rtp, const char *mediatype, int endreport)
+static void sip_rtcp_report(struct sip_pvt *p, struct ast_rtp *rtp, const char *mediatype, int reporttype)
{
struct ast_rtp_quality qual;
char *rtpqstring = NULL;
char localjitter[10], remotejitter[10];
int qosrealtime = ast_check_realtime("rtpqos");
unsigned int duration; /* Duration in secs */
+ int readtrans = FALSE, writetrans = FALSE;
memset(&qual, sizeof(qual), 0);
-
+
if (p && p->owner) {
struct ast_channel *bridgepeer = ast_bridged_channel(p->owner);
if (bridgepeer) {
/* Store the bridged peer data while we have it */
ast_rtcp_set_bridged(rtp, bridgepeer->name, bridgepeer->uniqueid);
- ast_log(log_debug, "---- Setting bridged peer name to %s\n", bridgepeer->name);
- }
+ 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) {
+ if (readtrans && p->owner->readtrans->translator) {
+ ast_log(LOG_DEBUG, "--- Read translator: %s Cost %d\n", p->owner->readtrans->translator->name, p->owner->readtrans->translator->cost);
+ }
+ if (writetrans && p->owner->writetrans->translator) {
+ ast_log(LOG_DEBUG, "--- Write translator: %s \n", p->owner->writetrans->translator->name, p->owner->writetrans->translator->cost);
+ }
+ }
+ }
+
}
if (global_rtcpevents) {
@@ -13487,12 +13519,14 @@
"RTPInLocalPlPercent: %5.2f\r\n"
"RTPOutPacketLoss: %d\r\n"
"RTPOutPlPercent: %5.2f\r\n"
+ "ChanTranslatRead: %s\r\n"
+ "ChanTranslatWrite: %s\r\n"
"\r\n",
p->owner ? p->owner->name : "",
p->owner ? p->owner->uniqueid : "",
qual.bridgedchan[0] ? qual.bridgedchan : "" ,
qual.bridgeduniqueid[0] ? qual.bridgeduniqueid : "",
- endreport ? "Final" : "Update",
+ reporttype == 1 ? "Final" : "Update",
qual.numberofreports == 0 ? "Inactive" : "Active",
duration,
p->callid,
@@ -13514,40 +13548,61 @@
/* The remote counter of lost packets (if we got the reports)
divided with our counter of sent packets
*/
- (qual.local_count + qual.remote_lostpackets) > 0 ? (double) qual.remote_lostpackets / qual.local_count * 100 : 0
+ (qual.local_count + qual.remote_lostpackets) > 0 ? (double) qual.remote_lostpackets / qual.local_count * 100 : 0,
+ readtrans ? "yes" : "no",
+ writetrans ? "yes" : "no"
);
}
-#ifdef REALTIME2
+
/* CDR records are not reliable when it comes to near-death-of-channel events, so we need to store the RTCP
- report in realtime when we have it */
- if (endreport && qosrealtime) {
- char buf_duration[10], buf_lssrc[30], buf_rssrc[30], buf_rtt[30];
- duration = (unsigned int)(ast_tvdiff_ms(ast_tvnow(), qual.start) / 1000);
-
- if (rtpqstring == NULL) {
- rtpqstring = ast_rtp_get_quality(rtp, &qual);
- }
- sprintf(localjitter, "%f", qual.local_jitter);
- sprintf(remotejitter, "%f", qual.remote_jitter);
- sprintf(buf_lssrc, "%u", qual.local_ssrc);
- sprintf(buf_rssrc, "%u", qual.remote_ssrc);
- sprintf(buf_rtt, "%f", qual.rtt);
- sprintf(buf_duration, "%ld", duration);
- ast_store_realtime("rtpqos",
- "channel", p->owner ? p->owner->name : "",
- "uniqueid", p->owner ? p->owner->uniqueid : "",
- "bridgedchan", qual.bridgedchan[0] ? qual.bridgedchan : "" ,
- "bridgeduniqueid", qual.bridgeduniqueid[0] ? qual.bridgeduniqueid : "",
- "pvtcallid", p->callid,
- "rtpmedia", mediatype,
- "localssrc", buf_lssrc, "remotessrc", buf_rssrc,
- "rtt", buf_rtt,
- "localjitter", localjitter, "remotejitter", remotejitter,
- "sendformat", ast_getformatname(qual.lasttxformat),
- "receiveformat", ast_getformatname(qual.lastrxformat),
- "rtcpstatus", qual.numberofreports == 0 ? "Inactive" : "Active",
- "duration", buf_duration,
- NULL);
+ report in realtime when we have it.
+ Tests have proven that storing to realtime from the call thread is NOT a good thing. Therefore, we just save
+ 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) {
+ p->qual = ast_calloc(sizeof(struct ast_rtp_quality), 1)
+ p->qual = qual;
+ p->qual->end = ast_tvnow();
+ }
+}
+
+/*! \brief Write quality report to realtime storage */
+void qos_write_realtime(struct ast_rtp_quality *qual)
+{
+ unsigned int duration; /* Duration in secs */
+ char buf_duration[10], buf_lssrc[30], buf_rssrc[30], buf_rtt[30];
+ char localjitter[10], remotejitter[10];
+
+ /* Since the CDR is already gone, we need to calculate our own duration.
+ The CDR duration is the definitive resource for billing, this is
+ the RTP stream duration which may include early media (ringing and
+ provider messages). Only useful for measurements.
+ */
+ duration = (unsigned int)(ast_tvdiff_ms(qual->end, qual->start) / 1000);
+
+ /* Realtime is based on strings, so let's make strings */
+ sprintf(localjitter, "%f", qual->local_jitter);
+ sprintf(remotejitter, "%f", qual->remote_jitter);
+ sprintf(buf_lssrc, "%u", qual->local_ssrc);
+ sprintf(buf_rssrc, "%u", qual->remote_ssrc);
+ sprintf(buf_rtt, "%f", qual->rtt);
+ sprintf(buf_duration, "%ld", duration);
+ ast_store_realtime("rtpqos",
+ "channel", p->owner ? p->owner->name : "",
+ "uniqueid", p->owner ? p->owner->uniqueid : "",
+ "bridgedchan", qual->bridgedchan[0] ? qual->bridgedchan : "" ,
+ "bridgeduniqueid", qual->bridgeduniqueid[0] ? qual->bridgeduniqueid : "",
+ "pvtcallid", p->callid,
+ "rtpmedia", mediatype,
+ "localssrc", buf_lssrc, "remotessrc", buf_rssrc,
+ "rtt", buf_rtt,
+ "localjitter", localjitter, "remotejitter", remotejitter,
+ "sendformat", ast_getformatname(qual->lasttxformat),
+ "receiveformat", ast_getformatname(qual->lastrxformat),
+ "rtcpstatus", qual->numberofreports == 0 ? "Inactive" : "Active",
+ "duration", buf_duration,
+ NULL);
}
}
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=252225&r1=252224&r2=252225
==============================================================================
--- team/oej/pinefrog-1.4/include/asterisk/rtp.h (original)
+++ team/oej/pinefrog-1.4/include/asterisk/rtp.h Sat Mar 13 02:01:18 2010
@@ -101,6 +101,9 @@
int lastrxformat; /*!< Last used codec on received stream */
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];
};
@@ -203,6 +206,16 @@
a new name
*/
void ast_rtcp_set_bridged(struct ast_rtp *rtp, const char *bridged_name, const char *bridged_uniqueid);
+
+/*! \brief Store translator information
+
+In order to measure quality of a phone call, information about transcoding is very useful. Translation
+adds processing and latency to the bridged call.
+*/
+void ast_rtcp_set_translation(struct ast_rtp *rtp, const char *writetranslator, const int writecost,
+ const char *readtranslator, cost int readcost);
+
+
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=252225&r1=252224&r2=252225
==============================================================================
--- team/oej/pinefrog-1.4/main/rtp.c (original)
+++ team/oej/pinefrog-1.4/main/rtp.c Sat Mar 13 02:01:18 2010
@@ -197,7 +197,7 @@
struct ast_rtp *bridged; /*!< Who we are Packet bridged to */
int set_marker_bit:1; /*!< Whether to set the marker bit or not */
unsigned int constantssrc:1;
- int isactive:1; /*!< Whether to RTP stream is active or not */
+ int isactive:2; /*!< Whether the RTP stream is active or not */
};
/* Forward declarations */
@@ -278,7 +278,11 @@
unsigned int rtt_count; /*! Number of reports received */
int sendfur;
char bridgedchan[AST_MAX_EXTENSION]; /*!< Bridged channel name */
- char bridgeduniqueid[AST_MAX_EXTENSION]; /*!< Bridged channel name */
+ char bridgeduniqueid[AST_MAX_EXTENSION]; /*!< Bridged channel uniqueid */
+ char readtranslator[80]; /* Translation done on reading audio from PBX */
+ 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 */
};
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