[svn-commits] oej: branch oej/pinequality-manager-qos-reports-1.4 r226833 - in /team/oej/pi...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Nov 2 11:22:27 CST 2009
Author: oej
Date: Mon Nov 2 11:22:23 2009
New Revision: 226833
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=226833
Log:
Playing around with RTCP reports
Modified:
team/oej/pinequality-manager-qos-reports-1.4/channels/chan_sip.c
team/oej/pinequality-manager-qos-reports-1.4/include/asterisk/rtp.h
Modified: team/oej/pinequality-manager-qos-reports-1.4/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinequality-manager-qos-reports-1.4/channels/chan_sip.c?view=diff&rev=226833&r1=226832&r2=226833
==============================================================================
--- team/oej/pinequality-manager-qos-reports-1.4/channels/chan_sip.c (original)
+++ team/oej/pinequality-manager-qos-reports-1.4/channels/chan_sip.c Mon Nov 2 11:22:23 2009
@@ -1458,6 +1458,7 @@
static int sip_do_reload(enum channelreloadreason reason);
static int sip_reload(int fd, int argc, char *argv[]);
static int acf_channel_read(struct ast_channel *chan, char *funcname, char *preparse, char *buf, size_t buflen);
+static int get_rtp_quality(struct sip_pvt *dialog, const char *chantype, const char *qostype, char *buf, size_t buf_len);
/*--- Debugging
Functions for enabling debug per IP or fully, or enabling history logging for
@@ -10945,6 +10946,107 @@
}
}
#undef FORMAT
+
+/*! \brief Add manager headers for QoS to existing manager reply */
+static int manager_add_qos(struct mansession *s, char *mediatype, struct sip_pvt *dialog)
+{
+ char mybuf[SIPBUFSIZE];
+
+ get_rtp_qos(dialog, mediatype, "local_ssrc", buf, SIPBUFSIZE );
+ astman_append(s, "LocalSSRC(%s): %s\r\n", mediatype, buf);
+ get_rtp_qos(dialog, mediatype, "remote_ssrc", buf, SIPBUFSIZE );
+ astman_append(s, "RemoteSSRC(%s): %s\r\n", mediatype, buf);
+ get_rtp_qos(dialog, mediatype, "local_jitter", buf, SIPBUFSIZE );
+ astman_append(s, "LocalJitter(%s): %s\r\n", mediatype, buf);
+ get_rtp_qos(dialog, mediatype, "local_count", buf, SIPBUFSIZE );
+ astman_append(s, "LocalPacketCount(%s): %s\r\n", mediatype, buf);
+ get_rtp_qos(dialog, mediatype, "remote_count", buf, SIPBUFSIZE );
+ astman_append(s, "RemotePacketCount(%s): %s\r\n", mediatype, buf);
+ get_rtp_qos(dialog, mediatype, "local_lostpackets", buf, SIPBUFSIZE );
+ astman_append(s, "LocalLostPackets(%s): %s\r\n", mediatype, buf);
+ get_rtp_qos(dialog, mediatype, "remote_lostpackets", buf, SIPBUFSIZE );
+ astman_append(s, "RemoteLostPackets(%s): %s\r\n", mediatype, buf);
+ get_rtp_qos(dialog, mediatype, "remote_jitter", buf, SIPBUFSIZE );
+ astman_append(s, "RemoteJitter(%s): %s\r\n", mediatype, buf);
+ get_rtp_qos(dialog, mediatype, "rtt", buf, SIPBUFSIZE );
+ astman_append(s, "MediaRtt(%s): %s\r\n", mediatype, buf);
+}
+
+static char mandescr_sip_channel[] =
+"Description: Show one SIP channel with details on current status.\n"
+"Variables: \n"
+" Channel: <name> The channel name (AST) you want to check.\n"
+" ActionID: <id> Optional action ID for this AMI transaction."
+" Datatype: <name> Optional parameter name required. If not specified, all data will be sent"
+""
+" Datatype Description"
+" qos Current QoS value for this channel (sender and receiver)"
+"\n";
+
+/*! \brief Show SIP channel data in the manager API */
+static int manager_sip_channel(struct mansession *s, const struct message *m)
+{
+ const char *channel;
+ const char *datatype;
+ const char *actionid;
+ int ret;
+ int all = FALSE;
+ struct ast_channel *chan = NULL;
+ struct sip_pvt *dialog;
+
+ actionid = astman_get_header(m,"ActionID");
+ channel = astman_get_header(m,"Channel");
+
+ /* Do we have a channel name? */
+ if (ast_strlen_zero(channel)) {
+ astman_send_error(s, m, "Channel: <channelname> missing.");
+ return 0;
+ }
+ /* Can we find a channel with that name? */
+ chan = ast_get_channel_by_name_locked(name);
+ if (!chan) {
+ astman_send_error(s, m, "No such channel");
+ return 0;
+ }
+ /* Sanity check - is this pvt a SIP channel? */
+ if (chan->tech != &sip_tech && chan->tech != &sip_tech_info) {
+ astman_send_error(s, m, "Cannot get SIPchannel-info on a non-SIP channel\n");
+ return 0;
+ }
+ /* Try to grab the channel pvt handle */
+ dialog = chan->tech_pvt;
+ if (!dialog) {
+ astman_send_error(s, m, "No active SIP channel\n");
+ return 0;
+ }
+ /* Ok, we have a channel with a SIP dialog attached,
+ time to find out what they want
+ ...and when my plane is about to leave from ARN */
+ datatype = astman_get_header(m,"Datatype");
+ if (ast_strlen_zero(channel) || !strcasecmp(datatype, "all")) {
+ all = TRUE;
+ }
+ astman_append(s, "Response: Success\r\n");
+ if (!ast_strlen_zero(id)) {
+ astman_append(s, "ActionID: %s\r\n",id);
+ }
+
+
+ if (all || !strcasecmp(datatype, "qos")) {
+ if (p->rtp) {
+ manager_add_qos(s, "audio", dialog);
+ }
+ if (p->vrtp) {
+ manager_add_qos(s, "video", dialog);
+ }
+ } else if (!all) {
+ astman_send_error(s, m, "Unknown datatype: %s\n", datatype);
+ return 0;
+ }
+ astman_append(s, "\r\n\r\n" );
+ ast_channel_unlock(c);
+ return(0);
+}
static char mandescr_show_peer[] =
"Description: Show one SIP peer with details on current status.\n"
@@ -15723,9 +15825,8 @@
static int acf_channel_read(struct ast_channel *chan, char *funcname, char *preparse, char *buf, size_t buflen)
{
- struct ast_rtp_quality qos;
struct sip_pvt *p = chan->tech_pvt;
- char *all = "", *parse = ast_strdupa(preparse);
+ char *parse = ast_strdupa(preparse);
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(param);
AST_APP_ARG(type);
@@ -15738,6 +15839,10 @@
ast_log(LOG_ERROR, "Cannot call %s on a non-SIP channel\n", funcname);
return 0;
}
+ if (p == NULL) {
+ ast_log(LOG_ERROR, "Cannot call %s on a non-SIP channel\n", funcname);
+ return -1;
+ }
if (strcasecmp(args.param, "rtpqos"))
return 0;
@@ -15749,40 +15854,50 @@
args.field = "all";
memset(buf, 0, buflen);
+ if (get_rtp_quality(p, args.type, args.field, buf, buflen)) {
+ ast_log(LOG_WARNING, "Unrecognized argument '%s' to %s\n", preparse, funcname);
+ return -1;
+ }
+ return 0;
+}
+
+/*! \brief Get RTP quality string */
+static int get_rtp_quality(struct sip_pvt *dialog, const char *chantype, const char *qostype, char *buf, size_t buflen)
+{
+ struct ast_rtp_quality qos;
+ char *all = "";
memset(&qos, 0, sizeof(qos));
- if (p == NULL) {
- return -1;
- }
-
- if (strcasecmp(args.type, "AUDIO") == 0) {
- all = ast_rtp_get_quality(p->rtp, &qos);
- } else if (strcasecmp(args.type, "VIDEO") == 0) {
- all = ast_rtp_get_quality(p->vrtp, &qos);
- }
-
- if (strcasecmp(args.field, "local_ssrc") == 0)
+ if (strcasecmp(chantype, "AUDIO") == 0) {
+ all = ast_rtp_get_quality(dialog->rtp, &qos);
+ } else if (strcasecmp(chantype, "VIDEO") == 0) {
+ all = ast_rtp_get_quality(dialog->vrtp, &qos);
+ } else {
+ return -1; /* Unknown media */
+ }
+
+ if (strcasecmp(qostype, "local_ssrc") == 0) {
snprintf(buf, buflen, "%u", qos.local_ssrc);
- else if (strcasecmp(args.field, "local_lostpackets") == 0)
+ } else if (strcasecmp(qostype, "local_lostpackets") == 0) {
snprintf(buf, buflen, "%u", qos.local_lostpackets);
- else if (strcasecmp(args.field, "local_jitter") == 0)
+ } else if (strcasecmp(qostype, "local_jitter") == 0) {
snprintf(buf, buflen, "%.0lf", qos.local_jitter * 1000.0);
- else if (strcasecmp(args.field, "local_count") == 0)
+ } else if (strcasecmp(qostype, "local_count") == 0) {
snprintf(buf, buflen, "%u", qos.local_count);
- else if (strcasecmp(args.field, "remote_ssrc") == 0)
+ } else if (strcasecmp(qostype, "remote_ssrc") == 0) {
snprintf(buf, buflen, "%u", qos.remote_ssrc);
- else if (strcasecmp(args.field, "remote_lostpackets") == 0)
+ } else if (strcasecmp(qostype, "remote_lostpackets") == 0) {
snprintf(buf, buflen, "%u", qos.remote_lostpackets);
- else if (strcasecmp(args.field, "remote_jitter") == 0)
+ } else if (strcasecmp(qostype, "remote_jitter") == 0) {
snprintf(buf, buflen, "%.0lf", qos.remote_jitter * 1000.0);
- else if (strcasecmp(args.field, "remote_count") == 0)
+ } else if (strcasecmp(qostype, "remote_count") == 0) {
snprintf(buf, buflen, "%u", qos.remote_count);
- else if (strcasecmp(args.field, "rtt") == 0)
+ } else if (strcasecmp(qostype, "rtt") == 0) {
snprintf(buf, buflen, "%.0lf", qos.rtt * 1000.0);
- else if (strcasecmp(args.field, "all") == 0)
+ } else if (strcasecmp(qostype, "all") == 0) {
ast_copy_string(buf, all, buflen);
- else {
- ast_log(LOG_WARNING, "Unrecognized argument '%s' to %s\n", preparse, funcname);
+ } else {
+ /* Unrecognized argument, error! Let the caller print error message */
return -1;
}
return 0;
@@ -19376,6 +19491,8 @@
"List SIP peers (text format)", mandescr_show_peers);
ast_manager_register2("SIPshowpeer", EVENT_FLAG_SYSTEM, manager_sip_show_peer,
"Show SIP peer (text format)", mandescr_show_peer);
+ ast_manager_register2("SIPchannel", EVENT_FLAG_SYSTEM, manager_sip_channel,
+ "Show information about SIP channel", mandescr_sip_channel);
sip_poke_all_peers();
sip_send_all_registers();
@@ -19414,6 +19531,7 @@
ast_udptl_proto_unregister(&sip_udptl);
/* Unregister AMI actions */
+ ast_manager_unregister("SIPchannel");
ast_manager_unregister("SIPpeers");
ast_manager_unregister("SIPshowpeer");
Modified: team/oej/pinequality-manager-qos-reports-1.4/include/asterisk/rtp.h
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinequality-manager-qos-reports-1.4/include/asterisk/rtp.h?view=diff&rev=226833&r1=226832&r2=226833
==============================================================================
--- team/oej/pinequality-manager-qos-reports-1.4/include/asterisk/rtp.h (original)
+++ team/oej/pinequality-manager-qos-reports-1.4/include/asterisk/rtp.h Mon Nov 2 11:22:23 2009
@@ -64,6 +64,7 @@
};
struct ast_rtp;
+struct ast_rtp_quality;
struct ast_rtp_protocol {
/*! Get RTP struct, or NULL if unwilling to transfer */
@@ -73,6 +74,8 @@
/*! Set RTP peer */
int (* const set_rtp_peer)(struct ast_channel *chan, struct ast_rtp *peer, struct ast_rtp *vpeer, int codecs, int nat_active);
int (* const get_codec)(struct ast_channel *chan);
+ /*! RTCP callback */
+ int (* const rtcpcallback)(struct ast_rtp_quality *qualityreport);
const char * const type;
AST_LIST_ENTRY(ast_rtp_protocol) list;
};
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