[svn-commits] oej: branch oej/teapot-1.8 r402885 - in /team/oej/teapot-1.8: ./ channels/ ch...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Nov 19 02:28:53 CST 2013
Author: oej
Date: Tue Nov 19 02:28:49 2013
New Revision: 402885
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=402885
Log:
Adding Pinefrog RTCP patch to the Teapot
Added:
team/oej/teapot-1.8/patches/pinefrog-rtcp-1.8.diff (with props)
Modified:
team/oej/teapot-1.8/CREDITS
team/oej/teapot-1.8/channels/chan_sip.c
team/oej/teapot-1.8/channels/sip/dialplan_functions.c
team/oej/teapot-1.8/channels/sip/include/sip.h
team/oej/teapot-1.8/configs/extconfig.conf.sample
team/oej/teapot-1.8/include/asterisk/logger.h
team/oej/teapot-1.8/include/asterisk/rtp_engine.h
team/oej/teapot-1.8/main/logger.c
team/oej/teapot-1.8/main/rtp_engine.c
team/oej/teapot-1.8/res/res_rtp_asterisk.c
Modified: team/oej/teapot-1.8/CREDITS
URL: http://svnview.digium.com/svn/asterisk/team/oej/teapot-1.8/CREDITS?view=diff&rev=402885&r1=402884&r2=402885
==============================================================================
--- team/oej/teapot-1.8/CREDITS (original)
+++ team/oej/teapot-1.8/CREDITS Tue Nov 19 02:28:49 2013
@@ -22,6 +22,9 @@
Omnitor AB, Gunnar Hellström, for funding work with videocaps, T.140 RED,
originate with video/text and many more contributions.
+
+Nordicom Norge AS, Kristiansand, Norway, for funding work with RTCP support
+and Call Quality Records.
=== WISHLIST CONTRIBUTERS ===
Jeremy McNamara - SpeeX support
@@ -116,6 +119,7 @@
SIP presence support, SIP call state updates (dialog-info),
QUEUE_EXISTS function, device state provider architecture,
multiparking (together with mvanbaak), meetme and parking device states,
+ RTCP improvements, Call Quality Records,
MiniVM - the small voicemail system, many documentation
updates/corrections, and many bug fixes.
oej(AT)edvina.net, http://edvina.net
@@ -217,7 +221,8 @@
Viagenie, Canada - IPv6 support in socket layers and SIP implementation
Developers: Marc Blanchet, Simon Perreault and Jean-Philippe Dionne
-ClearIT AB, Sweden - res_mutestream, queue_exists and various other patches (developed by oej)
+ClearIT AB, Sweden - res_mutestream, queue_exists, RTCP improvements and various
+ other patches (developed by oej)
Despegar.com, Argentina - AstData API implementation, also sponsored by Google as part of the
gsoc/2009 program (developed by Eliel)
Modified: team/oej/teapot-1.8/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/teapot-1.8/channels/chan_sip.c?view=diff&rev=402885&r1=402884&r2=402885
==============================================================================
--- team/oej/teapot-1.8/channels/chan_sip.c (original)
+++ team/oej/teapot-1.8/channels/chan_sip.c Tue Nov 19 02:28:49 2013
@@ -276,6 +276,7 @@
#include "asterisk/xml.h"
#include "sip/include/dialog.h"
#include "sip/include/dialplan_functions.h"
+#include "sip/include/rtcp.h"
/*** DOCUMENTATION
@@ -673,7 +674,6 @@
{ AST_REDIRECTING_REASON_CALL_FWD_DTE, "unknown"}
};
-
/*! \name DefaultSettings
Default setttings are used as a channel setting and as a default when
configuring devices
@@ -6220,6 +6220,22 @@
if (dumphistory)
sip_dump_history(p);
+ AST_SCHED_DEL(sched, p->rtcpeventid);
+
+ if (p->audioqual) {
+ /* We have a quality report to write to realtime before we leave this world. */
+ qos_write_realtime(p, p->audioqual);
+ free(p->audioqual);
+ p->audioqual = NULL;
+ }
+ if (p->videoqual) {
+ /* We have a quality report to write to realtime before we leave this world. */
+ qos_write_realtime(p, p->videoqual);
+ free(p->videoqual);
+ p->videoqual = NULL;
+ }
+
+
if (p->options) {
if (p->options->outboundproxy) {
ao2_ref(p->options->outboundproxy, -1);
@@ -6785,6 +6801,7 @@
ast_rtp_instance_update_source(p->rtp);
res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL, oldsdp, TRUE);
ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
+ start_rtcp_events(p, sched);
}
sip_pvt_unlock(p);
return res;
@@ -8048,8 +8065,10 @@
set_socket_transport(&p->socket, SIP_TRANSPORT_UDP);
}
+ p->sip_cfg = &sip_cfg;
p->socket.fd = -1;
p->method = intended_method;
+ p->rtcpeventid = -1;
p->initid = -1;
p->waitid = -1;
p->reinviteid = -1;
@@ -8111,6 +8130,14 @@
build_callid_pvt(p);
else
ast_string_field_set(p, callid, callid);
+
+ /* Set cnames for the RTCP SDES */
+ if (p->rtp) {
+ ast_rtp_instance_setcname(p->rtp, p->callid, strlen(p->callid));
+ }
+ if (p->vrtp) {
+ ast_rtp_instance_setcname(p->vrtp, p->callid, strlen(p->callid));
+ }
/* Assign default music on hold class */
ast_string_field_set(p, mohinterpret, default_mohinterpret);
ast_string_field_set(p, mohsuggest, default_mohsuggest);
@@ -9034,6 +9061,7 @@
dialog->owner->uniqueid);
append_history(dialog, holdstate ? "Hold" : "Unhold", "%s", ast_str_buffer(req->data));
if (!holdstate) { /* Put off remote hold */
+ ast_rtp_instance_hold(dialog->rtp, 0); /* Turn off RTP hold */
ast_clear_flag(&dialog->flags[1], SIP_PAGE2_CALL_ONHOLD); /* Clear both flags */
return;
}
@@ -9802,9 +9830,9 @@
ast_queue_control_data(p->owner, AST_CONTROL_HOLD,
S_OR(p->mohsuggest, NULL),
!ast_strlen_zero(p->mohsuggest) ? strlen(p->mohsuggest) + 1 : 0);
- if (sendonly)
- ast_rtp_instance_stop(p->rtp);
- /* RTCP needs to go ahead, even if we're on hold!!! */
+ if (sendonly == 1 || sendonly == 2) { /* sendonly (from the other side) or inactive */
+ ast_rtp_instance_hold(p->rtp, 1);
+ }
/* Activate a re-invite */
ast_queue_frame(p->owner, &ast_null_frame);
change_hold_state(p, req, TRUE, sendonly);
@@ -18144,8 +18172,10 @@
int x = 0, load_realtime;
format_t codec = 0;
int realtimepeers;
+ int realtimertpqos = FALSE;
realtimepeers = ast_check_realtime("sippeers");
+ realtimertpqos = ast_check_realtime("rtpcqr");
if (argc < 4)
return CLI_SHOWUSAGE;
@@ -18764,9 +18794,11 @@
{
int realtimepeers;
int realtimeregs;
+ int realtimertpqos;
char codec_buf[SIPBUFSIZE];
const char *msg; /* temporary msg pointer */
struct sip_auth_container *credentials;
+
switch (cmd) {
case CLI_INIT:
@@ -18784,6 +18816,7 @@
realtimepeers = ast_check_realtime("sippeers");
realtimeregs = ast_check_realtime("sipregs");
+ realtimertpqos = ast_check_realtime("rtpcqr");
ast_mutex_lock(&authl_lock);
credentials = authl;
@@ -18854,6 +18887,8 @@
}
ast_cli(a->fd, " Record SIP history: %s\n", AST_CLI_ONOFF(recordhistory));
ast_cli(a->fd, " Call Events: %s\n", AST_CLI_ONOFF(sip_cfg.callevents));
+ ast_cli(a->fd, " RTCP Events: %s\n", AST_CLI_ONOFF(sip_cfg.rtcpevents));
+ ast_cli(a->fd, " RTCP Event timer: %d\n", sip_cfg.rtcptimer);
ast_cli(a->fd, " Auth. Failure Events: %s\n", AST_CLI_ONOFF(global_authfailureevents));
ast_cli(a->fd, " T.38 support: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT)));
@@ -18863,6 +18898,7 @@
ast_cli(a->fd, " SIP realtime: Disabled\n" );
else
ast_cli(a->fd, " SIP realtime: Enabled\n" );
+ ast_cli(a->fd, " QoS realtime reports: %s\n", realtimertpqos ? "Enabled" : "Disabled" );
ast_cli(a->fd, " Qualify Freq : %d ms\n", global_qualifyfreq);
ast_cli(a->fd, " Q.850 Reason header: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[1], SIP_PAGE2_Q850_REASON)));
ast_cli(a->fd, " Store SIP_CAUSE: %s\n", AST_CLI_YESNO(global_store_sip_cause));
@@ -20853,6 +20889,7 @@
if (!req->ignore && p->invitestate != INV_CANCELLED && sip_cancel_destroy(p))
ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n");
check_pendings(p);
+ start_rtcp_events(p, sched);
break;
case 180: /* 180 Ringing */
@@ -21140,6 +21177,7 @@
ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, TRUE);
check_pendings(p);
+ start_rtcp_events(p, sched);
break;
case 407: /* Proxy authentication */
@@ -21803,16 +21841,21 @@
}
}
-/*! \brief Immediately stop RTP, VRTP and UDPTL as applicable */
+/*! \brief Immediately stop RTP, VRTP, TEXT and UDPTL as applicable */
static void stop_media_flows(struct sip_pvt *p)
{
/* Immediately stop RTP, VRTP and UDPTL as applicable */
- if (p->rtp)
+ if (p->rtp && !ast_rtp_instance_isactive(p->rtp)) {
+ sip_rtcp_report(p, p->rtp, SDP_AUDIO, TRUE);
ast_rtp_instance_stop(p->rtp);
- if (p->vrtp)
+ }
+ if (p->vrtp && !ast_rtp_instance_isactive(p->vrtp)) {
+ sip_rtcp_report(p, p->vrtp, SDP_VIDEO, TRUE);
ast_rtp_instance_stop(p->vrtp);
- if (p->trtp)
+ }
+ if (p->trtp && !ast_rtp_instance_isactive(p->trtp)) {
ast_rtp_instance_stop(p->trtp);
+ }
if (p->udptl)
ast_udptl_stop(p->udptl);
}
@@ -24839,6 +24882,7 @@
if (p->rtp && (quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
+ sip_rtcp_report(p, p->rtp, SDP_AUDIO, TRUE);
if (p->do_history) {
append_history(p, "RTCPaudio", "Quality:%s", quality);
@@ -24869,6 +24913,7 @@
}
if (p->vrtp && (quality = ast_rtp_instance_get_quality(p->vrtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
+ sip_rtcp_report(p, p->rtp, SDP_VIDEO, TRUE);
if (p->do_history) {
append_history(p, "RTCPvideo", "Quality:%s", quality);
}
@@ -29188,6 +29233,8 @@
/* Misc settings for the channel */
global_relaxdtmf = FALSE;
sip_cfg.callevents = DEFAULT_CALLEVENTS;
+ sip_cfg.rtcpevents = FALSE;
+ sip_cfg.rtcptimer = 0; /* Only report at end of call (default) */
global_authfailureevents = FALSE;
global_t1 = DEFAULT_TIMER_T1;
global_timer_b = 64 * DEFAULT_TIMER_T1;
@@ -29616,6 +29663,13 @@
} else {
ast_log(LOG_WARNING, "Invalid qualifyfreq number '%s' at line %d of %s\n", v->value, v->lineno, config);
global_qualifyfreq = DEFAULT_QUALIFYFREQ;
+ }
+ } else if (!strcasecmp(v->name, "rtcpevents")) {
+ sip_cfg.rtcpevents = ast_true(v->value);
+ } else if (!strcasecmp(v->name, "rtcpeventtimer")) {
+ if (sscanf(v->value, "%30d", &sip_cfg.rtcptimer) != 1) {
+ ast_log(LOG_WARNING, "RTCP event timer needs to be value (seconds between reports) at line %d of sip.conf\n", v->lineno);
+ sip_cfg.rtcptimer = 0;
}
} else if (!strcasecmp(v->name, "callevents")) {
sip_cfg.callevents = ast_true(v->value);
Modified: team/oej/teapot-1.8/channels/sip/dialplan_functions.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/teapot-1.8/channels/sip/dialplan_functions.c?view=diff&rev=402885&r1=402884&r2=402885
==============================================================================
--- team/oej/teapot-1.8/channels/sip/dialplan_functions.c (original)
+++ team/oej/teapot-1.8/channels/sip/dialplan_functions.c Tue Nov 19 02:28:49 2013
@@ -140,7 +140,9 @@
snprintf(buf, buflen, "%s", ast_sockaddr_stringify(&sa));
} else if (!strcasecmp(args.param, "rtpqos")) {
+/* OEJ - Check this part, different from 1.4 */
struct ast_rtp_instance *rtp = NULL;
+ struct ast_rtp_quality *qos;
if (ast_strlen_zero(args.type)) {
args.type = "audio";
@@ -162,6 +164,11 @@
if (!ast_rtp_instance_get_quality(rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf))) {
return -1;
}
+#ifdef OEJ
+ if (!ast_rtp_instance_get_qualdata(rtp, ???, &qos)) {
+ this_needs_some_love;
+ }
+#endif
ast_copy_string(buf, quality_buf, buflen);
return res;
Modified: team/oej/teapot-1.8/channels/sip/include/sip.h
URL: http://svnview.digium.com/svn/asterisk/team/oej/teapot-1.8/channels/sip/include/sip.h?view=diff&rev=402885&r1=402884&r2=402885
==============================================================================
--- team/oej/teapot-1.8/channels/sip/include/sip.h (original)
+++ team/oej/teapot-1.8/channels/sip/include/sip.h Tue Nov 19 02:28:49 2013
@@ -706,6 +706,8 @@
int compactheaders; /*!< send compact sip headers */
int allow_external_domains; /*!< Accept calls to external SIP domains? */
int callevents; /*!< Whether we send manager events or not */
+ int rtcpevents; /*!< Whether we send manager RTCP events or not */
+ int rtcptimer; /*!< How often, during a call, to report RTCP stats */
int regextenonqualify; /*!< Whether to add/remove regexten when qualifying peers */
int legacy_useroption_parsing; /*!< Whether to strip useroptions in URI via semicolons */
int matchexternaddrlocally; /*!< Match externaddr/externhost setting against localnet setting */
@@ -1096,6 +1098,7 @@
int waitid; /*!< Wait ID for scheduler after 491 or other delays */
int reinviteid; /*!< Reinvite in case of provisional, but no final response */
int autokillid; /*!< Auto-kill ID (scheduler) */
+ int rtcpeventid; /*!< Scheduler ID for RTCP Events */
int t38id; /*!< T.38 Response ID */
struct sip_refer *refer; /*!< REFER: SIP transfer data structure */
enum subscriptiontype subscribed; /*!< SUBSCRIBE: Is this dialog a subscription? */
@@ -1151,6 +1154,12 @@
* The large-scale changes would be a good idea for implementing during an SDP rewrite.
*/
struct offered_media offered_media[OFFERED_MEDIA_COUNT];
+ //struct ast_rtp_quality *audioqual; /*!< Audio: The latest quality report, for realtime storage */
+ //struct ast_rtp_quality *videoqual; /*!< Video: The latest quality report, for realtime storage */
+ struct ast_rtp_instance_stats *audioqual; /*!< Audio: The latest quality report, for realtime storage */
+ struct ast_rtp_instance_stats *videoqual; /*!< Video: The latest quality report, for realtime storage */
+ struct sip_settings *sip_cfg; /*! Which sip_cfg is associated with this dialog */
+
struct ast_cc_config_params *cc_params;
struct sip_epa_entry *epa_entry;
int fromdomainport; /*!< Domain port to show in from field */
Modified: team/oej/teapot-1.8/configs/extconfig.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/oej/teapot-1.8/configs/extconfig.conf.sample?view=diff&rev=402885&r1=402884&r2=402885
==============================================================================
--- team/oej/teapot-1.8/configs/extconfig.conf.sample (original)
+++ team/oej/teapot-1.8/configs/extconfig.conf.sample Tue Nov 19 02:28:49 2013
@@ -80,6 +80,7 @@
;queue_members => odbc,asterisk
;musiconhold => mysql,general
;queue_log => mysql,general
+;rtpcqr => mysql,general ; RTP Call Quality Records
;
;
; While most dynamic realtime engines are automatically used when defined in
Modified: team/oej/teapot-1.8/include/asterisk/logger.h
URL: http://svnview.digium.com/svn/asterisk/team/oej/teapot-1.8/include/asterisk/logger.h?view=diff&rev=402885&r1=402884&r2=402885
==============================================================================
--- team/oej/teapot-1.8/include/asterisk/logger.h (original)
+++ team/oej/teapot-1.8/include/asterisk/logger.h Tue Nov 19 02:28:49 2013
@@ -181,7 +181,18 @@
#endif
#define AST_LOG_DTMF __LOG_DTMF, _A_
-#define NUMLOGLEVELS 32
+#ifdef LOG_CQR
+#undef LOG_CQR
+#endif
+#define __LOG_CQR 7
+#define LOG_CQR __LOG_CQR, _A_
+
+#ifdef AST_LOG_CQR
+#undef AST_LOG_CQR
+#endif
+#define AST_LOG_CQR __LOG_CQR, _A_
+
+#define NUMLOGLEVELS 64 /* Highest bit */
/*!
* \brief Get the debug level for a module
Modified: team/oej/teapot-1.8/include/asterisk/rtp_engine.h
URL: http://svnview.digium.com/svn/asterisk/team/oej/teapot-1.8/include/asterisk/rtp_engine.h?view=diff&rev=402885&r1=402884&r2=402885
==============================================================================
--- team/oej/teapot-1.8/include/asterisk/rtp_engine.h (original)
+++ team/oej/teapot-1.8/include/asterisk/rtp_engine.h Tue Nov 19 02:28:49 2013
@@ -74,6 +74,7 @@
#include "asterisk/netsock2.h"
#include "asterisk/sched.h"
#include "asterisk/res_srtp.h"
+#include "asterisk/channel.h"
/* Maximum number of payloads supported */
#define AST_RTP_MAX_PT 256
@@ -211,6 +212,14 @@
AST_RTP_INSTANCE_STAT_LOCAL_SSRC,
/*! Retrieve remote SSRC */
AST_RTP_INSTANCE_STAT_REMOTE_SSRC,
+ /*! Retrieve local CNAME */
+ AST_RTP_INSTANCE_STAT_LOCAL_CNAME,
+ /*! Retrieve remote SDES */
+ AST_RTP_INSTANCE_STAT_REMOTE_CNAME,
+ /*! Retrieve start time */
+ AST_RTP_INSTANCE_STAT_START,
+ /*! Retrieve IP Address */
+ AST_RTP_INSTANCE_STAT_IP,
};
/* Codes for RTP-specific data - not defined by our AST_FORMAT codes */
@@ -237,10 +246,12 @@
unsigned int txcount;
/*! Number of packets received */
unsigned int rxcount;
+
/*! Jitter on transmitted packets */
double txjitter;
/*! Jitter on received packets */
double rxjitter;
+
/*! Maximum jitter on remote side */
double remote_maxjitter;
/*! Minimum jitter on remote side */
@@ -291,6 +302,32 @@
unsigned int local_ssrc;
/*! Their SSRC */
unsigned int remote_ssrc;
+
+ /* --- Pinefrog additions */
+ /*! Remote: Number of packets transmitted */
+ unsigned int remote_txcount;
+ /*! Remote: Number of packets received */
+ unsigned int remote_rxcount;
+ char channel[AST_MAX_EXTENSION]; /*!< Name of channel */
+ char uniqueid[AST_MAX_EXTENSION]; /*!< uniqueid of channel */
+ char bridgedchannel[AST_MAX_EXTENSION]; /*!< Name of bridged channel */
+ char bridgeduniqueid[AST_MAX_EXTENSION]; /*!< uniqueid of bridged channel */
+ unsigned int numberofreports; /*!< Number of reports received from remote end */
+ int lasttxformat; /*!< Last used codec on transmitted stream */
+ int lastrxformat; /*!< Last used codec on received stream */
+ struct sockaddr_in them; /*!< The IP address used for media by remote end */
+ struct sockaddr_in us; /*!< The IP address used for media by our end */
+ char ourcname[255]; /*!< Our SDES RTP session name (CNAME) */
+ size_t ourcnamelength; /*!< Length of CNAME (utf8) */
+ char theircname[255]; /*!< Their SDES RTP session name (CNAME) */
+ size_t theircnamelength; /*!< Length of CNAME (utf8) */
+ struct timeval start; /*!< When the stream started */
+ struct timeval end; /*!< When the stream ended */
+ 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 */
+ int mediatype; /*! Type of media */
};
#define AST_RTP_STAT_SET(current_stat, combined, placement, value) \
@@ -318,6 +355,9 @@
int (*destroy)(struct ast_rtp_instance *instance);
/*! Callback for writing out a frame */
int (*write)(struct ast_rtp_instance *instance, struct ast_frame *frame);
+ /*! Callback for stopping the outbound RTP media for an instance,
+ but keeping the RTCP flow (and the RTP keepalives if needed) */
+ void (*hold)(struct ast_rtp_instance *instance, int status);
/*! Callback for stopping the RTP instance */
void (*stop)(struct ast_rtp_instance *instance);
/*! Callback for starting RFC2833 DTMF transmission */
@@ -377,6 +417,15 @@
format_t (*available_formats)(struct ast_rtp_instance *instance, format_t to_endpoint, format_t to_asterisk);
/*! Callback to send CNG */
int (*sendcng)(struct ast_rtp_instance *instance, int level);
+ /*! Callback to check if a media stram is active */
+ int (*isactive)(struct ast_rtp_instance *instance);
+ /*! Callback to set CNAME in rtcp */
+ void (*setcname)(struct ast_rtp_instance *instance, const char *cname, size_t length);
+ /*! Callback to set information about bridged channel for CQR record */
+ void (*set_bridged_chan)(struct ast_rtp_instance *instance, const char *channel, const char *uniqueid, const char *bridgedchan, const char *bridgeduniqueid);
+ /*! Callback to set translation information for the CQR record */
+ void (*set_translator) (struct ast_rtp_instance *instance, const char *readtranslator, const int readcost, const char *writetranslator, const int writecost);
+ int (*rtcp_write_empty)(struct ast_rtp_instance *instance);
/*! Linked list information */
AST_RWLIST_ENTRY(ast_rtp_engine) entry;
};
@@ -1311,6 +1360,26 @@
int ast_rtp_instance_set_qos(struct ast_rtp_instance *instance, int tos, int cos, const char *desc);
/*!
+ * \brief Stop the RTP outbound media in a stream, but keep the RTCP flow going
+ * And propably RTP keepalives too.
+ *
+ * \param instance Instance that media is no longer going to at this time
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_instance_stop(instance);
+ * \endcode
+ *
+ * This tells the RTP engine being used for the instance pointed to by instance
+ * that media is no longer going to it at this time, but may in the future.
+ * Keep the RTCP flow happy
+ *
+ * \since 1.42
+ */
+void ast_rtp_instance_hold(struct ast_rtp_instance *instance, int status);
+
+/*!
* \brief Stop an RTP instance
*
* \param instance Instance that media is no longer going to at this time
@@ -1834,6 +1903,36 @@
int ast_rtp_instance_sendcng(struct ast_rtp_instance *instance, int level);
/*!
+ * \brief Send empty RTCP report
+ *
+ * \param instance The RTP instance
+ * \param fd File descriptor to use
+ *
+ * \retval 0 Success
+ * \retval non-zero Failure
+ */
+int ast_rtcp_write_empty(struct ast_rtp_instance *instance);
+
+
+/*!
+ * \brief Check if RTP stream is active
+ *
+ * \param instance The RTP instance
+ *
+ * \retval 0 Active (success)
+ * \retval -1 Not supported by RTP engine, 1 Not active
+ */
+int ast_rtp_instance_isactive(struct ast_rtp_instance *instance);
+
+/*!
+ * \brief Set the name of the RTP session (used in RTCP)
+ * \param cname Session name (UTF 8 possible)
+ * \param length Name of string (needed for UTF 8 always)
+ *
+ */
+int ast_rtp_instance_setcname(struct ast_rtp_instance *instance, const char *cname, size_t length);
+
+/*!
* \brief Add or replace the SRTP policies for the given RTP instance
*
* \param instance the RTP instance
@@ -1854,6 +1953,22 @@
*/
struct ast_srtp *ast_rtp_instance_get_srtp(struct ast_rtp_instance *instance);
+/*!
+ * \brief set the channel information for the CQR records
+ *
+ * \retval 0 on success
+ * \retval -1 not implemented by RTP engine
+ */
+int ast_rtp_instance_set_bridged_chan(struct ast_rtp_instance *instance, const char *channel, const char *uniqueid, const char *bridgedchan, const char *bridgeduniqueid);
+
+/*!
+ * \brief set the channel translator information for the CQR records
+ *
+ * \retval 0 on success
+ * \retval -1 not implemented by RTP engine
+ */
+int ast_rtp_instance_set_translator(struct ast_rtp_instance *instance, const char *readtranslator, const int readcost, const char *writetranslator, const int writecost);
+
#if defined(__cplusplus) || defined(c_plusplus)
}
#endif
Modified: team/oej/teapot-1.8/main/logger.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/teapot-1.8/main/logger.c?view=diff&rev=402885&r1=402884&r2=402885
==============================================================================
--- team/oej/teapot-1.8/main/logger.c (original)
+++ team/oej/teapot-1.8/main/logger.c Tue Nov 19 02:28:49 2013
@@ -174,6 +174,7 @@
"ERROR",
"VERBOSE",
"DTMF",
+ "CQR",
};
/*! \brief Colors used in the console for logging */
@@ -185,7 +186,7 @@
COLOR_RED,
COLOR_GREEN,
COLOR_BRGREEN,
- 0,
+ COLOR_BRBLUE,
0,
0,
0,
Modified: team/oej/teapot-1.8/main/rtp_engine.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/teapot-1.8/main/rtp_engine.c?view=diff&rev=402885&r1=402884&r2=402885
==============================================================================
--- team/oej/teapot-1.8/main/rtp_engine.c (original)
+++ team/oej/teapot-1.8/main/rtp_engine.c Tue Nov 19 02:28:49 2013
@@ -21,6 +21,9 @@
* \brief Pluggable RTP Architecture
*
* \author Joshua Colp <jcolp at digium.com>
+ *
+ * Improved RTCP support by
+ * \author Olle E. Johansson <oej at edvina.net>
*/
/*** MODULEINFO
@@ -776,6 +779,13 @@
return instance->engine->qos ? instance->engine->qos(instance, tos, cos, desc) : -1;
}
+void ast_rtp_instance_hold(struct ast_rtp_instance *instance, int status)
+{
+ if (instance->engine->hold) {
+ instance->engine->hold(instance, status);
+ }
+}
+
void ast_rtp_instance_stop(struct ast_rtp_instance *instance)
{
if (instance->engine->stop) {
@@ -829,6 +839,7 @@
return AST_BRIDGE_FAILED_NOWARN;
}
+ /* Now let go of the channel locks and be on our way */
ast_channel_unlock(c0);
ast_channel_unlock(c1);
@@ -836,6 +847,10 @@
instance1->bridged = instance0;
ast_poll_channel_add(c0, c1);
+
+ /* Kick the RTCP stream going by sending one empty stupid little packet */
+ ast_rtcp_write_empty(instance0);
+ ast_rtcp_write_empty(instance1);
/* Hop into a loop waiting for a frame from either channel */
cs[0] = c0;
@@ -1836,11 +1851,58 @@
return instance->srtp;
}
+int ast_rtp_instance_isactive(struct ast_rtp_instance *instance)
+{
+ if (instance->engine->isactive) {
+ return instance->engine->isactive(instance);
+ }
+
+ return -1;
+}
+
+int ast_rtcp_write_empty(struct ast_rtp_instance *instance)
+{
+ if (instance->engine->rtcp_write_empty) {
+ instance->engine->rtcp_write_empty(instance);
+ return 0;
+ }
+ return -1;
+}
+
int ast_rtp_instance_sendcng(struct ast_rtp_instance *instance, int level)
{
if (instance->engine->sendcng) {
return instance->engine->sendcng(instance, level);
}
-
return -1;
}
+
+int ast_rtp_instance_setcname(struct ast_rtp_instance *instance, const char *cname, size_t length)
+{
+ if (instance->engine->setcname) {
+ instance->engine->setcname(instance, cname, length);
+ return 0;
+ }
+
+ return -1; /* Function does not exist */
+}
+
+int ast_rtp_instance_set_bridged_chan(struct ast_rtp_instance *instance, const char *channel, const char *uniqueid, const char *bridgedchan, const char *bridgeduniqueid)
+{
+ if (instance->engine->set_bridged_chan) {
+ instance->engine->set_bridged_chan(instance, channel, uniqueid, bridgedchan, bridgeduniqueid);
+ return 0;
+ }
+
+ return -1; /* Function does not exist */
+}
+
+int ast_rtp_instance_set_translator(struct ast_rtp_instance *instance, const char *readtranslator, const int readcost, const char *writetranslator, const int writecost)
+{
+ if (instance->engine->set_translator) {
+ instance->engine->set_translator(instance, readtranslator, readcost, writetranslator, writecost);
+ return 0;
+ }
+
+ return -1; /* Function does not exist */
+}
Added: team/oej/teapot-1.8/patches/pinefrog-rtcp-1.8.diff
URL: http://svnview.digium.com/svn/asterisk/team/oej/teapot-1.8/patches/pinefrog-rtcp-1.8.diff?view=auto&rev=402885
==============================================================================
--- team/oej/teapot-1.8/patches/pinefrog-rtcp-1.8.diff (added)
+++ team/oej/teapot-1.8/patches/pinefrog-rtcp-1.8.diff Tue Nov 19 02:28:49 2013
@@ -1,0 +1,2651 @@
+Index: channels/chan_sip.c
+===================================================================
+--- channels/chan_sip.c (.../branches/1.8) (revision 402883)
++++ channels/chan_sip.c (.../team/oej/pinefrog-rtcp-1.8) (revision 402883)
+@@ -275,6 +275,7 @@
+ #include "asterisk/xml.h"
+ #include "sip/include/dialog.h"
+ #include "sip/include/dialplan_functions.h"
++#include "sip/include/rtcp.h"
+
+
+ /*** DOCUMENTATION
+@@ -670,7 +671,6 @@
+ { AST_REDIRECTING_REASON_CALL_FWD_DTE, "unknown"}
+ };
+
+-
+ /*! \name DefaultSettings
+ Default setttings are used as a channel setting and as a default when
+ configuring devices
+@@ -6175,6 +6175,22 @@
+ if (dumphistory)
+ sip_dump_history(p);
+
++ AST_SCHED_DEL(sched, p->rtcpeventid);
++
++ if (p->audioqual) {
++ /* We have a quality report to write to realtime before we leave this world. */
++ qos_write_realtime(p, p->audioqual);
++ free(p->audioqual);
++ p->audioqual = NULL;
++ }
++ if (p->videoqual) {
++ /* We have a quality report to write to realtime before we leave this world. */
++ qos_write_realtime(p, p->videoqual);
++ free(p->videoqual);
++ p->videoqual = NULL;
++ }
++
++
+ if (p->options) {
+ if (p->options->outboundproxy) {
+ ao2_ref(p->options->outboundproxy, -1);
+@@ -6901,6 +6917,7 @@
+ ast_rtp_instance_update_source(p->rtp);
+ res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL, oldsdp, TRUE);
+ ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
++ start_rtcp_events(p, sched);
+ }
+ sip_pvt_unlock(p);
+ return res;
+@@ -8156,8 +8173,10 @@
+ set_socket_transport(&p->socket, SIP_TRANSPORT_UDP);
+ }
+
++ p->sip_cfg = &sip_cfg;
+ p->socket.fd = -1;
+ p->method = intended_method;
++ p->rtcpeventid = -1;
+ p->initid = -1;
+ p->waitid = -1;
+ p->reinviteid = -1;
+@@ -8219,6 +8238,14 @@
+ build_callid_pvt(p);
+ else
+ ast_string_field_set(p, callid, callid);
++
++ /* Set cnames for the RTCP SDES */
++ if (p->rtp) {
++ ast_rtp_instance_setcname(p->rtp, p->callid, strlen(p->callid));
++ }
++ if (p->vrtp) {
++ ast_rtp_instance_setcname(p->vrtp, p->callid, strlen(p->callid));
++ }
+ /* Assign default music on hold class */
+ ast_string_field_set(p, mohinterpret, default_mohinterpret);
+ ast_string_field_set(p, mohsuggest, default_mohsuggest);
+@@ -9142,6 +9169,7 @@
+ dialog->owner->uniqueid);
+ append_history(dialog, holdstate ? "Hold" : "Unhold", "%s", ast_str_buffer(req->data));
+ if (!holdstate) { /* Put off remote hold */
++ ast_rtp_instance_hold(dialog->rtp, 0); /* Turn off RTP hold */
+ ast_clear_flag(&dialog->flags[1], SIP_PAGE2_CALL_ONHOLD); /* Clear both flags */
+ return;
+ }
+@@ -9910,9 +9938,9 @@
+ ast_queue_control_data(p->owner, AST_CONTROL_HOLD,
+ S_OR(p->mohsuggest, NULL),
+ !ast_strlen_zero(p->mohsuggest) ? strlen(p->mohsuggest) + 1 : 0);
+- if (sendonly)
+- ast_rtp_instance_stop(p->rtp);
+- /* RTCP needs to go ahead, even if we're on hold!!! */
++ if (sendonly == 1 || sendonly == 2) { /* sendonly (from the other side) or inactive */
++ ast_rtp_instance_hold(p->rtp, 1);
++ }
+ /* Activate a re-invite */
+ ast_queue_frame(p->owner, &ast_null_frame);
+ change_hold_state(p, req, TRUE, sendonly);
+@@ -18148,8 +18176,10 @@
+ int x = 0, load_realtime;
+ format_t codec = 0;
+ int realtimepeers;
++ int realtimertpqos = FALSE;
+
+ realtimepeers = ast_check_realtime("sippeers");
++ realtimertpqos = ast_check_realtime("rtpcqr");
+
+ if (argc < 4)
+ return CLI_SHOWUSAGE;
+@@ -18764,10 +18794,12 @@
+ {
+ int realtimepeers;
+ int realtimeregs;
++ int realtimertpqos;
+ char codec_buf[SIPBUFSIZE];
+ const char *msg; /* temporary msg pointer */
+ struct sip_auth_container *credentials;
+
++
+ switch (cmd) {
+ case CLI_INIT:
+ e->command = "sip show settings";
+@@ -18784,6 +18816,7 @@
+
+ realtimepeers = ast_check_realtime("sippeers");
+ realtimeregs = ast_check_realtime("sipregs");
++ realtimertpqos = ast_check_realtime("rtpcqr");
+
+ ast_mutex_lock(&authl_lock);
+ credentials = authl;
+@@ -18853,6 +18886,8 @@
+ }
+ ast_cli(a->fd, " Record SIP history: %s\n", AST_CLI_ONOFF(recordhistory));
+ ast_cli(a->fd, " Call Events: %s\n", AST_CLI_ONOFF(sip_cfg.callevents));
++ ast_cli(a->fd, " RTCP Events: %s\n", AST_CLI_ONOFF(sip_cfg.rtcpevents));
++ ast_cli(a->fd, " RTCP Event timer: %d\n", sip_cfg.rtcptimer);
+ ast_cli(a->fd, " Auth. Failure Events: %s\n", AST_CLI_ONOFF(global_authfailureevents));
+
+ ast_cli(a->fd, " T.38 support: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT)));
+@@ -18862,6 +18897,7 @@
+ ast_cli(a->fd, " SIP realtime: Disabled\n" );
+ else
+ ast_cli(a->fd, " SIP realtime: Enabled\n" );
++ ast_cli(a->fd, " QoS realtime reports: %s\n", realtimertpqos ? "Enabled" : "Disabled" );
+ ast_cli(a->fd, " Qualify Freq : %d ms\n", global_qualifyfreq);
+ ast_cli(a->fd, " Q.850 Reason header: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[1], SIP_PAGE2_Q850_REASON)));
+ ast_cli(a->fd, " Store SIP_CAUSE: %s\n", AST_CLI_YESNO(global_store_sip_cause));
+@@ -20803,6 +20839,7 @@
+ if (!req->ignore && p->invitestate != INV_CANCELLED && sip_cancel_destroy(p))
+ ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n");
+ check_pendings(p);
++ start_rtcp_events(p, sched);
+ break;
+
+ case 180: /* 180 Ringing */
+@@ -21080,6 +21117,7 @@
+ ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
+ xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, TRUE);
+ check_pendings(p);
++ start_rtcp_events(p, sched);
+ break;
+
+ case 407: /* Proxy authentication */
+@@ -21730,16 +21768,21 @@
+ }
+ }
+
+-/*! \brief Immediately stop RTP, VRTP and UDPTL as applicable */
++/*! \brief Immediately stop RTP, VRTP, TEXT and UDPTL as applicable */
+ static void stop_media_flows(struct sip_pvt *p)
+ {
+ /* Immediately stop RTP, VRTP and UDPTL as applicable */
+- if (p->rtp)
++ if (p->rtp && !ast_rtp_instance_isactive(p->rtp)) {
++ sip_rtcp_report(p, p->rtp, SDP_AUDIO, TRUE);
+ ast_rtp_instance_stop(p->rtp);
+- if (p->vrtp)
++ }
++ if (p->vrtp && !ast_rtp_instance_isactive(p->vrtp)) {
++ sip_rtcp_report(p, p->vrtp, SDP_VIDEO, TRUE);
+ ast_rtp_instance_stop(p->vrtp);
+- if (p->trtp)
++ }
++ if (p->trtp && !ast_rtp_instance_isactive(p->trtp)) {
+ ast_rtp_instance_stop(p->trtp);
++ }
+ if (p->udptl)
+ ast_udptl_stop(p->udptl);
+ }
+@@ -24660,6 +24703,7 @@
+
+
+ if (p->rtp && (quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
++ sip_rtcp_report(p, p->rtp, SDP_AUDIO, TRUE);
+ if (p->do_history) {
+ append_history(p, "RTCPaudio", "Quality:%s", quality);
+
+@@ -24690,6 +24734,7 @@
+ }
+
+ if (p->vrtp && (quality = ast_rtp_instance_get_quality(p->vrtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
++ sip_rtcp_report(p, p->rtp, SDP_VIDEO, TRUE);
+ if (p->do_history) {
+ append_history(p, "RTCPvideo", "Quality:%s", quality);
+ }
+@@ -28999,6 +29044,8 @@
+ /* Misc settings for the channel */
+ global_relaxdtmf = FALSE;
+ sip_cfg.callevents = DEFAULT_CALLEVENTS;
++ sip_cfg.rtcpevents = FALSE;
++ sip_cfg.rtcptimer = 0; /* Only report at end of call (default) */
+ global_authfailureevents = FALSE;
+ global_t1 = DEFAULT_TIMER_T1;
+ global_timer_b = 64 * DEFAULT_TIMER_T1;
+@@ -29426,6 +29473,13 @@
+ ast_log(LOG_WARNING, "Invalid qualifyfreq number '%s' at line %d of %s\n", v->value, v->lineno, config);
+ global_qualifyfreq = DEFAULT_QUALIFYFREQ;
+ }
++ } else if (!strcasecmp(v->name, "rtcpevents")) {
++ sip_cfg.rtcpevents = ast_true(v->value);
++ } else if (!strcasecmp(v->name, "rtcpeventtimer")) {
++ if (sscanf(v->value, "%30d", &sip_cfg.rtcptimer) != 1) {
++ ast_log(LOG_WARNING, "RTCP event timer needs to be value (seconds between reports) at line %d of sip.conf\n", v->lineno);
++ sip_cfg.rtcptimer = 0;
++ }
+ } else if (!strcasecmp(v->name, "callevents")) {
+ sip_cfg.callevents = ast_true(v->value);
+ } else if (!strcasecmp(v->name, "authfailureevents")) {
+Index: channels/sip/include/sip.h
+===================================================================
+--- channels/sip/include/sip.h (.../branches/1.8) (revision 402883)
++++ channels/sip/include/sip.h (.../team/oej/pinefrog-rtcp-1.8) (revision 402883)
+@@ -696,6 +696,8 @@
+ int compactheaders; /*!< send compact sip headers */
+ int allow_external_domains; /*!< Accept calls to external SIP domains? */
+ int callevents; /*!< Whether we send manager events or not */
++ int rtcpevents; /*!< Whether we send manager RTCP events or not */
++ int rtcptimer; /*!< How often, during a call, to report RTCP stats */
+ int regextenonqualify; /*!< Whether to add/remove regexten when qualifying peers */
+ int legacy_useroption_parsing; /*!< Whether to strip useroptions in URI via semicolons */
+ int matchexternaddrlocally; /*!< Match externaddr/externhost setting against localnet setting */
+@@ -1080,6 +1082,7 @@
+ int waitid; /*!< Wait ID for scheduler after 491 or other delays */
+ int reinviteid; /*!< Reinvite in case of provisional, but no final response */
+ int autokillid; /*!< Auto-kill ID (scheduler) */
++ int rtcpeventid; /*!< Scheduler ID for RTCP Events */
+ int t38id; /*!< T.38 Response ID */
+ struct sip_refer *refer; /*!< REFER: SIP transfer data structure */
+ enum subscriptiontype subscribed; /*!< SUBSCRIBE: Is this dialog a subscription? */
+@@ -1135,6 +1138,12 @@
+ * The large-scale changes would be a good idea for implementing during an SDP rewrite.
+ */
+ struct offered_media offered_media[OFFERED_MEDIA_COUNT];
++ //struct ast_rtp_quality *audioqual; /*!< Audio: The latest quality report, for realtime storage */
++ //struct ast_rtp_quality *videoqual; /*!< Video: The latest quality report, for realtime storage */
++ struct ast_rtp_instance_stats *audioqual; /*!< Audio: The latest quality report, for realtime storage */
++ struct ast_rtp_instance_stats *videoqual; /*!< Video: The latest quality report, for realtime storage */
++ struct sip_settings *sip_cfg; /*! Which sip_cfg is associated with this dialog */
++
+ struct ast_cc_config_params *cc_params;
+ struct sip_epa_entry *epa_entry;
+ int fromdomainport; /*!< Domain port to show in from field */
+Index: channels/sip/include/rtcp.h
+===================================================================
+--- channels/sip/include/rtcp.h (.../branches/1.8) (revision 0)
++++ channels/sip/include/rtcp.h (.../team/oej/pinefrog-rtcp-1.8) (revision 402883)
+@@ -0,0 +1,50 @@
++/*
++ * Asterisk -- An open source telephony toolkit.
++ *
++ * Copyright (C) 2013 Olle E. Johansson, Edvina AB
++ *
++ * See http://www.asterisk.org for more information about
++ * the Asterisk project. Please do not directly contact
++ * any of the maintainers of this project for assistance;
++ * the project provides a web site, mailing lists and IRC
++ * channels for your use.
++ *
++ * This program is free software, distributed under the terms of
++ * the GNU General Public License Version 2. See the LICENSE file
++ * at the top of the source tree.
++ */
++
++/*! \file rtcp.h
++ *
++ * \brief RTCP additional functions
++ *
++ * \author Olle E. Johansson <oej at edvina.net>
++ */
++
++#include "asterisk.h"
++
++#include "asterisk/utils.h"
++#include "asterisk/rtp_engine.h"
++#include "sip.h"
++
++#ifndef _SIP_RTCP_H
++#define _SIP_RTCP_H
++
++/*! \brief Set various data items in the RTP structure, like channel identifier.
++ */
++void sip_rtcp_set_data(struct sip_pvt *dialog, struct ast_rtp_instance *instance, enum media_type type);
++
++int send_rtcp_events(const void *data);
++void start_rtcp_events(struct sip_pvt *dialog, struct sched_context *sched);
++/*
++# For 1.4:
++# static void sip_rtcp_report(struct sip_pvt *p, struct ast_rtp *rtp, enum media_type type, int reporttype);
++*/
++
++void sip_rtcp_report(struct sip_pvt *dialog, struct ast_rtp_instance *instance, enum media_type type, int reporttype);
++//void qos_write_realtime(struct sip_pvt *dialog, struct ast_rtp_quality *qual);
++void qos_write_realtime(struct sip_pvt *dialog, struct ast_rtp_instance_stats *qual);
++
++
++
++#endif /* _SIP_RTCP_H */
+
+Property changes on: channels/sip/include/rtcp.h
+___________________________________________________________________
+Added: svn:mime-type
+## -0,0 +1 ##
++text/plain
+\ No newline at end of property
+Added: svn:keywords
+## -0,0 +1 ##
++Author Date Id Revision
+\ No newline at end of property
+Added: svn:eol-style
[... 3614 lines stripped ...]
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