[svn-commits] tilghman: trunk r247124 - in /trunk/channels: ./ sip/ sip/include/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Wed Feb 17 00:25:22 CST 2010
Author: tilghman
Date: Wed Feb 17 00:25:15 2010
New Revision: 247124
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=247124
Log:
Make all of the various rtpqos parameters in this branch available from the CHANNEL function.
Also includes a test for retrieving rtpqos parameters, including a NULL RTP
driver. Additionally, some further separation of the SIP internal API into
headers was necessary.
(closes issue #16652)
Reported by: kkm
Patches:
20100204__issue16652.diff.txt uploaded by tilghman (license 14)
Review: https://reviewboard.asterisk.org/r/501/
Added:
trunk/channels/sip/dialplan_functions.c (with props)
trunk/channels/sip/include/dialog.h (with props)
trunk/channels/sip/include/dialplan_functions.h (with props)
trunk/channels/sip/include/globals.h (with props)
Modified:
trunk/channels/Makefile
trunk/channels/chan_sip.c
trunk/channels/sip/include/config_parser.h
trunk/channels/sip/include/sip_utils.h
Modified: trunk/channels/Makefile
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/Makefile?view=diff&rev=247124&r1=247123&r2=247124
==============================================================================
--- trunk/channels/Makefile (original)
+++ trunk/channels/Makefile Wed Feb 17 00:25:15 2010
@@ -70,7 +70,7 @@
rm -f h323/Makefile
$(if $(filter chan_iax2,$(EMBEDDED_MODS)),modules.link,chan_iax2.so): iax2-parser.o iax2-provision.o
-$(if $(filter chan_sip,$(EMBEDDED_MODS)),modules.link,chan_sip.so): sip/config_parser.o sip/reqresp_parser.o
+$(if $(filter chan_sip,$(EMBEDDED_MODS)),modules.link,chan_sip.so): $(subst .c,.o,$(wildcard sip/*.c))
$(if $(filter chan_dahdi,$(EMBEDDED_MODS)),modules.link,chan_dahdi.so): sig_analog.o sig_pri.o
ifneq ($(filter chan_h323,$(EMBEDDED_MODS)),)
Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=247124&r1=247123&r2=247124
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Wed Feb 17 00:25:15 2010
@@ -263,9 +263,12 @@
#include "asterisk/stun.h"
#include "asterisk/cel.h"
#include "sip/include/sip.h"
+#include "sip/include/globals.h"
#include "sip/include/config_parser.h"
#include "sip/include/reqresp_parser.h"
#include "sip/include/sip_utils.h"
+#include "sip/include/dialog.h"
+#include "sip/include/dialplan_functions.h"
/*** DOCUMENTATION
<application name="SIPDtmfMode" language="en_US">
@@ -810,7 +813,7 @@
static int sip_reloading = FALSE; /*!< Flag for avoiding multiple reloads at the same time */
static enum channelreloadreason sip_reloadreason; /*!< Reason for last reload/load of configuration */
-static struct sched_context *sched; /*!< The scheduling context */
+struct sched_context *sched; /*!< The scheduling context */
static struct io_context *io; /*!< The IO context */
static int *sipsock_read_id; /*!< ID of IO entry for sipsock FD */
struct sip_pkt;
@@ -906,7 +909,7 @@
*/
static int sipsock = -1;
-static struct sockaddr_in bindaddr; /*!< UDP: The address we bind to */
+struct sockaddr_in bindaddr; /*!< UDP: The address we bind to */
/*! \brief our (internal) default address/port to put in SIP/SDP messages
* internip is initialized picking a suitable address from one of the
@@ -1028,23 +1031,11 @@
static void parse_moved_contact(struct sip_pvt *p, struct sip_request *req, char **name, char **number, int set_call_forward);
static int sip_send_mwi_to_peer(struct sip_peer *peer, const struct ast_event *event, int cache_only);
-/*--- Dialog management */
-static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *sin,
- int useglobal_nat, const int intended_method, struct sip_request *req);
+/* Misc dialog routines */
static int __sip_autodestruct(const void *data);
-static void sip_scheddestroy(struct sip_pvt *p, int ms);
-static int sip_cancel_destroy(struct sip_pvt *p);
-static struct sip_pvt *sip_destroy(struct sip_pvt *p);
-static void *dialog_unlink_all(struct sip_pvt *dialog, int lockowner, int lockdialoglist);
static void *registry_unref(struct sip_registry *reg, char *tag);
-static void __sip_destroy(struct sip_pvt *p, int lockowner, int lockdialoglist);
-static int __sip_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod);
-static void __sip_pretend_ack(struct sip_pvt *p);
-static int __sip_semi_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod);
+static int update_call_counter(struct sip_pvt *fup, int event);
static int auto_congest(const void *arg);
-static int update_call_counter(struct sip_pvt *fup, int event);
-static int hangup_sip2cause(int cause);
-static const char *hangup_cause2sip(int cause);
static struct sip_pvt *find_call(struct sip_request *req, struct sockaddr_in *sin, const int intended_method);
static void free_old_route(struct sip_route *route);
static void list_route(struct sip_route *route);
@@ -1172,7 +1163,6 @@
static int sip_addheader(struct ast_channel *chan, const char *data);
static int sip_do_reload(enum channelreloadreason reason);
static char *sip_reload(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
-static int acf_channel_read(struct ast_channel *chan, const char *funcname, char *preparse, char *buf, size_t buflen);
/*--- Debugging
Functions for enabling debug per IP or fully, or enabling history logging for
@@ -1345,7 +1335,7 @@
static int __sip_subscribe_mwi_do(struct sip_subscription_mwi *mwi);
/*! \brief Definition of this channel for PBX channel registration */
-static const struct ast_channel_tech sip_tech = {
+const struct ast_channel_tech sip_tech = {
.type = "SIP",
.description = "Session Initiation Protocol (SIP)",
.capabilities = AST_FORMAT_AUDIO_MASK, /* all audio formats */
@@ -1368,7 +1358,7 @@
.bridge = ast_rtp_instance_bridge, /* XXX chan unlocked ? */
.early_bridge = ast_rtp_instance_early_bridge,
.send_text = sip_sendtext, /* called with chan locked */
- .func_channel_read = acf_channel_read,
+ .func_channel_read = sip_acf_channel_read,
.setoption = sip_setoption,
.queryoption = sip_queryoption,
.get_pvt_uniqueid = sip_get_callid,
@@ -1380,7 +1370,7 @@
* The struct is initialized just before registering the channel driver,
* and is for use with channels using SIP INFO DTMF.
*/
-static struct ast_channel_tech sip_tech_info;
+struct ast_channel_tech sip_tech_info;
/*! \brief Working TLS connection configuration */
static struct ast_tls_config sip_tls_cfg;
@@ -1410,54 +1400,33 @@
.worker_fn = sip_tcp_worker_fn,
};
-/* wrapper macro to tell whether t points to one of the sip_tech descriptors */
-#define IS_SIP_TECH(t) ((t) == &sip_tech || (t) == &sip_tech_info)
-
/*! \brief Append to SIP dialog history
\return Always returns 0 */
#define append_history(p, event, fmt , args... ) append_history_full(p, "%-15s " fmt, event, ## args)
-/*! \brief
- * when we create or delete references, make sure to use these
- * functions so we keep track of the refcounts.
- * To simplify the code, we allow a NULL to be passed to dialog_unref().
- */
+struct sip_pvt *dialog_ref_debug(struct sip_pvt *p, char *tag, char *file, int line, const char *func)
+{
+ if (p)
#ifdef REF_DEBUG
-#define dialog_ref(arg1,arg2) dialog_ref_debug((arg1),(arg2), __FILE__, __LINE__, __PRETTY_FUNCTION__)
-#define dialog_unref(arg1,arg2) dialog_unref_debug((arg1),(arg2), __FILE__, __LINE__, __PRETTY_FUNCTION__)
-
-static struct sip_pvt *dialog_ref_debug(struct sip_pvt *p, char *tag, char *file, int line, const char *func)
-{
- if (p)
__ao2_ref_debug(p, 1, tag, file, line, func);
+#else
+ ao2_ref(p, 1);
+#endif
else
ast_log(LOG_ERROR, "Attempt to Ref a null pointer\n");
return p;
}
-static struct sip_pvt *dialog_unref_debug(struct sip_pvt *p, char *tag, char *file, int line, const char *func)
+struct sip_pvt *dialog_unref_debug(struct sip_pvt *p, char *tag, char *file, int line, const char *func)
{
if (p)
+#ifdef REF_DEBUG
__ao2_ref_debug(p, -1, tag, file, line, func);
+#else
+ ao2_ref(p, -1);
+#endif
return NULL;
}
-#else
-static struct sip_pvt *dialog_ref(struct sip_pvt *p, char *tag)
-{
- if (p)
- ao2_ref(p, 1);
- else
- ast_log(LOG_ERROR, "Attempt to Ref a null pointer\n");
- return p;
-}
-
-static struct sip_pvt *dialog_unref(struct sip_pvt *p, char *tag)
-{
- if (p)
- ao2_ref(p, -1);
- return NULL;
-}
-#endif
/*! \brief map from an integer value to a string.
* If no match is found, return errorstring
@@ -1941,7 +1910,7 @@
* \note A reference to the dialog must be held before calling this function, and this
* function does not release that reference.
*/
-static void *dialog_unlink_all(struct sip_pvt *dialog, int lockowner, int lockdialoglist)
+void *dialog_unlink_all(struct sip_pvt *dialog, int lockowner, int lockdialoglist)
{
struct sip_pkt *cp;
@@ -2006,7 +1975,7 @@
return NULL;
}
-static void *registry_unref(struct sip_registry *reg, char *tag)
+void *registry_unref(struct sip_registry *reg, char *tag)
{
ast_debug(3, "SIP Registry %s: refcount now %d\n", reg->hostname, reg->refcount - 1);
ASTOBJ_UNREF(reg, sip_registry_destroy);
@@ -2787,7 +2756,7 @@
}
/*! \brief Schedule destruction of SIP dialog */
-static void sip_scheddestroy(struct sip_pvt *p, int ms)
+void sip_scheddestroy(struct sip_pvt *p, int ms)
{
if (ms < 0) {
if (p->timer_t1 == 0) {
@@ -2815,7 +2784,7 @@
* Be careful as this also absorbs the reference - if you call it
* from within the scheduler, this might be the last reference.
*/
-static int sip_cancel_destroy(struct sip_pvt *p)
+int sip_cancel_destroy(struct sip_pvt *p)
{
int res = 0;
if (p->autokillid > -1) {
@@ -2832,7 +2801,7 @@
/*! \brief Acknowledges receipt of a packet and stops retransmission
* called with p locked*/
-static int __sip_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod)
+int __sip_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod)
{
struct sip_pkt *cur, *prev = NULL;
const char *msg = "Not Found"; /* used only for debugging */
@@ -2897,7 +2866,7 @@
/*! \brief Pretend to ack all packets
* called with p locked */
-static void __sip_pretend_ack(struct sip_pvt *p)
+void __sip_pretend_ack(struct sip_pvt *p)
{
struct sip_pkt *cur = NULL;
@@ -2914,7 +2883,7 @@
}
/*! \brief Acks receipt of packet, keep it around (used for provisional responses) */
-static int __sip_semi_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod)
+int __sip_semi_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod)
{
struct sip_pkt *cur;
int res = FALSE;
@@ -4282,7 +4251,7 @@
}
/*! \brief Execute destruction of SIP dialog structure, release memory */
-static void __sip_destroy(struct sip_pvt *p, int lockowner, int lockdialoglist)
+void __sip_destroy(struct sip_pvt *p, int lockowner, int lockdialoglist)
{
struct sip_request *req;
@@ -4415,7 +4384,7 @@
*
* \return 0 if call is ok (no call limit, below threshold)
* -1 on rejection of call
- *
+ *
*/
static int update_call_counter(struct sip_pvt *fup, int event)
{
@@ -4569,7 +4538,7 @@
* foo = sip_destroy(foo);
* and reduce the chance of bugs due to dangling pointers.
*/
-static struct sip_pvt * sip_destroy(struct sip_pvt *p)
+struct sip_pvt *sip_destroy(struct sip_pvt *p)
{
ast_debug(3, "Destroying SIP dialog %s\n", p->callid);
__sip_destroy(p, TRUE, TRUE);
@@ -4577,7 +4546,7 @@
}
/*! \brief Convert SIP hangup causes to Asterisk hangup causes */
-static int hangup_sip2cause(int cause)
+int hangup_sip2cause(int cause)
{
/* Possible values taken from causes.h */
@@ -4634,7 +4603,7 @@
return AST_CAUSE_FAILURE;
case 501: /* Call rejected */
return AST_CAUSE_FACILITY_REJECTED;
- case 502:
+ case 502:
return AST_CAUSE_DESTINATION_OUT_OF_ORDER;
case 503: /* Service unavailable */
return AST_CAUSE_CONGESTION;
@@ -4675,7 +4644,7 @@
In addition to these, a lot of PRI codes is defined in causes.h
...should we take care of them too ?
-
+
Quote RFC 3398
ISUP Cause value SIP response
@@ -4699,7 +4668,7 @@
31 normal unspecified 480 Temporarily unavailable
\endverbatim
*/
-static const char *hangup_cause2sip(int cause)
+const char *hangup_cause2sip(int cause)
{
switch (cause) {
case AST_CAUSE_UNALLOCATED: /* 1 */
@@ -5958,7 +5927,7 @@
* Returns a reference to the object so whoever uses it later must
* remember to release the reference.
*/
-static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *sin,
+struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *sin,
int useglobal_nat, const int intended_method, struct sip_request *req)
{
struct sip_pvt *p;
@@ -20185,122 +20154,6 @@
transmit_response(p, "481 Call Leg Does Not Exist", req);
return 0;
}
-}
-
-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 *parse = ast_strdupa(preparse);
- int res = 0;
- AST_DECLARE_APP_ARGS(args,
- AST_APP_ARG(param);
- AST_APP_ARG(type);
- AST_APP_ARG(field);
- );
- AST_STANDARD_APP_ARGS(args, parse);
-
- /* Sanity check */
- if (!IS_SIP_TECH(chan->tech)) {
- ast_log(LOG_ERROR, "Cannot call %s on a non-SIP channel\n", funcname);
- return 0;
- }
-
- memset(buf, 0, buflen);
-
- if (p == NULL) {
- return -1;
- }
-
- if (!strcasecmp(args.param, "peerip")) {
- ast_copy_string(buf, p->sa.sin_addr.s_addr ? ast_inet_ntoa(p->sa.sin_addr) : "", buflen);
- } else if (!strcasecmp(args.param, "recvip")) {
- ast_copy_string(buf, p->recv.sin_addr.s_addr ? ast_inet_ntoa(p->recv.sin_addr) : "", buflen);
- } else if (!strcasecmp(args.param, "from")) {
- ast_copy_string(buf, p->from, buflen);
- } else if (!strcasecmp(args.param, "uri")) {
- ast_copy_string(buf, p->uri, buflen);
- } else if (!strcasecmp(args.param, "useragent")) {
- ast_copy_string(buf, p->useragent, buflen);
- } else if (!strcasecmp(args.param, "peername")) {
- ast_copy_string(buf, p->peername, buflen);
- } else if (!strcasecmp(args.param, "t38passthrough")) {
- ast_copy_string(buf, (p->t38.state == T38_DISABLED) ? "0" : "1", buflen);
- } else if (!strcasecmp(args.param, "rtpdest")) {
- struct sockaddr_in sin;
-
- if (ast_strlen_zero(args.type))
- args.type = "audio";
-
- if (!strcasecmp(args.type, "audio"))
- ast_rtp_instance_get_remote_address(p->rtp, &sin);
- else if (!strcasecmp(args.type, "video"))
- ast_rtp_instance_get_remote_address(p->vrtp, &sin);
- else if (!strcasecmp(args.type, "text"))
- ast_rtp_instance_get_remote_address(p->trtp, &sin);
- else
- return -1;
-
- snprintf(buf, buflen, "%s:%d", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
- } else if (!strcasecmp(args.param, "rtpqos")) {
- struct ast_rtp_instance *rtp = NULL;
-
- if (ast_strlen_zero(args.type)) {
- args.type = "audio";
- }
-
- 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 (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;
- }
- return res;
}
/*! \brief Handle incoming BYE request */
@@ -25159,6 +25012,7 @@
{
sip_config_parser_register_tests();
sip_request_parser_register_tests();
+ sip_dialplan_function_register_tests();
}
/*! \brief SIP test registration */
@@ -25166,6 +25020,7 @@
{
sip_config_parser_unregister_tests();
sip_request_parser_unregister_tests();
+ sip_dialplan_function_unregister_tests();
}
/*! \brief PBX load module - initialization */
Added: trunk/channels/sip/dialplan_functions.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/sip/dialplan_functions.c?view=auto&rev=247124
==============================================================================
--- trunk/channels/sip/dialplan_functions.c (added)
+++ trunk/channels/sip/dialplan_functions.c Wed Feb 17 00:25:15 2010
@@ -1,0 +1,366 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2010, Digium, Inc.
+ *
+ * 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
+ * \brief sip channel dialplan functions and unit tests
+ */
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include <math.h>
+
+#include "asterisk/channel.h"
+#include "asterisk/rtp_engine.h"
+#include "asterisk/pbx.h"
+
+#include "include/sip.h"
+#include "include/globals.h"
+#include "include/dialog.h"
+#include "include/dialplan_functions.h"
+#include "include/sip_utils.h"
+
+
+int sip_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 *parse = ast_strdupa(preparse);
+ int res = 0;
+ AST_DECLARE_APP_ARGS(args,
+ AST_APP_ARG(param);
+ AST_APP_ARG(type);
+ AST_APP_ARG(field);
+ );
+ AST_STANDARD_APP_ARGS(args, parse);
+
+ /* Sanity check */
+ if (!IS_SIP_TECH(chan->tech)) {
+ ast_log(LOG_ERROR, "Cannot call %s on a non-SIP channel\n", funcname);
+ return 0;
+ }
+
+ memset(buf, 0, buflen);
+
+ if (p == NULL) {
+ return -1;
+ }
+
+ if (!strcasecmp(args.param, "peerip")) {
+ ast_copy_string(buf, p->sa.sin_addr.s_addr ? ast_inet_ntoa(p->sa.sin_addr) : "", buflen);
+ } else if (!strcasecmp(args.param, "recvip")) {
+ ast_copy_string(buf, p->recv.sin_addr.s_addr ? ast_inet_ntoa(p->recv.sin_addr) : "", buflen);
+ } else if (!strcasecmp(args.param, "from")) {
+ ast_copy_string(buf, p->from, buflen);
+ } else if (!strcasecmp(args.param, "uri")) {
+ ast_copy_string(buf, p->uri, buflen);
+ } else if (!strcasecmp(args.param, "useragent")) {
+ ast_copy_string(buf, p->useragent, buflen);
+ } else if (!strcasecmp(args.param, "peername")) {
+ ast_copy_string(buf, p->peername, buflen);
+ } else if (!strcasecmp(args.param, "t38passthrough")) {
+ ast_copy_string(buf, (p->t38.state == T38_DISABLED) ? "0" : "1", buflen);
+ } else if (!strcasecmp(args.param, "rtpdest")) {
+ struct sockaddr_in sin;
+
+ if (ast_strlen_zero(args.type))
+ args.type = "audio";
+
+ if (!strcasecmp(args.type, "audio"))
+ ast_rtp_instance_get_remote_address(p->rtp, &sin);
+ else if (!strcasecmp(args.type, "video"))
+ ast_rtp_instance_get_remote_address(p->vrtp, &sin);
+ else if (!strcasecmp(args.type, "text"))
+ ast_rtp_instance_get_remote_address(p->trtp, &sin);
+ else
+ return -1;
+
+ snprintf(buf, buflen, "%s:%d", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
+ } else if (!strcasecmp(args.param, "rtpqos")) {
+ struct ast_rtp_instance *rtp = NULL;
+
+ if (ast_strlen_zero(args.type)) {
+ args.type = "audio";
+ }
+
+ 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 (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;
+ int i;
+ struct {
+ const char *name;
+ enum { INT, DBL } type;
+ union {
+ unsigned int *i4;
+ double *d8;
+ };
+ } lookup[] = {
+ { "txcount", INT, { .i4 = &stats.txcount, }, },
+ { "rxcount", INT, { .i4 = &stats.rxcount, }, },
+ { "txjitter", INT, { .i4 = &stats.txjitter, }, },
+ { "rxjitter", INT, { .i4 = &stats.rxjitter, }, },
+ { "remote_maxjitter", DBL, { .d8 = &stats.remote_maxjitter, }, },
+ { "remote_minjitter", DBL, { .d8 = &stats.remote_minjitter, }, },
+ { "remote_normdevjitter", DBL, { .d8 = &stats.remote_normdevjitter, }, },
+ { "remote_stdevjitter", DBL, { .d8 = &stats.remote_stdevjitter, }, },
+ { "local_maxjitter", DBL, { .d8 = &stats.local_maxjitter, }, },
+ { "local_minjitter", DBL, { .d8 = &stats.local_minjitter, }, },
+ { "local_normdevjitter", DBL, { .d8 = &stats.local_normdevjitter, }, },
+ { "local_stdevjitter", DBL, { .d8 = &stats.local_stdevjitter, }, },
+ { "txploss", INT, { .i4 = &stats.txploss, }, },
+ { "rxploss", INT, { .i4 = &stats.rxploss, }, },
+ { "remote_maxrxploss", DBL, { .d8 = &stats.remote_maxrxploss, }, },
+ { "remote_minrxploss", DBL, { .d8 = &stats.remote_minrxploss, }, },
+ { "remote_normdevrxploss", DBL, { .d8 = &stats.remote_normdevrxploss, }, },
+ { "remote_stdevrxploss", DBL, { .d8 = &stats.remote_stdevrxploss, }, },
+ { "local_maxrxploss", DBL, { .d8 = &stats.local_maxrxploss, }, },
+ { "local_minrxploss", DBL, { .d8 = &stats.local_minrxploss, }, },
+ { "local_normdevrxploss", DBL, { .d8 = &stats.local_normdevrxploss, }, },
+ { "local_stdevrxploss", DBL, { .d8 = &stats.local_stdevrxploss, }, },
+ { "rtt", INT, { .i4 = &stats.rtt, }, },
+ { "maxrtt", DBL, { .d8 = &stats.maxrtt, }, },
+ { "minrtt", DBL, { .d8 = &stats.minrtt, }, },
+ { "normdevrtt", DBL, { .d8 = &stats.normdevrtt, }, },
+ { "stdevrtt", DBL, { .d8 = &stats.stdevrtt, }, },
+ { "local_ssrc", INT, { .i4 = &stats.local_ssrc, }, },
+ { "remote_ssrc", INT, { .i4 = &stats.remote_ssrc, }, },
+ { NULL, },
+ };
+
+ if (ast_rtp_instance_get_stats(rtp, &stats, AST_RTP_INSTANCE_STAT_ALL)) {
+ return -1;
+ }
+
+ for (i = 0; !ast_strlen_zero(lookup[i].name); i++) {
+ if (!strcasecmp(args.field, lookup[i].name)) {
+ if (lookup[i].type == INT) {
+ snprintf(buf, buflen, "%u", *lookup[i].i4);
+ } else {
+ snprintf(buf, buflen, "%f", *lookup[i].d8);
+ }
+ return 0;
+ }
+ }
+ ast_log(LOG_WARNING, "Unrecognized argument '%s' to %s\n", preparse, funcname);
+ return -1;
+ }
+ } else {
+ res = -1;
+ }
+ return res;
+}
+
+#ifdef TEST_FRAMEWORK
+static int test_sip_rtpqos_1_new(struct ast_rtp_instance *instance, struct sched_context *sched, struct sockaddr_in *sin, void *data)
+{
+ /* Needed to pass sanity checks */
+ ast_rtp_instance_set_data(instance, data);
+ return 0;
+}
+
+static int test_sip_rtpqos_1_destroy(struct ast_rtp_instance *instance)
+{
+ /* Needed to pass sanity checks */
+ return 0;
+}
+
+static struct ast_frame *test_sip_rtpqos_1_read(struct ast_rtp_instance *instance, int rtcp)
+{
+ /* Needed to pass sanity checks */
+ return &ast_null_frame;
+}
+
+static int test_sip_rtpqos_1_write(struct ast_rtp_instance *instance, struct ast_frame *frame)
+{
+ /* Needed to pass sanity checks */
+ return 0;
+}
+
+static int test_sip_rtpqos_1_get_stat(struct ast_rtp_instance *instance, struct ast_rtp_instance_stats *stats, enum ast_rtp_instance_stat stat)
+{
+ struct ast_rtp_instance_stats *s = ast_rtp_instance_get_data(instance);
+ memcpy(stats, s, sizeof(*stats));
+ return 0;
+}
+
+AST_TEST_DEFINE(test_sip_rtpqos_1)
+{
+ int i, res = AST_TEST_PASS;
+ struct ast_rtp_engine test_engine = {
+ .name = "test",
+ .new = test_sip_rtpqos_1_new,
+ .destroy = test_sip_rtpqos_1_destroy,
+ .read = test_sip_rtpqos_1_read,
+ .write = test_sip_rtpqos_1_write,
+ .get_stat = test_sip_rtpqos_1_get_stat,
+ };
+ struct sockaddr_in sin = { .sin_port = 31337, .sin_addr = { 4 * 16777216 + 3 * 65536 + 2 * 256 + 1 } };
+ struct ast_rtp_instance_stats mine = { 0, };
+ struct sip_pvt *p = NULL;
+ struct ast_channel *chan = NULL;
+ struct ast_str *varstr = NULL, *buffer = NULL;
+ struct {
+ const char *name;
+ enum { INT, DBL } type;
+ union {
+ unsigned int *i4;
+ double *d8;
+ };
+ } lookup[] = {
+ { "txcount", INT, { .i4 = &mine.txcount, }, },
+ { "rxcount", INT, { .i4 = &mine.rxcount, }, },
+ { "txjitter", INT, { .i4 = &mine.txjitter, }, },
+ { "rxjitter", INT, { .i4 = &mine.rxjitter, }, },
+ { "remote_maxjitter", DBL, { .d8 = &mine.remote_maxjitter, }, },
+ { "remote_minjitter", DBL, { .d8 = &mine.remote_minjitter, }, },
+ { "remote_normdevjitter", DBL, { .d8 = &mine.remote_normdevjitter, }, },
+ { "remote_stdevjitter", DBL, { .d8 = &mine.remote_stdevjitter, }, },
+ { "local_maxjitter", DBL, { .d8 = &mine.local_maxjitter, }, },
+ { "local_minjitter", DBL, { .d8 = &mine.local_minjitter, }, },
+ { "local_normdevjitter", DBL, { .d8 = &mine.local_normdevjitter, }, },
+ { "local_stdevjitter", DBL, { .d8 = &mine.local_stdevjitter, }, },
+ { "txploss", INT, { .i4 = &mine.txploss, }, },
+ { "rxploss", INT, { .i4 = &mine.rxploss, }, },
+ { "remote_maxrxploss", DBL, { .d8 = &mine.remote_maxrxploss, }, },
+ { "remote_minrxploss", DBL, { .d8 = &mine.remote_minrxploss, }, },
+ { "remote_normdevrxploss", DBL, { .d8 = &mine.remote_normdevrxploss, }, },
+ { "remote_stdevrxploss", DBL, { .d8 = &mine.remote_stdevrxploss, }, },
+ { "local_maxrxploss", DBL, { .d8 = &mine.local_maxrxploss, }, },
+ { "local_minrxploss", DBL, { .d8 = &mine.local_minrxploss, }, },
+ { "local_normdevrxploss", DBL, { .d8 = &mine.local_normdevrxploss, }, },
+ { "local_stdevrxploss", DBL, { .d8 = &mine.local_stdevrxploss, }, },
+ { "rtt", INT, { .i4 = &mine.rtt, }, },
+ { "maxrtt", DBL, { .d8 = &mine.maxrtt, }, },
+ { "minrtt", DBL, { .d8 = &mine.minrtt, }, },
+ { "normdevrtt", DBL, { .d8 = &mine.normdevrtt, }, },
+ { "stdevrtt", DBL, { .d8 = &mine.stdevrtt, }, },
+ { "local_ssrc", INT, { .i4 = &mine.local_ssrc, }, },
+ { "remote_ssrc", INT, { .i4 = &mine.remote_ssrc, }, },
+ { NULL, },
+ };
+
+ switch (cmd) {
+ case TEST_INIT:
+ info->name = "test_sip_rtpqos";
+ info->category = "channels/chan_sip/";
+ info->summary = "Test retrieval of SIP RTP QOS stats";
+ info->description =
+ "Verify values in the RTP instance structure can be accessed through the dialplan.";
+ return AST_TEST_NOT_RUN;
+ case TEST_EXECUTE:
+ break;
+ }
+
+ ast_rtp_engine_register2(&test_engine, NULL);
+ /* Have to associate this with a SIP pvt and an ast_channel */
+ if (!(p = sip_alloc(NULL, NULL, 0, SIP_NOTIFY, NULL))) {
+ res = AST_TEST_NOT_RUN;
+ goto done;
+ }
+ if (!(p->rtp = ast_rtp_instance_new("test", sched, &bindaddr, &mine))) {
+ res = AST_TEST_NOT_RUN;
+ goto done;
+ }
+ ast_rtp_instance_set_remote_address(p->rtp, &sin);
+ if (!(chan = ast_dummy_channel_alloc())) {
+ res = AST_TEST_NOT_RUN;
+ goto done;
+ }
+ chan->tech = &sip_tech;
+ chan->tech_pvt = p;
+ p->owner = chan;
+
+ varstr = ast_str_create(16);
+ buffer = ast_str_create(16);
+ if (!varstr || !buffer) {
+ res = AST_TEST_NOT_RUN;
+ goto done;
+ }
+
+ /* Populate "mine" with values, then retrieve them with the CHANNEL dialplan function */
+ for (i = 0; !ast_strlen_zero(lookup[i].name); i++) {
+ ast_str_set(&varstr, 0, "${CHANNEL(rtpqos,audio,%s)}", lookup[i].name);
+ if (lookup[i].type == INT) {
+ int j;
+ char cmpstr[256];
+ for (j = 1; j < 25; j++) {
+ *lookup[i].i4 = j;
+ ast_str_substitute_variables(&buffer, 0, chan, ast_str_buffer(varstr));
+ snprintf(cmpstr, sizeof(cmpstr), "%d", j);
+ if (strcmp(cmpstr, ast_str_buffer(buffer))) {
+ res = AST_TEST_FAIL;
+ ast_test_status_update(test, "%s != %s != %s\n", ast_str_buffer(varstr), cmpstr, ast_str_buffer(buffer));
+ break;
+ }
+ }
+ } else {
+ double j, cmpdbl = 0.0;
+ for (j = 1.0; j < 10.0; j += 0.3) {
+ *lookup[i].d8 = j;
+ ast_str_substitute_variables(&buffer, 0, chan, ast_str_buffer(varstr));
+ if (sscanf(ast_str_buffer(buffer), "%lf", &cmpdbl) != 1 || fabs(j - cmpdbl > .05)) {
+ res = AST_TEST_FAIL;
+ ast_test_status_update(test, "%s != %f != %s\n", ast_str_buffer(varstr), j, ast_str_buffer(buffer));
+ break;
+ }
+ }
+ }
+ }
+
+done:
+ ast_free(varstr);
+ ast_free(buffer);
+
+ /* This unref will take care of destroying the channel, RTP instance, and SIP pvt */
+ if (p) {
+ dialog_unref(p, "Destroy test object");
+ }
+ ast_rtp_engine_unregister(&test_engine);
+ return res;
+}
+#endif
+
+/*! \brief SIP test registration */
+void sip_dialplan_function_register_tests(void)
+{
+ AST_TEST_REGISTER(test_sip_rtpqos_1);
+}
+
+/*! \brief SIP test registration */
+void sip_dialplan_function_unregister_tests(void)
+{
+ AST_TEST_UNREGISTER(test_sip_rtpqos_1);
+}
+
Propchange: trunk/channels/sip/dialplan_functions.c
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svn:eol-style = native
Propchange: trunk/channels/sip/dialplan_functions.c
------------------------------------------------------------------------------
svn:keywords = Date Author Id Revision Yttribium
Propchange: trunk/channels/sip/dialplan_functions.c
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svn:mime-type = text/plain
Modified: trunk/channels/sip/include/config_parser.h
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/sip/include/config_parser.h?view=diff&rev=247124&r1=247123&r2=247124
==============================================================================
--- trunk/channels/sip/include/config_parser.h (original)
+++ trunk/channels/sip/include/config_parser.h Wed Feb 17 00:25:15 2010
@@ -43,7 +43,7 @@
*/
int sip_parse_host(char *line, int lineno, char **hostname, int *portnum, enum sip_transport *transport);
-/*!
+/*!
* \brief register config parsing tests
*/
void sip_config_parser_register_tests(void);
Added: trunk/channels/sip/include/dialog.h
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/sip/include/dialog.h?view=auto&rev=247124
==============================================================================
--- trunk/channels/sip/include/dialog.h (added)
+++ trunk/channels/sip/include/dialog.h Wed Feb 17 00:25:15 2010
@@ -1,0 +1,75 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2010, Digium, Inc.
+ *
+ * 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
+ * \brief sip dialog management header file
+ */
+
+#include "sip.h"
+
+#ifndef _SIP_DIALOG_H
+#define _SIP_DIALOG_H
+
+/*! \brief
+ * when we create or delete references, make sure to use these
+ * functions so we keep track of the refcounts.
+ * To simplify the code, we allow a NULL to be passed to dialog_unref().
+ */
+#define dialog_ref(arg1,arg2) dialog_ref_debug((arg1),(arg2), __FILE__, __LINE__, __PRETTY_FUNCTION__)
+#define dialog_unref(arg1,arg2) dialog_unref_debug((arg1),(arg2), __FILE__, __LINE__, __PRETTY_FUNCTION__)
+struct sip_pvt *dialog_ref_debug(struct sip_pvt *p, char *tag, char *file, int line, const char *func);
+struct sip_pvt *dialog_unref_debug(struct sip_pvt *p, char *tag, char *file, int line, const char *func);
+
+struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *sin,
+ int useglobal_nat, const int intended_method, struct sip_request *req);
+void sip_scheddestroy(struct sip_pvt *p, int ms);
+int sip_cancel_destroy(struct sip_pvt *p);
+
+/*! \brief Destroy SIP call structure.
+ * Make it return NULL so the caller can do things like
+ * foo = sip_destroy(foo);
+ * and reduce the chance of bugs due to dangling pointers.
+ */
+struct sip_pvt *sip_destroy(struct sip_pvt *p);
+
+/*! \brief Destroy SIP call structure.
+ * Make it return NULL so the caller can do things like
+ * foo = sip_destroy(foo);
+ * and reduce the chance of bugs due to dangling pointers.
+ */
+void __sip_destroy(struct sip_pvt *p, int lockowner, int lockdialoglist);
+/*!
+ * \brief Unlink a dialog from the dialogs container, as well as any other places
+ * that it may be currently stored.
+ *
+ * \note A reference to the dialog must be held before calling this function, and this
+ * function does not release that reference.
+ */
+void *dialog_unlink_all(struct sip_pvt *dialog, int lockowner, int lockdialoglist);
+
+/*! \brief Acknowledges receipt of a packet and stops retransmission
+ * called with p locked*/
+int __sip_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod);
+
+/*! \brief Pretend to ack all packets
+ * called with p locked */
+void __sip_pretend_ack(struct sip_pvt *p);
+
+/*! \brief Acks receipt of packet, keep it around (used for provisional responses) */
+int __sip_semi_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod);
+
+#endif /* defined(_SIP_DIALOG_H) */
Propchange: trunk/channels/sip/include/dialog.h
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svn:eol-style = native
Propchange: trunk/channels/sip/include/dialog.h
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svn:keywords = Date Author Id Revision Yttribium
Propchange: trunk/channels/sip/include/dialog.h
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: trunk/channels/sip/include/dialplan_functions.h
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/sip/include/dialplan_functions.h?view=auto&rev=247124
==============================================================================
--- trunk/channels/sip/include/dialplan_functions.h (added)
+++ trunk/channels/sip/include/dialplan_functions.h Wed Feb 17 00:25:15 2010
@@ -1,0 +1,41 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2010, Digium, Inc.
+ *
+ * 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
+ * \brief SIP dialplan functions header file
+ */
+
+#include "sip.h"
+
+#ifndef _SIP_DIALPLAN_FUNCTIONS_H
+#define _SIP_DIALPLAN_FUNCTIONS_H
+
+/*!
+ * \brief Channel read dialplan function for SIP
+ */
+int sip_acf_channel_read(struct ast_channel *chan, const char *funcname, char *preparse, char *buf, size_t buflen);
+
+/*!
+ * \brief register dialplan function tests
+ */
+void sip_dialplan_function_register_tests(void);
+/*!
+ * \brief unregister dialplan function tests
+ */
+void sip_dialplan_function_unregister_tests(void);
+
+#endif /* !defined(_SIP_DIALPLAN_FUNCTIONS_H) */
Propchange: trunk/channels/sip/include/dialplan_functions.h
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: trunk/channels/sip/include/dialplan_functions.h
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svn:keywords = Date Author Id Revision Yttribium
Propchange: trunk/channels/sip/include/dialplan_functions.h
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svn:mime-type = text/plain
Added: trunk/channels/sip/include/globals.h
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/sip/include/globals.h?view=auto&rev=247124
==============================================================================
--- trunk/channels/sip/include/globals.h (added)
+++ trunk/channels/sip/include/globals.h Wed Feb 17 00:25:15 2010
@@ -1,0 +1,42 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2010, Digium, Inc.
+ *
+ * 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
+ * \brief sip global declaration header file
+ */
+
+#include "sip.h"
+
+#ifndef _SIP_GLOBALS_H
+#define _SIP_GLOBALS_H
+
+extern struct sockaddr_in bindaddr; /*!< UDP: The address we bind to */
+extern struct sched_context *sched; /*!< The scheduling context */
+
+/*! \brief Definition of this channel for PBX channel registration */
+extern const struct ast_channel_tech sip_tech;
+
+/*! \brief This version of the sip channel tech has no send_digit_begin
+ * callback so that the core knows that the channel does not want
+ * DTMF BEGIN frames.
+ * The struct is initialized just before registering the channel driver,
+ * and is for use with channels using SIP INFO DTMF.
+ */
+extern struct ast_channel_tech sip_tech_info;
+
+#endif /* !defined(SIP_GLOBALS_H) */
+
Propchange: trunk/channels/sip/include/globals.h
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: trunk/channels/sip/include/globals.h
------------------------------------------------------------------------------
svn:keywords = Date Author Id Revision Yttribium
Propchange: trunk/channels/sip/include/globals.h
------------------------------------------------------------------------------
svn:mime-type = text/plain
Modified: trunk/channels/sip/include/sip_utils.h
[... 61 lines stripped ...]
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