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/*
* 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
*/
<support_level>extended</support_level>
<info name="CHANNEL" language="en_US" tech="SIP">
<enumlist>
<enum name="peerip">
<para>R/O Get the IP address of the peer.</para>
</enum>
<enum name="recvip">
<para>R/O Get the source IP address of the peer.</para>
</enum>
<enum name="recvport">
<para>R/O Get the source port of the peer.</para>
</enum>
<enum name="from">
<para>R/O Get the URI from the From: header.</para>
</enum>
<enum name="uri">
<para>R/O Get the URI from the Contact: header.</para>
</enum>
<enum name="ruri">
<para>R/O Get the Request-URI from the INVITE header.</para>
</enum>
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<enum name="useragent">
<para>R/O Get the useragent.</para>
</enum>
<enum name="peername">
<para>R/O Get the name of the peer.</para>
</enum>
<enum name="t38passthrough">
<para>R/O <literal>1</literal> if T38 is offered or enabled in this channel,
otherwise <literal>0</literal></para>
</enum>
<enum name="rtpqos">
<para>R/O Get QOS information about the RTP stream</para>
<para> This option takes two additional arguments:</para>
<para> Argument 1:</para>
<para> <literal>audio</literal> Get data about the audio stream</para>
<para> <literal>video</literal> Get data about the video stream</para>
<para> <literal>text</literal> Get data about the text stream</para>
<para> Argument 2:</para>
<para> <literal>local_ssrc</literal> Local SSRC (stream ID)</para>
<para> <literal>local_lostpackets</literal> Local lost packets</para>
<para> <literal>local_jitter</literal> Local calculated jitter</para>
<para> <literal>local_maxjitter</literal> Local calculated jitter (maximum)</para>
<para> <literal>local_minjitter</literal> Local calculated jitter (minimum)</para>
<para> <literal>local_normdevjitter</literal>Local calculated jitter (normal deviation)</para>
<para> <literal>local_stdevjitter</literal> Local calculated jitter (standard deviation)</para>
<para> <literal>local_count</literal> Number of received packets</para>
<para> <literal>remote_ssrc</literal> Remote SSRC (stream ID)</para>
<para> <literal>remote_lostpackets</literal>Remote lost packets</para>
<para> <literal>remote_jitter</literal> Remote reported jitter</para>
<para> <literal>remote_maxjitter</literal> Remote calculated jitter (maximum)</para>
<para> <literal>remote_minjitter</literal> Remote calculated jitter (minimum)</para>
<para> <literal>remote_normdevjitter</literal>Remote calculated jitter (normal deviation)</para>
<para> <literal>remote_stdevjitter</literal>Remote calculated jitter (standard deviation)</para>
<para> <literal>remote_count</literal> Number of transmitted packets</para>
<para> <literal>rtt</literal> Round trip time</para>
<para> <literal>maxrtt</literal> Round trip time (maximum)</para>
<para> <literal>minrtt</literal> Round trip time (minimum)</para>
<para> <literal>normdevrtt</literal> Round trip time (normal deviation)</para>
<para> <literal>stdevrtt</literal> Round trip time (standard deviation)</para>
<para> <literal>all</literal> All statistics (in a form suited to logging,
but not for parsing)</para>
</enum>
<enum name="rtpdest">
<para>R/O Get remote RTP destination information.</para>
<para> This option takes one additional argument:</para>
<para> Argument 1:</para>
<para> <literal>audio</literal> Get audio destination</para>
<para> <literal>video</literal> Get video destination</para>
<para> <literal>text</literal> Get text destination</para>
<para> Defaults to <literal>audio</literal> if unspecified.</para>
</enum>
<enum name="rtpsource">
<para>R/O Get source RTP destination information.</para>
<para> This option takes one additional argument:</para>
<para> Argument 1:</para>
<para> <literal>audio</literal> Get audio destination</para>
<para> <literal>video</literal> Get video destination</para>
<para> <literal>text</literal> Get text destination</para>
<para> Defaults to <literal>audio</literal> if unspecified.</para>
</enum>
</enumlist>
</info>
***/
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#include "asterisk.h"
#include <math.h>
#include "asterisk/channel.h"
#include "asterisk/rtp_engine.h"
#include "asterisk/pbx.h"
#include "asterisk/acl.h"
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#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 = ast_channel_tech_pvt(chan);
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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);
);
/* Check for zero arguments */
if (ast_strlen_zero(parse)) {
ast_log(LOG_ERROR, "Cannot call %s without arguments\n", funcname);
return -1;
}
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AST_STANDARD_APP_ARGS(args, parse);
/* Sanity check */
if (!IS_SIP_TECH(ast_channel_tech(chan))) {
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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, ast_sockaddr_isnull(&p->sa) ? "" : ast_sockaddr_stringify_addr(&p->sa), buflen);
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} else if (!strcasecmp(args.param, "recvip")) {
ast_copy_string(buf, ast_sockaddr_isnull(&p->recv) ? "" : ast_sockaddr_stringify_addr(&p->recv), buflen);
} else if (!strcasecmp(args.param, "recvport")) {
ast_copy_string(buf, ast_sockaddr_isnull(&p->recv) ? "" : ast_sockaddr_stringify_port(&p->recv), buflen);
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} 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, "ruri")) {
char *tmpruri = REQ_OFFSET_TO_STR(&p->initreq, rlpart2);
ast_copy_string(buf, tmpruri, buflen);
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} 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 ast_rtp_instance *stream;
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if (ast_strlen_zero(args.type))
args.type = "audio";
if (!strcasecmp(args.type, "audio"))
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else if (!strcasecmp(args.type, "video"))
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else if (!strcasecmp(args.type, "text"))
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else
return -1;
/* Return 0 to suppress a console warning message */
return 0;
ast_rtp_instance_get_remote_address(stream, &addr);
snprintf(buf, buflen, "%s", ast_sockaddr_stringify(&addr));
} else if (!strcasecmp(args.param, "rtpsource")) {
struct ast_rtp_instance *stream;
if (ast_strlen_zero(args.type))
args.type = "audio";
if (!strcasecmp(args.type, "audio"))
stream = p->rtp;
else if (!strcasecmp(args.type, "video"))
stream = p->vrtp;
else if (!strcasecmp(args.type, "text"))
stream = p->trtp;
else
return -1;
/* Return 0 to suppress a console warning message */
return 0;
if (ast_sockaddr_isnull(&sa)) {
struct ast_sockaddr dest_sa;
ast_rtp_instance_get_remote_address(stream, &dest_sa);
ast_ouraddrfor(&dest_sa, &sa);
snprintf(buf, buflen, "%s", ast_sockaddr_stringify(&sa));
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} 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];
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if (!ast_rtp_instance_get_quality(rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf))) {
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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, }, },
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{ "txjitter", DBL, { .d8 = &stats.txjitter, }, },
{ "rxjitter", DBL, { .d8 = &stats.rxjitter, }, },
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{ "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, }, },
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{ "rtt", DBL, { .d8 = &stats.rtt, }, },
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{ "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 if (!strcasecmp(args.param, "secure_signaling")) {
snprintf(buf, buflen, "%s", p->socket.type == AST_TRANSPORT_TLS ? "1" : "");
} else if (!strcasecmp(args.param, "secure_media")) {
snprintf(buf, buflen, "%s", p->srtp ? "1" : "");
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} else {
res = -1;
}
return res;
}
#ifdef TEST_FRAMEWORK
static int test_sip_rtpqos_1_new(struct ast_rtp_instance *instance, struct ast_sched_context *sched, struct ast_sockaddr *addr, void *data)
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{
/* 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;
static struct ast_rtp_engine test_engine = {
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.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,
};
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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", DBL, { .d8 = &mine.txjitter, }, },
{ "rxjitter", DBL, { .d8 = &mine.rxjitter, }, },
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{ "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, }, },
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{ "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";
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info->category = "/channels/chan_sip/";
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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_register(&test_engine);
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/* Have to associate this with a SIP pvt and an ast_channel */
if (!(p = sip_alloc(0, NULL, 0, SIP_NOTIFY, NULL, 0))) {
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res = AST_TEST_NOT_RUN;
goto done;
}
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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, &sa);
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if (!(chan = ast_dummy_channel_alloc())) {
res = AST_TEST_NOT_RUN;
goto done;
}
ast_channel_tech_set(chan, &sip_tech);
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ast_channel_tech_pvt_set(chan, dialog_ref(p, "Give the owner channel a reference to the dialog"));
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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) {
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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);
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/* This unlink and unref will take care of destroying the channel, RTP instance, and SIP pvt */
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if (p) {
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dialog_unlink_all(p);
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dialog_unref(p, "Destroy test object");
}
if (chan) {
ast_channel_unref(chan);
}
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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);
}