asterisk/channels/sip/dialplan_functions.c

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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
*/
/*** MODULEINFO
<support_level>extended</support_level>
***/
/*** DOCUMENTATION
<info name="SIPCHANNEL" 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="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>
***/
#include "asterisk.h"
git migration: Refactor the ASTERISK_FILE_VERSION macro Git does not support the ability to replace a token with a version string during check-in. While it does have support for replacing a token on clone, this is somewhat sub-optimal: the token is replaced with the object hash, which is not particularly easy for human consumption. What's more, in practice, the source file version was often not terribly useful. Generally, when triaging bugs, the overall version of Asterisk is far more useful than an individual SVN version of a file. As a result, this patch removes Asterisk's support for showing source file versions. Specifically, it does the following: * Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and remove passing the version in with the macro. Other facilities than 'core show file version' make use of the file names, such as setting a debug level only on a specific file. As such, the act of registering source files with the Asterisk core still has use. The macro rename now reflects the new macro purpose. * main/asterisk: - Refactor the file_version structure to reflect that it no longer tracks a version field. - Remove the "core show file version" CLI command. Without the file version, it is no longer useful. - Remove the ast_file_version_find function. The file version is no longer tracked. - Rename ast_register_file_version/ast_unregister_file_version to ast_register_file/ast_unregister_file, respectively. * main/manager: Remove value from the Version key of the ModuleCheck Action. The actual key itself has not been removed, as doing so would absolutely constitute a backwards incompatible change. However, since the file version is no longer tracked, there is no need to attempt to include it in the Version key. * UPGRADE: Add notes for: - Modification to the ModuleCheck AMI Action - Removal of the "core show file version" CLI command Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e
2015-04-12 02:38:22 +00:00
ASTERISK_REGISTER_FILE()
#include <math.h>
#include "asterisk/channel.h"
#include "asterisk/rtp_engine.h"
#include "asterisk/pbx.h"
#include "asterisk/acl.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 = ast_channel_tech_pvt(chan);
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;
}
AST_STANDARD_APP_ARGS(args, parse);
/* Sanity check */
if (!IS_SIP_TECH(ast_channel_tech(chan))) {
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);
} 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);
} 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 ast_sockaddr addr;
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 */
if (!stream) {
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_sockaddr sa;
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 */
if (!stream) {
return 0;
}
ast_rtp_instance_get_local_address(stream, &sa);
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));
} 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];
if (!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", DBL, { .d8 = &stats.txjitter, }, },
{ "rxjitter", DBL, { .d8 = &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", DBL, { .d8 = &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 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" : "");
} 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)
{
/* 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 = {
.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 ast_sockaddr sa = { {0, } };
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, }, },
{ "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", DBL, { .d8 = &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_register(&test_engine);
/* Have to associate this with a SIP pvt and an ast_channel */
if (!(p = sip_alloc(0, NULL, 0, SIP_NOTIFY, NULL, 0))) {
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, &sa);
if (!(chan = ast_dummy_channel_alloc())) {
res = AST_TEST_NOT_RUN;
goto done;
}
ast_channel_tech_set(chan, &sip_tech);
ast_channel_tech_pvt_set(chan, dialog_ref(p, "Give the owner channel a reference to the dialog"));
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 unlink and unref will take care of destroying the channel, RTP instance, and SIP pvt */
if (p) {
dialog_unlink_all(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);
}