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chan_sip: Remove deprecated module.

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ASTERISK-30297

Change-Id: Ic700168c80b68879d9cee8bb07afe2712fb17996
This commit is contained in:
Mike Bradeen
2022-11-28 13:05:21 -07:00
committed by George Joseph
parent e66c5da145
commit 4095a382da
80 changed files with 177 additions and 45491 deletions
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2
channels/Makefile
View File

@@ -26,13 +26,11 @@ endif
$(call MOD_ADD_C,chan_iax2,$(wildcard iax2/*.c))
iax2/parser.o: _ASTCFLAGS+=$(call get_menuselect_cflags,MALLOC_DEBUG)
$(call MOD_ADD_C,chan_sip,$(wildcard sip/*.c))
$(call MOD_ADD_C,chan_pjsip,$(wildcard pjsip/*.c))
$(call MOD_ADD_C,chan_dahdi,$(wildcard dahdi/*.c) sig_analog.c sig_pri.c sig_ss7.c)
chan_dahdi.o: _ASTCFLAGS+=$(call get_menuselect_cflags,LOTS_OF_SPANS)
chan_unistim.o: _ASTCFLAGS+=$(AST_NO_FORMAT_TRUNCATION)
chan_phone.o: _ASTCFLAGS+=$(AST_NO_FORMAT_TRUNCATION)
chan_sip.o: _ASTCFLAGS+=$(AST_NO_FORMAT_TRUNCATION)
$(call MOD_ADD_C,console_video.c vgrabbers.c console_board.c)

35946
channels/chan_sip.c
View File

File diff suppressed because it is too large Load Diff

14
channels/console_video.c
View File

@@ -99,14 +99,20 @@ if the formats are equivalent. This will save some unnecessary format
conversion.
In order to handle video you need to add to sip.conf (and presumably
iax.conf too) the following:
In order to handle video you need to add the following to the endpoint in
pjsip.conf
[general](+)
videosupport=yes
allow=h263 ; this or other video formats
allow=h263p ; this or other video formats
(Presumably, iax.conf would require):
[general](+)
videosupport=yes
allow=h263 ; this or other video formats
allow=h263p ; this or other video formats
*/
/*

927
channels/sip/config_parser.c
View File

@@ -1,927 +0,0 @@
/*
* 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 config parsing functions and unit tests
*/
/*** MODULEINFO
<support_level>deprecated</support_level>
***/
#include "asterisk.h"
#include "include/sip.h"
#include "include/config_parser.h"
#include "include/sip_utils.h"
/*! \brief Parse register=> line in sip.conf
*
* \retval 0 on success
* \retval -1 on failure
*/
int sip_parse_register_line(struct sip_registry *reg, int default_expiry, const char *value, int lineno)
{
int portnum = 0;
int domainport = 0;
enum ast_transport transport = AST_TRANSPORT_UDP;
char buf[256] = "";
char *userpart = NULL, *hostpart = NULL;
/* register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry] */
AST_DECLARE_APP_ARGS(pre1,
AST_APP_ARG(peer);
AST_APP_ARG(userpart);
);
AST_DECLARE_APP_ARGS(pre2,
AST_APP_ARG(transport);
AST_APP_ARG(blank);
AST_APP_ARG(userpart);
);
AST_DECLARE_APP_ARGS(user1,
AST_APP_ARG(userpart);
AST_APP_ARG(secret);
AST_APP_ARG(authuser);
);
AST_DECLARE_APP_ARGS(user2,
AST_APP_ARG(user);
AST_APP_ARG(domain);
);
AST_DECLARE_APP_ARGS(user3,
AST_APP_ARG(authuser);
AST_APP_ARG(domainport);
);
AST_DECLARE_APP_ARGS(host1,
AST_APP_ARG(hostpart);
AST_APP_ARG(expiry);
);
AST_DECLARE_APP_ARGS(host2,
AST_APP_ARG(hostpart);
AST_APP_ARG(extension);
);
AST_DECLARE_APP_ARGS(host3,
AST_APP_ARG(host);
AST_APP_ARG(port);
);
if (!reg) {
return -1;
}
reg->expire = -1;
reg->timeout = -1;
if (!value) {
return -1;
}
ast_copy_string(buf, value, sizeof(buf));
/*
* register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry]
* becomes
* userpart => [peer?][transport://]user[@domain][:secret[:authuser]]
* hostpart => host[:port][/extension][~expiry]
*/
if ((hostpart = strrchr(buf, '@'))) {
*hostpart++ = '\0';
userpart = buf;
}
if (ast_strlen_zero(userpart) || ast_strlen_zero(hostpart)) {
ast_log(LOG_WARNING, "Format for registration is [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry] at line %d\n", lineno);
return -1;
}
/*
* pre1.peer => peer
* pre1.userpart => [transport://]user[@domain][:secret[:authuser]]
* hostpart => host[:port][/extension][~expiry]
*/
AST_NONSTANDARD_RAW_ARGS(pre1, userpart, '?');
if (ast_strlen_zero(pre1.userpart)) {
pre1.userpart = pre1.peer;
pre1.peer = NULL;
}
/*
* pre1.peer => peer
* pre2.transport = transport
* pre2.userpart => user[@domain][:secret[:authuser]]
* hostpart => host[:port][/extension][~expiry]
*/
AST_NONSTANDARD_RAW_ARGS(pre2, pre1.userpart, '/');
if (ast_strlen_zero(pre2.userpart)) {
pre2.userpart = pre2.transport;
pre2.transport = NULL;
} else {
pre2.transport[strlen(pre2.transport) - 1] = '\0'; /* Remove trailing : */
}
if (!ast_strlen_zero(pre2.blank)) {
ast_log(LOG_WARNING, "Format for registration is [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry] at line %d\n", lineno);
return -1;
}
/*
* pre1.peer => peer
* pre2.transport = transport
* user1.userpart => user[@domain]
* user1.secret => secret
* user1.authuser => authuser
* hostpart => host[:port][/extension][~expiry]
*/
AST_NONSTANDARD_RAW_ARGS(user1, pre2.userpart, ':');
/*
* pre1.peer => peer
* pre2.transport = transport
* user1.userpart => user[@domain]
* user1.secret => secret
* user1.authuser => authuser
* host1.hostpart => host[:port][/extension]
* host1.expiry => [expiry]
*/
AST_NONSTANDARD_RAW_ARGS(host1, hostpart, '~');
/*
* pre1.peer => peer
* pre2.transport = transport
* user1.userpart => user[@domain]
* user1.secret => secret
* user1.authuser => authuser
* host2.hostpart => host[:port]
* host2.extension => [extension]
* host1.expiry => [expiry]
*/
AST_NONSTANDARD_RAW_ARGS(host2, host1.hostpart, '/');
/*
* pre1.peer => peer
* pre2.transport = transport
* user1.userpart => user[@domain]
* user1.secret => secret
* user1.authuser => authuser
* host3.host => host
* host3.port => port
* host2.extension => extension
* host1.expiry => expiry
*/
AST_NONSTANDARD_RAW_ARGS(host3, host2.hostpart, ':');
/*
* pre1.peer => peer
* pre2.transport = transport
* user2.user => user
* user2.domain => domain
* user1.secret => secret
* user1.authuser => authuser
* host3.host => host
* host3.port => port
* host2.extension => extension
* host1.expiry => expiry
*/
AST_NONSTANDARD_RAW_ARGS(user2, user1.userpart, '@');
/*
* pre1.peer => peer
* pre2.transport = transport
* user2.user => user
* user2.domain => domain
* user1.secret => secret
* user3.authuser => authuser
* user3.domainport => domainport
* host3.host => host
* host3.port => port
* host2.extension => extension
* host1.expiry => expiry
*/
AST_NONSTANDARD_RAW_ARGS(user3, user1.authuser, ':');
/* Reordering needed due to fields being [(:secret[:username])|(:regdomainport:secret:username)]
but parsing being [secret[:username[:regdomainport]]] */
if (user3.argc == 2) {
char *reorder = user3.domainport;
user3.domainport = user1.secret;
user1.secret = user3.authuser;
user3.authuser = reorder;
}
if (host3.port) {
if (!(portnum = port_str2int(host3.port, 0))) {
ast_log(LOG_NOTICE, "'%s' is not a valid port number on line %d of sip.conf. using default.\n", host3.port, lineno);
}
}
if (user3.domainport) {
if (!(domainport = port_str2int(user3.domainport, 0))) {
ast_log(LOG_NOTICE, "'%s' is not a valid domain port number on line %d of sip.conf. using default.\n", user3.domainport, lineno);
}
}
/* set transport type */
if (!pre2.transport) {
transport = AST_TRANSPORT_UDP;
} else if (!strncasecmp(pre2.transport, "tcp", 3)) {
transport = AST_TRANSPORT_TCP;
} else if (!strncasecmp(pre2.transport, "tls", 3)) {
transport = AST_TRANSPORT_TLS;
} else if (!strncasecmp(pre2.transport, "udp", 3)) {
transport = AST_TRANSPORT_UDP;
} else {
transport = AST_TRANSPORT_UDP;
ast_log(LOG_NOTICE, "'%.3s' is not a valid transport type on line %d of sip.conf. defaulting to udp.\n", pre2.transport, lineno);
}
/* if no portnum specified, set default for transport */
if (!portnum) {
if (transport == AST_TRANSPORT_TLS) {
portnum = STANDARD_TLS_PORT;
} else {
portnum = STANDARD_SIP_PORT;
}
}
/* copy into sip_registry object */
ast_string_field_set(reg, callback, ast_strip_quoted(S_OR(host2.extension, "s"), "\"", "\""));
ast_string_field_set(reg, username, ast_strip_quoted(S_OR(user2.user, ""), "\"", "\""));
ast_string_field_set(reg, hostname, ast_strip_quoted(S_OR(host3.host, ""), "\"", "\""));
ast_string_field_set(reg, authuser, ast_strip_quoted(S_OR(user3.authuser, ""), "\"", "\""));
ast_string_field_set(reg, secret, ast_strip_quoted(S_OR(user1.secret, ""), "\"", "\""));
ast_string_field_set(reg, peername, ast_strip_quoted(S_OR(pre1.peer, ""), "\"", "\""));
ast_string_field_set(reg, regdomain, ast_strip_quoted(S_OR(user2.domain, ""), "\"", "\""));
reg->transport = transport;
reg->portno = portnum;
reg->regdomainport = domainport;
reg->callid_valid = FALSE;
reg->ocseq = INITIAL_CSEQ;
reg->refresh = reg->expiry = reg->configured_expiry = (host1.expiry ? atoi(ast_strip_quoted(host1.expiry, "\"", "\"")) : default_expiry);
return 0;
}
#ifdef TEST_FRAMEWORK
AST_TEST_DEFINE(sip_parse_register_line_test)
{
int res = AST_TEST_PASS;
struct sip_registry *reg;
int default_expiry = 120;
const char *reg1 = "name@domain";
const char *reg2 = "name:pass@domain";
const char *reg3 = "name@namedomain:pass:authuser@domain";
const char *reg4 = "name@namedomain:pass:authuser@domain/extension";
const char *reg5 = "tcp://name@namedomain:pass:authuser@domain/extension";
const char *reg6 = "tls://name@namedomain:pass:authuser@domain/extension~111";
const char *reg7 = "peer?tcp://name@namedomain:pass:authuser@domain:1234/extension~111";
const char *reg8 = "peer?name@namedomain:pass:authuser@domain:1234/extension~111";
const char *reg9 = "peer?name:pass:authuser:1234/extension~111";
const char *reg10 = "@domin:1234";
const char *reg12 = "name@namedomain:4321:pass:authuser@domain";
const char *reg13 = "name@namedomain:4321::@domain";
switch (cmd) {
case TEST_INIT:
info->name = "sip_parse_register_line_test";
info->category = "/channels/chan_sip/";
info->summary = "tests sip register line parsing";
info->description =
"Tests parsing of various register line configurations. "
"Verifies output matches expected behavior.";
return AST_TEST_NOT_RUN;
case TEST_EXECUTE:
break;
}
/* ---Test reg 1, simple config --- */
if (!(reg = ast_calloc_with_stringfields(1, struct sip_registry, 256))) {
goto alloc_fail;
} else if (
sip_parse_register_line(reg, default_expiry, reg1, 1) ||
strcmp(reg->callback, "s") ||
strcmp(reg->username, "name") ||
strcmp(reg->regdomain, "") ||
strcmp(reg->hostname, "domain") ||
strcmp(reg->authuser, "") ||
strcmp(reg->secret, "") ||
strcmp(reg->peername, "") ||
reg->transport != AST_TRANSPORT_UDP ||
reg->timeout != -1 ||
reg->expire != -1 ||
reg->refresh != default_expiry ||
reg->expiry != default_expiry ||
reg->configured_expiry != default_expiry ||
reg->portno != STANDARD_SIP_PORT ||
(reg->regdomainport) ||
reg->callid_valid != FALSE ||
reg->ocseq != INITIAL_CSEQ) {
ast_test_status_update(test, "Test 1: simple config failed\n");
res = AST_TEST_FAIL;
}
ast_string_field_free_memory(reg);
ast_free(reg);
/* ---Test reg 2, add secret --- */
if (!(reg = ast_calloc_with_stringfields(1, struct sip_registry, 256))) {
goto alloc_fail;
} else if (
sip_parse_register_line(reg, default_expiry, reg2, 1) ||
strcmp(reg->callback, "s") ||
strcmp(reg->username, "name") ||
strcmp(reg->regdomain, "") ||
strcmp(reg->hostname, "domain") ||
strcmp(reg->authuser, "") ||
strcmp(reg->secret, "pass") ||
strcmp(reg->peername, "") ||
reg->transport != AST_TRANSPORT_UDP ||
reg->timeout != -1 ||
reg->expire != -1 ||
reg->refresh != default_expiry ||
reg->expiry != default_expiry ||
reg->configured_expiry != default_expiry ||
reg->portno != STANDARD_SIP_PORT ||
(reg->regdomainport) ||
reg->callid_valid != FALSE ||
reg->ocseq != INITIAL_CSEQ) {
ast_test_status_update(test, "Test 2: add secret failed\n");
res = AST_TEST_FAIL;
}
ast_string_field_free_memory(reg);
ast_free(reg);
/* ---Test reg 3, add userdomain and authuser --- */
if (!(reg = ast_calloc_with_stringfields(1, struct sip_registry, 256))) {
goto alloc_fail;
} else if (
sip_parse_register_line(reg, default_expiry, reg3, 1) ||
strcmp(reg->callback, "s") ||
strcmp(reg->username, "name") ||
strcmp(reg->regdomain, "namedomain") ||
strcmp(reg->hostname, "domain") ||
strcmp(reg->authuser, "authuser") ||
strcmp(reg->secret, "pass") ||
strcmp(reg->peername, "") ||
reg->transport != AST_TRANSPORT_UDP ||
reg->timeout != -1 ||
reg->expire != -1 ||
reg->refresh != default_expiry ||
reg->expiry != default_expiry ||
reg->configured_expiry != default_expiry ||
reg->portno != STANDARD_SIP_PORT ||
(reg->regdomainport) ||
reg->callid_valid != FALSE ||
reg->ocseq != INITIAL_CSEQ) {
ast_test_status_update(test, "Test 3: add userdomain and authuser failed\n");
res = AST_TEST_FAIL;
}
ast_string_field_free_memory(reg);
ast_free(reg);
/* ---Test reg 4, add callback extension --- */
if (!(reg = ast_calloc_with_stringfields(1, struct sip_registry, 256))) {
goto alloc_fail;
} else if (
sip_parse_register_line(reg, default_expiry, reg4, 1) ||
strcmp(reg->callback, "extension") ||
strcmp(reg->username, "name") ||
strcmp(reg->regdomain, "namedomain") ||
strcmp(reg->hostname, "domain") ||
strcmp(reg->authuser, "authuser") ||
strcmp(reg->secret, "pass") ||
strcmp(reg->peername, "") ||
reg->transport != AST_TRANSPORT_UDP ||
reg->timeout != -1 ||
reg->expire != -1 ||
reg->refresh != default_expiry ||
reg->expiry != default_expiry ||
reg->configured_expiry != default_expiry ||
reg->portno != STANDARD_SIP_PORT ||
(reg->regdomainport) ||
reg->callid_valid != FALSE ||
reg->ocseq != INITIAL_CSEQ) {
ast_test_status_update(test, "Test 4: add callback extension failed\n");
res = AST_TEST_FAIL;
}
ast_string_field_free_memory(reg);
ast_free(reg);
/* ---Test reg 5, add transport --- */
if (!(reg = ast_calloc_with_stringfields(1, struct sip_registry, 256))) {
goto alloc_fail;
} else if (
sip_parse_register_line(reg, default_expiry, reg5, 1) ||
strcmp(reg->callback, "extension") ||
strcmp(reg->username, "name") ||
strcmp(reg->regdomain, "namedomain") ||
strcmp(reg->hostname, "domain") ||
strcmp(reg->authuser, "authuser") ||
strcmp(reg->secret, "pass") ||
strcmp(reg->peername, "") ||
reg->transport != AST_TRANSPORT_TCP ||
reg->timeout != -1 ||
reg->expire != -1 ||
reg->refresh != default_expiry ||
reg->expiry != default_expiry ||
reg->configured_expiry != default_expiry ||
reg->portno != STANDARD_SIP_PORT ||
(reg->regdomainport) ||
reg->callid_valid != FALSE ||
reg->ocseq != INITIAL_CSEQ) {
ast_test_status_update(test, "Test 5: add transport failed\n");
res = AST_TEST_FAIL;
}
ast_string_field_free_memory(reg);
ast_free(reg);
/* ---Test reg 6, change to tls transport, add expiry --- */
if (!(reg = ast_calloc_with_stringfields(1, struct sip_registry, 256))) {
goto alloc_fail;
} else if (
sip_parse_register_line(reg, default_expiry, reg6, 1) ||
strcmp(reg->callback, "extension") ||
strcmp(reg->username, "name") ||
strcmp(reg->regdomain, "namedomain") ||
strcmp(reg->hostname, "domain") ||
strcmp(reg->authuser, "authuser") ||
strcmp(reg->secret, "pass") ||
strcmp(reg->peername, "") ||
reg->transport != AST_TRANSPORT_TLS ||
reg->timeout != -1 ||
reg->expire != -1 ||
reg->refresh != 111 ||
reg->expiry != 111 ||
reg->configured_expiry != 111 ||
reg->portno != STANDARD_TLS_PORT ||
(reg->regdomainport) ||
reg->callid_valid != FALSE ||
reg->ocseq != INITIAL_CSEQ) {
ast_test_status_update(test, "Test 6: change to tls transport and add expiry failed\n");
res = AST_TEST_FAIL;
}
ast_string_field_free_memory(reg);
ast_free(reg);
/* ---Test reg 7, change transport to tcp, add custom port, and add peer --- */
if (!(reg = ast_calloc_with_stringfields(1, struct sip_registry, 256))) {
goto alloc_fail;
} else if (
sip_parse_register_line(reg, default_expiry, reg7, 1) ||
strcmp(reg->callback, "extension") ||
strcmp(reg->username, "name") ||
strcmp(reg->regdomain, "namedomain") ||
strcmp(reg->hostname, "domain") ||
strcmp(reg->authuser, "authuser") ||
strcmp(reg->secret, "pass") ||
strcmp(reg->peername, "peer") ||
reg->transport != AST_TRANSPORT_TCP ||
reg->timeout != -1 ||
reg->expire != -1 ||
reg->refresh != 111 ||
reg->expiry != 111 ||
reg->configured_expiry != 111 ||
reg->portno != 1234 ||
(reg->regdomainport) ||
reg->callid_valid != FALSE ||
reg->ocseq != INITIAL_CSEQ) {
ast_test_status_update(test, "Test 7, change transport to tcp, add custom port, and add peer failed.\n");
res = AST_TEST_FAIL;
}
ast_string_field_free_memory(reg);
ast_free(reg);
/* ---Test reg 8, remove transport --- */
if (!(reg = ast_calloc_with_stringfields(1, struct sip_registry, 256))) {
goto alloc_fail;
} else if (
sip_parse_register_line(reg, default_expiry, reg8, 1) ||
strcmp(reg->callback, "extension") ||
strcmp(reg->username, "name") ||
strcmp(reg->regdomain, "namedomain") ||
strcmp(reg->hostname, "domain") ||
strcmp(reg->authuser, "authuser") ||
strcmp(reg->secret, "pass") ||
strcmp(reg->peername, "peer") ||
reg->transport != AST_TRANSPORT_UDP ||
reg->timeout != -1 ||
reg->expire != -1 ||
reg->refresh != 111 ||
reg->expiry != 111 ||
reg->configured_expiry != 111 ||
reg->portno != 1234 ||
(reg->regdomainport) ||
reg->callid_valid != FALSE ||
reg->ocseq != INITIAL_CSEQ) {
ast_test_status_update(test, "Test 8, remove transport failed.\n");
res = AST_TEST_FAIL;
}
ast_string_field_free_memory(reg);
ast_free(reg);
/* ---Test reg 9, missing domain, expected to fail --- */
if (!(reg = ast_calloc_with_stringfields(1, struct sip_registry, 256))) {
goto alloc_fail;
} else if (!sip_parse_register_line(reg, default_expiry, reg9, 1)) {
ast_test_status_update(test,
"Test 9, missing domain, expected to fail but did not.\n");
res = AST_TEST_FAIL;
}
ast_string_field_free_memory(reg);
ast_free(reg);
/* ---Test reg 10, missing user, expected to fail --- */
if (!(reg = ast_calloc_with_stringfields(1, struct sip_registry, 256))) {
goto alloc_fail;
} else if (!sip_parse_register_line(reg, default_expiry, reg10, 1)) {
ast_test_status_update(test,
"Test 10, missing user expected to fail but did not\n");
res = AST_TEST_FAIL;
}
ast_string_field_free_memory(reg);
ast_free(reg);
/* ---Test reg 11, no registry object, expected to fail--- */
if (!sip_parse_register_line(NULL, default_expiry, reg1, 1)) {
ast_test_status_update(test,
"Test 11, no registry object, expected to fail but did not.\n");
res = AST_TEST_FAIL;
}
/* ---Test reg 12, no registry line, expected to fail --- */
if (!(reg = ast_calloc_with_stringfields(1, struct sip_registry, 256))) {
goto alloc_fail;
} else if (!sip_parse_register_line(reg, default_expiry, NULL, 1)) {
ast_test_status_update(test,
"Test 12, NULL register line expected to fail but did not.\n");
res = AST_TEST_FAIL;
}
ast_string_field_free_memory(reg);
ast_free(reg);
/* ---Test reg13, add domain port --- */
if (!(reg = ast_calloc_with_stringfields(1, struct sip_registry, 256))) {
goto alloc_fail;
} else if (
sip_parse_register_line(reg, default_expiry, reg12, 1) ||
strcmp(reg->callback, "s") ||
strcmp(reg->username, "name") ||
strcmp(reg->regdomain, "namedomain") ||
strcmp(reg->hostname, "domain") ||
strcmp(reg->authuser, "authuser") ||
strcmp(reg->secret, "pass") ||
strcmp(reg->peername, "") ||
reg->transport != AST_TRANSPORT_UDP ||
reg->timeout != -1 ||
reg->expire != -1 ||
reg->refresh != default_expiry ||
reg->expiry != default_expiry ||
reg->configured_expiry != default_expiry ||
reg->portno != STANDARD_SIP_PORT ||
reg->regdomainport != 4321 ||
reg->callid_valid != FALSE ||
reg->ocseq != INITIAL_CSEQ) {
ast_test_status_update(test, "Test 13, add domain port failed.\n");
res = AST_TEST_FAIL;
}
ast_string_field_free_memory(reg);
ast_free(reg);
/* ---Test reg14, domain port without secret --- */
if (!(reg = ast_calloc_with_stringfields(1, struct sip_registry, 256))) {
goto alloc_fail;
} else if (
sip_parse_register_line(reg, default_expiry, reg13, 1) ||
strcmp(reg->callback, "s") ||
strcmp(reg->username, "name") ||
strcmp(reg->regdomain, "namedomain") ||
strcmp(reg->hostname, "domain") ||
strcmp(reg->authuser, "") ||
strcmp(reg->secret, "") ||
strcmp(reg->peername, "") ||
reg->transport != AST_TRANSPORT_UDP ||
reg->timeout != -1 ||
reg->expire != -1 ||
reg->refresh != default_expiry ||
reg->expiry != default_expiry ||
reg->configured_expiry != default_expiry ||
reg->portno != STANDARD_SIP_PORT ||
reg->regdomainport != 4321 ||
reg->callid_valid != FALSE ||
reg->ocseq != INITIAL_CSEQ) {
ast_test_status_update(test, "Test 14, domain port without secret failed.\n");
res = AST_TEST_FAIL;
}
ast_string_field_free_memory(reg);
ast_free(reg);
return res;
alloc_fail:
ast_test_status_update(test, "Out of memory. \n");
return res;
}
#endif
int sip_parse_host(char *line, int lineno, char **hostname, int *portnum, enum ast_transport *transport)
{
char *port;
if (ast_strlen_zero(line)) {
*hostname = NULL;
return -1;
}
if ((*hostname = strstr(line, "://"))) {
*hostname += 3;
if (!strncasecmp(line, "tcp", 3)) {
*transport = AST_TRANSPORT_TCP;
} else if (!strncasecmp(line, "tls", 3)) {
*transport = AST_TRANSPORT_TLS;
} else if (!strncasecmp(line, "udp", 3)) {
*transport = AST_TRANSPORT_UDP;
} else if (lineno) {
ast_log(LOG_NOTICE, "'%.3s' is not a valid transport type on line %d of sip.conf. defaulting to udp.\n", line, lineno);
} else {
ast_log(LOG_NOTICE, "'%.3s' is not a valid transport type in sip config. defaulting to udp.\n", line);
}
} else {
*hostname = line;
*transport = AST_TRANSPORT_UDP;
}
if ((line = strrchr(*hostname, '@')))
line++;
else
line = *hostname;
if (ast_sockaddr_split_hostport(line, hostname, &port, 0) == 0) {
if (lineno) {
ast_log(LOG_WARNING, "Cannot parse host '%s' on line %d of sip.conf.\n",
line, lineno);
} else {
ast_log(LOG_WARNING, "Cannot parse host '%s' in sip config.\n", line);
}
return -1;
}
if (port) {
if (!sscanf(port, "%5d", portnum)) {
if (lineno) {
ast_log(LOG_NOTICE, "'%s' is not a valid port number on line %d of sip.conf. using default.\n", port, lineno);
} else {
ast_log(LOG_NOTICE, "'%s' is not a valid port number in sip config. using default.\n", port);
}
port = NULL;
}
}
if (!port) {
if (*transport & AST_TRANSPORT_TLS) {
*portnum = STANDARD_TLS_PORT;
} else {
*portnum = STANDARD_SIP_PORT;
}
}
return 0;
}
#ifdef TEST_FRAMEWORK
AST_TEST_DEFINE(sip_parse_host_line_test)
{
int res = AST_TEST_PASS;
char *host;
int port;
enum ast_transport transport;
char host1[] = "www.blah.com";
char host2[] = "tcp://www.blah.com";
char host3[] = "tls://10.10.10.10";
char host4[] = "tls://10.10.10.10:1234";
char host5[] = "10.10.10.10:1234";
switch (cmd) {
case TEST_INIT:
info->name = "sip_parse_host_line_test";
info->category = "/channels/chan_sip/";
info->summary = "tests sip.conf host line parsing";
info->description =
"Tests parsing of various host line configurations. "
"Verifies output matches expected behavior.";
return AST_TEST_NOT_RUN;
case TEST_EXECUTE:
break;
}
/* test 1, simple host */
sip_parse_host(host1, 1, &host, &port, &transport);
if (port != STANDARD_SIP_PORT ||
ast_strlen_zero(host) || strcmp(host, "www.blah.com") ||
transport != AST_TRANSPORT_UDP) {
ast_test_status_update(test, "Test 1: simple host failed.\n");
res = AST_TEST_FAIL;
}
/* test 2, add tcp transport */
sip_parse_host(host2, 1, &host, &port, &transport);
if (port != STANDARD_SIP_PORT ||
ast_strlen_zero(host) || strcmp(host, "www.blah.com") ||
transport != AST_TRANSPORT_TCP) {
ast_test_status_update(test, "Test 2: tcp host failed.\n");
res = AST_TEST_FAIL;
}
/* test 3, add tls transport */
sip_parse_host(host3, 1, &host, &port, &transport);
if (port != STANDARD_TLS_PORT ||
ast_strlen_zero(host) || strcmp(host, "10.10.10.10") ||
transport != AST_TRANSPORT_TLS) {
ast_test_status_update(test, "Test 3: tls host failed. \n");
res = AST_TEST_FAIL;
}
/* test 4, add custom port with tls */
sip_parse_host(host4, 1, &host, &port, &transport);
if (port != 1234 || ast_strlen_zero(host) ||
strcmp(host, "10.10.10.10") ||
transport != AST_TRANSPORT_TLS) {
ast_test_status_update(test, "Test 4: tls host with custom port failed.\n");
res = AST_TEST_FAIL;
}
/* test 5, simple host with custom port */
sip_parse_host(host5, 1, &host, &port, &transport);
if (port != 1234 || ast_strlen_zero(host) ||
strcmp(host, "10.10.10.10") ||
transport != AST_TRANSPORT_UDP) {
ast_test_status_update(test, "Test 5: simple host with custom port failed.\n");
res = AST_TEST_FAIL;
}
/* test 6, expected failure with NULL input */
if (!sip_parse_host(NULL, 1, &host, &port, &transport)) {
ast_test_status_update(test, "Test 6: expected error on NULL input did not occur.\n");
res = AST_TEST_FAIL;
}
return res;
}
#endif
/*! \brief Parse the comma-separated nat= option values */
void sip_parse_nat_option(const char *value, struct ast_flags *mask, struct ast_flags *flags)
{
char *parse, *this;
if (!(parse = ast_strdupa(value))) {
return;
}
/* Since we need to completely override the general settings if we are being called
* later for a peer, always set the flags for all options on the mask */
ast_set_flag(&mask[0], SIP_NAT_FORCE_RPORT);
ast_set_flag(&mask[1], SIP_PAGE2_SYMMETRICRTP);
ast_set_flag(&mask[2], SIP_PAGE3_NAT_AUTO_RPORT);
ast_set_flag(&mask[2], SIP_PAGE3_NAT_AUTO_COMEDIA);
while ((this = strsep(&parse, ","))) {
if (ast_false(this)) {
ast_clear_flag(&flags[0], SIP_NAT_FORCE_RPORT);
ast_clear_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP);
ast_clear_flag(&flags[2], SIP_PAGE3_NAT_AUTO_RPORT);
ast_clear_flag(&flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA);
break; /* It doesn't make sense to have no + something else */
} else if (!strcasecmp(this, "yes")) {
ast_log(LOG_WARNING, "nat=yes is deprecated, use nat=force_rport,comedia instead\n");
ast_set_flag(&flags[0], SIP_NAT_FORCE_RPORT);
ast_set_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP);
ast_clear_flag(&flags[2], SIP_PAGE3_NAT_AUTO_RPORT);
ast_clear_flag(&flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA);
break; /* It doesn't make sense to have yes + something else */
} else if (!strcasecmp(this, "force_rport") && !ast_test_flag(&flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) {
ast_set_flag(&flags[0], SIP_NAT_FORCE_RPORT);
} else if (!strcasecmp(this, "comedia") && !ast_test_flag(&flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA)) {
ast_set_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP);
} else if (!strcasecmp(this, "auto_force_rport")) {
ast_set_flag(&flags[2], SIP_PAGE3_NAT_AUTO_RPORT);
/* In case someone did something dumb like nat=force_rport,auto_force_rport */
ast_clear_flag(&flags[0], SIP_NAT_FORCE_RPORT);
} else if (!strcasecmp(this, "auto_comedia")) {
ast_set_flag(&flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA);
/* In case someone did something dumb like nat=comedia,auto_comedia*/
ast_clear_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP);
}
}
}
#ifdef TEST_FRAMEWORK
#define TEST_FORCE_RPORT 1 << 0
#define TEST_COMEDIA 1 << 1
#define TEST_AUTO_FORCE_RPORT 1 << 2
#define TEST_AUTO_COMEDIA 1 << 3
static int match_nat_options(int val, struct ast_flags *flags)
{
if ((!ast_test_flag(&flags[0], SIP_NAT_FORCE_RPORT)) != !(val & TEST_FORCE_RPORT)) {
return 0;
}
if (!ast_test_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP) != !(val & TEST_COMEDIA)) {
return 0;
}
if (!ast_test_flag(&flags[2], SIP_PAGE3_NAT_AUTO_RPORT) != !(val & TEST_AUTO_FORCE_RPORT)) {
return 0;
}
if (!ast_test_flag(&flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA) != !(val & TEST_AUTO_COMEDIA)) {
return 0;
}
return 1;
}
AST_TEST_DEFINE(sip_parse_nat_test)
{
int i, res = AST_TEST_PASS;
struct ast_flags mask[3] = {{0}}, flags[3] = {{0}};
struct {
const char *str;
int i;
} options[] = {
{ "yes", TEST_FORCE_RPORT | TEST_COMEDIA },
{ "no", 0 },
{ "force_rport", TEST_FORCE_RPORT },
{ "comedia", TEST_COMEDIA },
{ "auto_force_rport", TEST_AUTO_FORCE_RPORT },
{ "auto_comedia", TEST_AUTO_COMEDIA },
{ "force_rport,auto_force_rport", TEST_AUTO_FORCE_RPORT },
{ "auto_force_rport,force_rport", TEST_AUTO_FORCE_RPORT },
{ "comedia,auto_comedia", TEST_AUTO_COMEDIA },
{ "auto_comedia,comedia", TEST_AUTO_COMEDIA },
{ "force_rport,comedia", TEST_FORCE_RPORT | TEST_COMEDIA },
{ "force_rport,auto_comedia", TEST_FORCE_RPORT | TEST_AUTO_COMEDIA },
{ "force_rport,yes,no", TEST_FORCE_RPORT | TEST_COMEDIA },
{ "auto_comedia,no,yes", 0 },
};
switch (cmd) {
case TEST_INIT:
info->name = "sip_parse_nat_test";
info->category = "/channels/chan_sip/";
info->summary = "tests sip.conf nat line parsing";
info->description =
"Tests parsing of various nat line configurations. "
"Verifies output matches expected behavior.";
return AST_TEST_NOT_RUN;
case TEST_EXECUTE:
break;
}
for (i = 0; i < ARRAY_LEN(options); i++) {
sip_parse_nat_option(options[i].str, mask, flags);
if (!match_nat_options(options[i].i, flags)) {
ast_test_status_update(test, "Failed nat=%s\n", options[i].str);
res = AST_TEST_FAIL;
}
memset(flags, 0, sizeof(flags));
memset(mask, 0, sizeof(mask));
}
return res;
}
#endif
/*! \brief SIP test registration */
void sip_config_parser_register_tests(void)
{
AST_TEST_REGISTER(sip_parse_register_line_test);
AST_TEST_REGISTER(sip_parse_host_line_test);
AST_TEST_REGISTER(sip_parse_nat_test);
}
/*! \brief SIP test registration */
void sip_config_parser_unregister_tests(void)
{
AST_TEST_UNREGISTER(sip_parse_register_line_test);
AST_TEST_UNREGISTER(sip_parse_host_line_test);
AST_TEST_UNREGISTER(sip_parse_nat_test);
}

515
channels/sip/dialplan_functions.c
View File

@@ -1,515 +0,0 @@
/*
* 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>deprecated</support_level>
***/
/*** DOCUMENTATION
<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>
<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"
#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, "ruri")) {
if (p->initreq.data) {
char *tmpruri = REQ_OFFSET_TO_STR(&p->initreq, rlpart2);
ast_copy_string(buf, tmpruri, buflen);
} else {
return -1;
}
} 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");
}
if (chan) {
ast_channel_unref(chan);
}
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);
}

68
channels/sip/include/config_parser.h
View File

@@ -1,68 +0,0 @@
/*
* 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.conf parser header file
*/
#include "sip.h"
#ifndef _SIP_CONF_PARSE_H
#define _SIP_CONF_PARSE_H
/*!
* \brief Parse register=> line in sip.conf
*
* \retval 0 on success
* \retval -1 on failure
*/
int sip_parse_register_line(struct sip_registry *reg, int default_expiry, const char *value, int lineno);
/*!
* \brief parses a config line for a host with a transport
*
* An example input would be:
* <code>tls://www.google.com:8056</code>
*
* \retval 0 on success
* \retval -1 on failure
*/
int sip_parse_host(char *line, int lineno, char **hostname, int *portnum, enum ast_transport *transport);
/*! \brief Parse the comma-separated nat= option values
* \param value The comma-separated value
* \param mask An array of ast_flags that will be set by this function
* and used as a mask for copying the flags later
* \param flags An array of ast_flags that will be set by this function
*
* \note The nat-related values in both mask and flags are assumed to empty. This function
* will treat the first "yes" or "no" value in a list of values as overriding all other values
* and will stop parsing. Auto values will override their non-auto counterparts.
*/
void sip_parse_nat_option(const char *value, struct ast_flags *mask, struct ast_flags *flags);
/*!
* \brief register config parsing tests
*/
void sip_config_parser_register_tests(void);
/*!
* \brief unregister config parsing tests
*/
void sip_config_parser_unregister_tests(void);
#endif

82
channels/sip/include/dialog.h
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@@ -1,82 +0,0 @@
/*
* 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(dialog, tag) ao2_t_bump(dialog, tag)
#define dialog_unref(dialog, tag) ({ ao2_t_cleanup(dialog, tag); (NULL); })
struct sip_pvt *__sip_alloc(ast_string_field callid, struct ast_sockaddr *sin,
int useglobal_nat, const int intended_method, struct sip_request *req, ast_callid logger_callid,
const char *file, int line, const char *func);
#define sip_alloc(callid, addr, useglobal_nat, intended_method, req, logger_callid) \
__sip_alloc(callid, addr, useglobal_nat, intended_method, req, logger_callid, __FILE__, __LINE__, __PRETTY_FUNCTION__)
/*!
* \brief Schedule final destruction of SIP dialog.
*
* \note This cannot be canceled.
*
* \details
* This function is used to keep a dialog around for a period of time in order
* to properly respond to any retransmits.
*/
void sip_scheddestroy_final(struct sip_pvt *p, int ms);
/*! \brief Schedule destruction of SIP dialog */
void sip_scheddestroy(struct sip_pvt *p, int ms);
/*! \brief Cancel destruction of SIP dialog. */
void sip_cancel_destroy(struct sip_pvt *pvt);
/*!
* \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.
*
* \note The dialog must not be locked when called.
*/
void dialog_unlink_all(struct sip_pvt *dialog);
/*! \brief Acknowledges receipt of a packet and stops retransmission
* called with p locked*/
int __sip_ack(struct sip_pvt *p, uint32_t 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, uint32_t seqno, int resp, int sipmethod);
#endif /* defined(_SIP_DIALOG_H) */

41
channels/sip/include/dialplan_functions.h
View File

@@ -1,41 +0,0 @@
/*
* 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) */

41
channels/sip/include/globals.h
View File

@@ -1,41 +0,0 @@
/*
* 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 ast_sockaddr bindaddr; /*!< UDP: The address we bind to */
extern struct ast_sched_context *sched; /*!< The scheduling context */
/*! \brief Definition of this channel for PBX channel registration */
extern 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) */

250
channels/sip/include/reqresp_parser.h
View File

@@ -1,250 +0,0 @@
/*
* 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 request response parser header file
*/
#ifndef _SIP_REQRESP_H
#define _SIP_REQRESP_H
/*! \brief uri parameters */
struct uriparams {
char *transport;
char *user;
char *method;
char *ttl;
char *maddr;
int lr;
};
struct contact {
AST_LIST_ENTRY(contact) list;
char *name;
char *user;
char *pass;
char *hostport;
struct uriparams params;
char *headers;
char *expires;
char *q;
};
AST_LIST_HEAD_NOLOCK(contactliststruct, contact);
/*!
* \brief parses a URI in its components.
*
* \note
* - Multiple scheme's can be specified ',' delimited. ex: "sip:,sips:"
* - If a component is not requested, do not split around it. This means
* that if we don't have domain, we cannot split name:pass.
* - It is safe to call with ret_name, pass, hostport pointing all to
* the same place.
* - If no secret parameter is provided, ret_name will return with both
* parts, user:secret.
* - If the URI contains a port number, hostport will return with both
* parts, host:port.
* - This function overwrites the URI string.
*
* \retval 0 on success
* \retval -1 on error.
*
* \verbatim
* general form we are expecting is sip:user:password;user-parameters@host:port;uri-parameters?headers
* \endverbatim
*/
int parse_uri(char *uri, const char *scheme, char **ret_name, char **pass,
char **hostport, char **transport);
/*!
* \brief parses a URI in to all of its components and any trailing residue
*
* \retval 0 on success
* \retval -1 on error.
*
*/
int parse_uri_full(char *uri, const char *scheme, char **user, char **pass,
char **hostport, struct uriparams *params, char **headers,
char **residue);
/*!
* \brief Get caller id name from SIP headers, copy into output buffer
*
* \return input string pointer placed after display-name field if possible
*/
const char *get_calleridname(const char *input, char *output, size_t outputsize);
/*!
* \brief Get name and number from sip header
*
* \note name and number point to malloced memory on return and must be
* freed. If name or number is not found, they will be returned as NULL.
*
* \retval 0 success
* \retval -1 failure
*/
int get_name_and_number(const char *hdr, char **name, char **number);
/*! \brief Pick out text in brackets from character string
* \return pointer to terminated stripped string
* \param tmp input string that will be modified
*
* Examples:
* \verbatim
* "foo" <bar> valid input, returns bar
* foo returns the whole string
* < "foo ... > returns the string between brackets
* < "foo... bogus (missing closing bracket), returns the whole string
* \endverbatim
*/
char *get_in_brackets(char *tmp);
/*! \brief Get text in brackets on a const without copy
*
* \param src String to search
* \param[out] start Set to first character inside left bracket.
* \param[out] length Set to lenght of string inside brackets
* \retval 0 success
* \retval -1 failure
* \retval 1 no brackets so got all
*/
int get_in_brackets_const(const char *src,const char **start,int *length);
/*! \brief Get text in brackets and any trailing residue
*
* \retval 0 success
* \retval -1 failure
* \retval 1 no brackets so got all
*/
int get_in_brackets_full(char *tmp, char **out, char **residue);
/*! \brief Parse the ABNF structure
* name-andor-addr = name-addr / addr-spec
* into its components and return any trailing message-header parameters
*
* \retval 0 success
* \retval -1 failure
*/
int parse_name_andor_addr(char *uri, const char *scheme, char **name,
char **user, char **pass, char **domain,
struct uriparams *params, char **headers,
char **residue);
/*! \brief Parse all contact header contacts
* \retval 0 success
* \retval -1 failure
* \retval 1 all contacts (star)
*/
int get_comma(char *parse, char **out);
int parse_contact_header(char *contactheader, struct contactliststruct *contactlist);
/*!
* \brief register request parsing tests
*/
void sip_request_parser_register_tests(void);
/*!
* \brief unregister request parsing tests
*/
void sip_request_parser_unregister_tests(void);
/*!
* \brief Parse supported header in incoming packet
*
* \details This function parses through the options parameters and
* builds a bit field representing all the SIP options in that field. When an
* item is found that is not supported, it is copied to the unsupported
* out buffer.
*
* \param options list
* \param[in,out] unsupported buffer (optional)
* \param[in,out] unsupported_len buffer length
*
* \note Because this function can be called multiple times, it will append
* whatever options are specified in \c options to \c unsupported. Callers
* of this function should make sure the unsupported buffer is clear before
* calling this function.
*/
unsigned int parse_sip_options(const char *options, char *unsupported, size_t unsupported_len);
/*!
* \brief Compare two URIs as described in RFC 3261 Section 19.1.4
*
* \param input1 First URI
* \param input2 Second URI
* \retval 0 URIs match
* \retval nonzero URIs do not match or one or both is malformed
*/
int sip_uri_cmp(const char *input1, const char *input2);
/*!
* \brief initialize request and response parser data
*
* \retval 0 Success
* \retval -1 Failure
*/
int sip_reqresp_parser_init(void);
/*!
* \brief Free resources used by request and response parser
*/
void sip_reqresp_parser_exit(void);
/*!
* \brief Parse a Via header
*
* This function parses the Via header and processes it according to section
* 18.2 of RFC 3261 and RFC 3581. Since we don't have a transport layer, we
* only care about the maddr and ttl parms. The received and rport params are
* not parsed.
*
* \note This function fails to parse some odd combinations of SWS in parameter
* lists.
*
* \code
* VIA syntax. RFC 3261 section 25.1
* Via = ( "Via" / "v" ) HCOLON via-parm *(COMMA via-parm)
* via-parm = sent-protocol LWS sent-by *( SEMI via-params )
* via-params = via-ttl / via-maddr
* / via-received / via-branch
* / via-extension
* via-ttl = "ttl" EQUAL ttl
* via-maddr = "maddr" EQUAL host
* via-received = "received" EQUAL (IPv4address / IPv6address)
* via-branch = "branch" EQUAL token
* via-extension = generic-param
* sent-protocol = protocol-name SLASH protocol-version
* SLASH transport
* protocol-name = "SIP" / token
* protocol-version = token
* transport = "UDP" / "TCP" / "TLS" / "SCTP"
* / other-transport
* sent-by = host [ COLON port ]
* ttl = 1*3DIGIT ; 0 to 255
* \endcode
*/
struct sip_via *parse_via(const char *header);
/*!
* \brief Free parsed Via data.
*/
void free_via(struct sip_via *v);
#endif

117
channels/sip/include/route.h
View File

@@ -1,117 +0,0 @@
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2013, 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_route header file
*/
#ifndef _SIP_ROUTE_H
#define _SIP_ROUTE_H
#include "asterisk/linkedlists.h"
#include "asterisk/strings.h"
/*!
* \brief Opaque storage of a sip route hop
*/
struct sip_route_hop;
/*!
* \internal \brief Internal enum to remember last calculated
*/
enum sip_route_type {
route_loose = 0, /*!< The first hop contains ;lr or does not exist */
route_strict, /*!< The first hop exists and does not contain ;lr */
route_invalidated, /*!< strict/loose routing needs to be rechecked */
};
/*!
* \brief Structure to store route information
*
* \note This must be zero-filled on allocation
*/
struct sip_route {
AST_LIST_HEAD_NOLOCK(, sip_route_hop) list;
enum sip_route_type type;
};
/*!
* \brief Add a new hop to the route
*
* \param route Route
* \param uri Address of this hop
* \param len Length of hop not including null terminator
* \param inserthead If true then inserted the new route to the top of the list
*
* \return Pointer to null terminated copy of URI on success
* \retval NULL on error
*/
const char *sip_route_add(struct sip_route *route, const char *uri, size_t len, int inserthead);
/*!
* \brief Add routes from header
*
* \note This procedure is for headers that require use of \<brackets\>.
*/
void sip_route_process_header(struct sip_route *route, const char *header, int inserthead);
/*!
* \brief copy route-set
*/
void sip_route_copy(struct sip_route *dst, const struct sip_route *src);
/*!
* \brief Free all routes in the list
*/
void sip_route_clear(struct sip_route *route);
/*!
* \brief Verbose dump of all hops for debugging
*/
void sip_route_dump(const struct sip_route *route);
/*!
* \brief Make the comma separated list of route hops
*
* \param route Source of route list
* \param formatcli Add's space after comma's, print's N/A if list is empty.
* \param skip Number of hops to skip
*
* \return an allocated struct ast_str on success
* \retval NULL on failure
*/
struct ast_str *sip_route_list(const struct sip_route *route, int formatcli, int skip)
__attribute__((__malloc__)) __attribute__((__warn_unused_result__));
/*!
* \brief Check if the route is strict
*
* \note The result is cached in route->type
*/
int sip_route_is_strict(struct sip_route *route);
/*!
* \brief Get the URI of the route's first hop
*/
const char *sip_route_first_uri(const struct sip_route *route);
/*!
* \brief Check if route has no URI's
*/
#define sip_route_empty(route) AST_LIST_EMPTY(&(route)->list)
#endif

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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2011, Digium, Inc.
*
* Michael L. Young <elgueromexicano@gmail.com>
*
* 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 Generate security events in the SIP channel
*
* \author Michael L. Young <elgueromexicano@gmail.com>
*/
#include "sip.h"
#ifndef _SIP_SECURITY_EVENTS_H
#define _SIP_SECURITY_EVENTS_H
void sip_report_invalid_peer(const struct sip_pvt *p);
void sip_report_failed_acl(const struct sip_pvt *p, const char *aclname);
void sip_report_inval_password(const struct sip_pvt *p, const char *responsechallenge, const char *responsehash);
void sip_report_auth_success(const struct sip_pvt *p, uint32_t using_password);
void sip_report_session_limit(const struct sip_pvt *p);
void sip_report_failed_challenge_response(const struct sip_pvt *p, const char *response, const char *expected_response);
void sip_report_chal_sent(const struct sip_pvt *p);
void sip_report_inval_transport(const struct sip_pvt *p, const char *transport);
void sip_digest_parser(char *c, struct digestkeys *keys);
int sip_report_security_event(const char *peer, struct ast_sockaddr *addr, const struct sip_pvt *p,
const struct sip_request *req, const int res);
#endif

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channels/sip/include/sip.h
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channels/sip/include/sip_utils.h
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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 utils header file
*/
#ifndef _SIP_UTILS_H
#define _SIP_UTILS_H
/* 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 converts ascii port to int representation.
*
* \arg pt[in] string that contains a port.
* \arg standard[in] port to return in case the port string input is NULL
* or if there is a parsing error.
*
* \return An integer port representation.
*/
unsigned int port_str2int(const char *pt, unsigned int standard);
/*! \brief Locate closing quote in a string, skipping escaped quotes.
* optionally with a limit on the search.
* start must be past the first quote.
*/
const char *find_closing_quote(const char *start, const char *lim);
/*! \brief Convert SIP hangup causes to Asterisk hangup causes */
int hangup_sip2cause(int cause);
/*! \brief Convert Asterisk hangup causes to SIP codes
\verbatim
Possible values from causes.h
AST_CAUSE_NOTDEFINED AST_CAUSE_NORMAL AST_CAUSE_BUSY
AST_CAUSE_FAILURE AST_CAUSE_CONGESTION AST_CAUSE_UNALLOCATED
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
---------------- ------------
1 unallocated number 404 Not Found
2 no route to network 404 Not found
3 no route to destination 404 Not found
16 normal call clearing --- (*)
17 user busy 486 Busy here
18 no user responding 408 Request Timeout
19 no answer from the user 480 Temporarily unavailable
20 subscriber absent 480 Temporarily unavailable
21 call rejected 403 Forbidden (+)
22 number changed (w/o diagnostic) 410 Gone
22 number changed (w/ diagnostic) 301 Moved Permanently
23 redirection to new destination 410 Gone
26 non-selected user clearing 404 Not Found (=)
27 destination out of order 502 Bad Gateway
28 address incomplete 484 Address incomplete
29 facility rejected 501 Not implemented
31 normal unspecified 480 Temporarily unavailable
\endverbatim
*/
const char *hangup_cause2sip(int cause);
/*! \brief Return a string describing the force_rport value for the given flags */
const char *force_rport_string(struct ast_flags *flags);
/*! \brief Return a string describing the comedia value for the given flags */
const char *comedia_string(struct ast_flags *flags);
#endif

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channels/sip/reqresp_parser.c
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channels/sip/route.c
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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2013, 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_route functions
*/
/*** MODULEINFO
<support_level>deprecated</support_level>
***/
#include "asterisk.h"
#include "asterisk/utils.h"
#include "include/route.h"
#include "include/reqresp_parser.h"
/*!
* \brief Traverse route hops
*/
#define sip_route_traverse(route,elem) AST_LIST_TRAVERSE(&(route)->list, elem, list)
#define sip_route_first(route) AST_LIST_FIRST(&(route)->list)
/*!
* \brief Structure to save a route hop
*/
struct sip_route_hop {
AST_LIST_ENTRY(sip_route_hop) list;
char uri[0];
};
const char *sip_route_add(struct sip_route *route, const char *uri, size_t len, int inserthead)
{
struct sip_route_hop *hop;
if (!uri || len < 1 || uri[0] == '\0') {
return NULL;
}
/* Expand len to include null terminator */
len++;
/* ast_calloc is not needed because all fields are initialized in this block */
hop = ast_malloc(sizeof(struct sip_route_hop) + len);
if (!hop) {
return NULL;
}
ast_copy_string(hop->uri, uri, len);
if (inserthead) {
AST_LIST_INSERT_HEAD(&route->list, hop, list);
route->type = route_invalidated;
} else {
if (sip_route_empty(route)) {
route->type = route_invalidated;
}
AST_LIST_INSERT_TAIL(&route->list, hop, list);
hop->list.next = NULL;
}
return hop->uri;
}
void sip_route_process_header(struct sip_route *route, const char *header, int inserthead)
{
const char *hop;
int len = 0;
const char *uri;
if (!route) {
ast_log(LOG_ERROR, "sip_route_process_header requires non-null route");
ast_do_crash();
return;
}
while (!get_in_brackets_const(header, &uri, &len)) {
header = strchr(header, ',');
if (header >= uri && header <= (uri + len)) {
/* comma inside brackets */
const char *next_br = strchr(header, '<');
if (next_br && next_br <= (uri + len)) {
header++;
continue;
}
continue;
}
if ((hop = sip_route_add(route, uri, len, inserthead))) {
ast_debug(2, "sip_route_process_header: <%s>\n", hop);
}
header = strchr(uri + len + 1, ',');
if (header == NULL) {
/* No more field-values, we're done with this header */
break;
}
/* Advance past comma */
header++;
}
}
void sip_route_copy(struct sip_route *dst, const struct sip_route *src)
{
struct sip_route_hop *hop;
/* make sure dst is empty */
sip_route_clear(dst);
sip_route_traverse(src, hop) {
const char *uri = sip_route_add(dst, hop->uri, strlen(hop->uri), 0);
if (uri) {
ast_debug(2, "sip_route_copy: copied hop: <%s>\n", uri);
}
}
dst->type = src->type;
}
void sip_route_clear(struct sip_route *route)
{
struct sip_route_hop *hop;
while ((hop = AST_LIST_REMOVE_HEAD(&route->list, list))) {
ast_free(hop);
}
route->type = route_loose;
}
void sip_route_dump(const struct sip_route *route)
{
if (sip_route_empty(route)) {
ast_verbose("sip_route_dump: no route/path\n");
} else {
struct sip_route_hop *hop;
sip_route_traverse(route, hop) {
ast_verbose("sip_route_dump: route/path hop: <%s>\n", hop->uri);
}
}
}
struct ast_str *sip_route_list(const struct sip_route *route, int formatcli, int skip)
{
struct sip_route_hop *hop;
const char *comma;
struct ast_str *buf;
int i = 0 - skip;
buf = ast_str_create(64);
if (!buf) {
return NULL;
}
comma = formatcli ? ", " : ",";
sip_route_traverse(route, hop) {
if (i >= 0) {
ast_str_append(&buf, 0, "%s<%s>", i ? comma : "", hop->uri);
}
i++;
}
if (formatcli && i <= 0) {
ast_str_append(&buf, 0, "N/A");
}
return buf;
}
int sip_route_is_strict(struct sip_route *route)
{
if (!route) {
return 0;
}
if (route->type == route_invalidated) {
struct sip_route_hop *hop = sip_route_first(route);
int ret = hop && (strstr(hop->uri, ";lr") == NULL);
route->type = ret ? route_strict : route_loose;
return ret;
}
return (route->type == route_strict) ? 1 : 0;
}
const char *sip_route_first_uri(const struct sip_route *route)
{
struct sip_route_hop *hop = sip_route_first(route);
return hop ? hop->uri : NULL;
}

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channels/sip/security_events.c
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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2012, Digium, Inc.
*
* Michael L. Young <elgueromexicano@gmail.com>
*
* 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 Generate security events in the SIP channel
*
* \author Michael L. Young <elgueromexicano@gmail.com>
*/
/*** MODULEINFO
<support_level>deprecated</support_level>
***/
#include "asterisk.h"
#include "include/sip.h"
#include "include/security_events.h"
/*! \brief Determine transport type used to receive request*/
static enum ast_transport security_event_get_transport(const struct sip_pvt *p)
{
return p->socket.type;
}
void sip_report_invalid_peer(const struct sip_pvt *p)
{
char session_id[32];
struct ast_security_event_inval_acct_id inval_acct_id = {
.common.event_type = AST_SECURITY_EVENT_INVAL_ACCT_ID,
.common.version = AST_SECURITY_EVENT_INVAL_ACCT_ID_VERSION,
.common.service = "SIP",
.common.account_id = p->exten,
.common.local_addr = {
.addr = &p->ourip,
.transport = security_event_get_transport(p)
},
.common.remote_addr = {
.addr = &p->sa,
.transport = security_event_get_transport(p)
},
.common.session_id = session_id,
};
snprintf(session_id, sizeof(session_id), "%p", p);
ast_security_event_report(AST_SEC_EVT(&inval_acct_id));
}
void sip_report_failed_acl(const struct sip_pvt *p, const char *aclname)
{
char session_id[32];
struct ast_security_event_failed_acl failed_acl_event = {
.common.event_type = AST_SECURITY_EVENT_FAILED_ACL,
.common.version = AST_SECURITY_EVENT_FAILED_ACL_VERSION,
.common.service = "SIP",
.common.account_id = p->exten,
.common.local_addr = {
.addr = &p->ourip,
.transport = security_event_get_transport(p)
},
.common.remote_addr = {
.addr = &p->sa,
.transport = security_event_get_transport(p)
},
.common.session_id = session_id,
.acl_name = aclname,
};
snprintf(session_id, sizeof(session_id), "%p", p);
ast_security_event_report(AST_SEC_EVT(&failed_acl_event));
}
void sip_report_inval_password(const struct sip_pvt *p, const char *response_challenge, const char *response_hash)
{
char session_id[32];
struct ast_security_event_inval_password inval_password = {
.common.event_type = AST_SECURITY_EVENT_INVAL_PASSWORD,
.common.version = AST_SECURITY_EVENT_INVAL_PASSWORD_VERSION,
.common.service = "SIP",
.common.account_id = p->exten,
.common.local_addr = {
.addr = &p->ourip,
.transport = security_event_get_transport(p)
},
.common.remote_addr = {
.addr = &p->sa,
.transport = security_event_get_transport(p)
},
.common.session_id = session_id,
.challenge = p->nonce,
.received_challenge = response_challenge,
.received_hash = response_hash,
};
snprintf(session_id, sizeof(session_id), "%p", p);
ast_security_event_report(AST_SEC_EVT(&inval_password));
}
void sip_report_auth_success(const struct sip_pvt *p, uint32_t using_password)
{
char session_id[32];
struct ast_security_event_successful_auth successful_auth = {
.common.event_type = AST_SECURITY_EVENT_SUCCESSFUL_AUTH,
.common.version = AST_SECURITY_EVENT_SUCCESSFUL_AUTH_VERSION,
.common.service = "SIP",
.common.account_id = p->exten,
.common.local_addr = {
.addr = &p->ourip,
.transport = security_event_get_transport(p)
},
.common.remote_addr = {
.addr = &p->sa,
.transport = security_event_get_transport(p)
},
.common.session_id = session_id,
.using_password = using_password,
};
snprintf(session_id, sizeof(session_id), "%p", p);
ast_security_event_report(AST_SEC_EVT(&successful_auth));
}
void sip_report_session_limit(const struct sip_pvt *p)
{
char session_id[32];
struct ast_security_event_session_limit session_limit = {
.common.event_type = AST_SECURITY_EVENT_SESSION_LIMIT,
.common.version = AST_SECURITY_EVENT_SESSION_LIMIT_VERSION,
.common.service = "SIP",
.common.account_id = p->exten,
.common.local_addr = {
.addr = &p->ourip,
.transport = security_event_get_transport(p)
},
.common.remote_addr = {
.addr = &p->sa,
.transport = security_event_get_transport(p)
},
.common.session_id = session_id,
};
snprintf(session_id, sizeof(session_id), "%p", p);
ast_security_event_report(AST_SEC_EVT(&session_limit));
}
void sip_report_failed_challenge_response(const struct sip_pvt *p, const char *response, const char *expected_response)
{
char session_id[32];
char account_id[256];
struct ast_security_event_chal_resp_failed chal_resp_failed = {
.common.event_type = AST_SECURITY_EVENT_CHAL_RESP_FAILED,
.common.version = AST_SECURITY_EVENT_CHAL_RESP_FAILED_VERSION,
.common.service = "SIP",
.common.account_id = account_id,
.common.local_addr = {
.addr = &p->ourip,
.transport = security_event_get_transport(p)
},
.common.remote_addr = {
.addr = &p->sa,
.transport = security_event_get_transport(p)
},
.common.session_id = session_id,
.challenge = p->nonce,
.response = response,
.expected_response = expected_response,
};
if (!ast_strlen_zero(p->from)) { /* When dialing, show account making call */
ast_copy_string(account_id, p->from, sizeof(account_id));
} else {
ast_copy_string(account_id, p->exten, sizeof(account_id));
}
snprintf(session_id, sizeof(session_id), "%p", p);
ast_security_event_report(AST_SEC_EVT(&chal_resp_failed));
}
void sip_report_chal_sent(const struct sip_pvt *p)
{
char session_id[32];
char account_id[256];
struct ast_security_event_chal_sent chal_sent = {
.common.event_type = AST_SECURITY_EVENT_CHAL_SENT,
.common.version = AST_SECURITY_EVENT_CHAL_SENT_VERSION,
.common.service = "SIP",
.common.account_id = account_id,
.common.local_addr = {
.addr = &p->ourip,
.transport = security_event_get_transport(p)
},
.common.remote_addr = {
.addr = &p->sa,
.transport = security_event_get_transport(p)
},
.common.session_id = session_id,
.challenge = p->nonce,
};
if (!ast_strlen_zero(p->from)) { /* When dialing, show account making call */
ast_copy_string(account_id, p->from, sizeof(account_id));
} else {
ast_copy_string(account_id, p->exten, sizeof(account_id));
}
snprintf(session_id, sizeof(session_id), "%p", p);
ast_security_event_report(AST_SEC_EVT(&chal_sent));
}
void sip_report_inval_transport(const struct sip_pvt *p, const char *transport)
{
char session_id[32];
struct ast_security_event_inval_transport inval_transport = {
.common.event_type = AST_SECURITY_EVENT_INVAL_TRANSPORT,
.common.version = AST_SECURITY_EVENT_INVAL_TRANSPORT_VERSION,
.common.service = "SIP",
.common.account_id = p->exten,
.common.local_addr = {
.addr = &p->ourip,
.transport = security_event_get_transport(p)
},
.common.remote_addr = {
.addr = &p->sa,
.transport = security_event_get_transport(p)
},
.common.session_id = session_id,
.transport = transport,
};
snprintf(session_id, sizeof(session_id), "%p", p);
ast_security_event_report(AST_SEC_EVT(&inval_transport));
}
int sip_report_security_event(const char *peer, struct ast_sockaddr *addr, const struct sip_pvt *p,
const struct sip_request *req, const int res)
{
struct sip_peer *peer_report;
enum check_auth_result res_report = res;
struct ast_str *buf;
char *c;
const char *authtoken;
char *reqheader, *respheader;
int result = 0;
char aclname[256];
struct digestkeys keys[] = {
[K_RESP] = { "response=", "" },
[K_URI] = { "uri=", "" },
[K_USER] = { "username=", "" },
[K_NONCE] = { "nonce=", "" },
[K_LAST] = { NULL, NULL}
};
peer_report = sip_find_peer(peer, addr, TRUE, FINDPEERS, FALSE, p->socket.type);
switch(res_report) {
case AUTH_DONT_KNOW:
break;
case AUTH_SUCCESSFUL:
if (peer_report) {
if (ast_strlen_zero(peer_report->secret) && ast_strlen_zero(peer_report->md5secret)) {
sip_report_auth_success(p, 0);
} else {
sip_report_auth_success(p, 1);
}
}
break;
case AUTH_CHALLENGE_SENT:
sip_report_chal_sent(p);
break;
case AUTH_SECRET_FAILED:
case AUTH_USERNAME_MISMATCH:
sip_auth_headers(WWW_AUTH, &respheader, &reqheader);
authtoken = sip_get_header(req, reqheader);
buf = ast_str_thread_get(&check_auth_buf, CHECK_AUTH_BUF_INITLEN);
ast_str_set(&buf, 0, "%s", authtoken);
c = ast_str_buffer(buf);
sip_digest_parser(c, keys);
if (res_report == AUTH_SECRET_FAILED) {
sip_report_inval_password(p, keys[K_NONCE].s, keys[K_RESP].s);
} else {
if (peer_report) {
sip_report_failed_challenge_response(p, keys[K_USER].s, peer_report->username);
}
}
break;
case AUTH_NOT_FOUND:
/* with sip_cfg.alwaysauthreject on, generates 2 events */
sip_report_invalid_peer(p);
break;
case AUTH_UNKNOWN_DOMAIN:
snprintf(aclname, sizeof(aclname), "domain_must_match");
sip_report_failed_acl(p, aclname);
break;
case AUTH_PEER_NOT_DYNAMIC:
snprintf(aclname, sizeof(aclname), "peer_not_dynamic");
sip_report_failed_acl(p, aclname);
break;
case AUTH_ACL_FAILED:
/* with sip_cfg.alwaysauthreject on, generates 2 events */
snprintf(aclname, sizeof(aclname), "device_must_match_acl");
sip_report_failed_acl(p, aclname);
break;
case AUTH_BAD_TRANSPORT:
sip_report_inval_transport(p, sip_get_transport(req->socket.type));
break;
case AUTH_RTP_FAILED:
break;
case AUTH_SESSION_LIMIT:
sip_report_session_limit(p);
break;
}
if (peer_report) {
sip_unref_peer(peer_report, "sip_report_security_event: sip_unref_peer: from handle_incoming");
}
return result;
}

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channels/sip/utils.c
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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2012, 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 Utility functions for chan_sip
*
* \author Terry Wilson <twilson@digium.com>
*/
/*** MODULEINFO
<support_level>deprecated</support_level>
***/
#include "asterisk.h"
#include "asterisk/utils.h"
#include "asterisk/cli.h"
#include "include/sip.h"
#include "include/sip_utils.h"
const char *force_rport_string(struct ast_flags *flags)
{
if (ast_test_flag(&flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) {
return ast_test_flag(&flags[0], SIP_NAT_FORCE_RPORT) ? "Auto (Yes)" : "Auto (No)";
}
return AST_CLI_YESNO(ast_test_flag(&flags[0], SIP_NAT_FORCE_RPORT));
}
const char *comedia_string(struct ast_flags *flags)
{
if (ast_test_flag(&flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA)) {
return ast_test_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP) ? "Auto (Yes)" : "Auto (No)";
}
return AST_CLI_YESNO(ast_test_flag(&flags[1], SIP_PAGE2_SYMMETRICRTP));
}