4202 lines
143 KiB
C
4202 lines
143 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2013, Digium, Inc.
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*
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* Mark Michelson <mmichelson@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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#ifndef _RES_PJSIP_H
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#define _RES_PJSIP_H
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#include <pjsip.h>
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/* Needed for SUBSCRIBE, NOTIFY, and PUBLISH method definitions */
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#include <pjsip_simple.h>
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#include <pjsip/sip_transaction.h>
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#include <pj/timer.h>
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/* Needed for pj_sockaddr */
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#include <pjlib.h>
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#include "asterisk/stringfields.h"
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/* Needed for struct ast_sockaddr */
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#include "asterisk/netsock2.h"
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/* Needed for linked list macros */
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#include "asterisk/linkedlists.h"
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/* Needed for ast_party_id */
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#include "asterisk/channel.h"
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/* Needed for ast_sorcery */
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#include "asterisk/sorcery.h"
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/* Needed for ast_dnsmgr */
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#include "asterisk/dnsmgr.h"
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/* Needed for ast_endpoint */
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#include "asterisk/endpoints.h"
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/* Needed for ast_t38_ec_modes */
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#include "asterisk/udptl.h"
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/* Needed for ast_rtp_dtls_cfg struct */
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#include "asterisk/rtp_engine.h"
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/* Needed for AST_VECTOR macro */
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#include "asterisk/vector.h"
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/* Needed for ast_sip_for_each_channel_snapshot struct */
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#include "asterisk/stasis_channels.h"
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#include "asterisk/stasis_endpoints.h"
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#include "asterisk/stream.h"
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#ifdef HAVE_PJSIP_TLS_TRANSPORT_RESTART
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/* Needed for knowing if the cert or priv key files changed */
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#include <sys/stat.h>
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#endif
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#define PJSIP_MINVERSION(m,n,p) (((m << 24) | (n << 16) | (p << 8)) >= PJ_VERSION_NUM)
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#ifndef PJSIP_EXPIRES_NOT_SPECIFIED
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/*
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* Added in pjproject 2.10.0. However define here if someone compiles against a
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* version of pjproject < 2.10.0.
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*
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* Usually defined in pjsip/include/pjsip/sip_msg.h (included as part of <pjsip.h>)
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*/
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#define PJSIP_EXPIRES_NOT_SPECIFIED ((pj_uint32_t)-1)
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#endif
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#define PJSTR_PRINTF_SPEC "%.*s"
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#define PJSTR_PRINTF_VAR(_v) ((int)(_v).slen), ((_v).ptr)
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#define AST_SIP_AUTH_MAX_REALM_LENGTH 255 /* From the auth/realm realtime column size */
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/* ":12345" */
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#define COLON_PORT_STRLEN 6
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/*
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* "<ipaddr>:<port>"
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* PJ_INET6_ADDRSTRLEN includes the NULL terminator
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*/
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#define IP6ADDR_COLON_PORT_BUFLEN (PJ_INET6_ADDRSTRLEN + COLON_PORT_STRLEN)
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/*!
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* \brief Fill a buffer with a pjsip transport's remote ip address and port
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*
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* \param _transport The pjsip_transport to use
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* \param _dest The destination buffer of at least IP6ADDR_COLON_PORT_BUFLEN bytes
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*/
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#define AST_SIP_MAKE_REMOTE_IPADDR_PORT_STR(_transport, _dest) \
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snprintf(_dest, IP6ADDR_COLON_PORT_BUFLEN, \
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PJSTR_PRINTF_SPEC ":%d", \
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PJSTR_PRINTF_VAR(_transport->remote_name.host), \
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_transport->remote_name.port);
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/* Forward declarations of PJSIP stuff */
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struct pjsip_rx_data;
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struct pjsip_module;
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struct pjsip_tx_data;
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struct pjsip_dialog;
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struct pjsip_transport;
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struct pjsip_tpfactory;
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struct pjsip_tls_setting;
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struct pjsip_tpselector;
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/*! \brief Maximum number of ciphers supported for a TLS transport */
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#define SIP_TLS_MAX_CIPHERS 64
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/*! Maximum number of challenges before assuming that we are in a loop */
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#define MAX_RX_CHALLENGES 10
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AST_VECTOR(ast_sip_service_route_vector, char *);
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static const pj_str_t AST_PJ_STR_EMPTY = { "", 0 };
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/*!
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* \brief Structure for SIP transport information
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*/
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struct ast_sip_transport_state {
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/*! \brief Transport itself */
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struct pjsip_transport *transport;
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/*! \brief Transport factory */
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struct pjsip_tpfactory *factory;
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/*!
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* Transport id
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* \since 13.8.0
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*/
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char *id;
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/*!
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* Transport type
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* \since 13.8.0
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*/
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enum ast_transport type;
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/*!
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* Address and port to bind to
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* \since 13.8.0
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*/
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pj_sockaddr host;
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/*!
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* TLS settings
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* \since 13.8.0
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*/
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pjsip_tls_setting tls;
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/*!
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* Configured TLS ciphers
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* \since 13.8.0
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*/
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pj_ssl_cipher ciphers[SIP_TLS_MAX_CIPHERS];
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/*!
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* Optional local network information, used for NAT purposes.
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* "deny" (set) means that it's in the local network. Use the
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* ast_sip_transport_is_nonlocal and ast_sip_transport_is_local
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* macro's.
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* \since 13.8.0
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*/
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struct ast_ha *localnet;
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/*!
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* DNS manager for refreshing the external signaling address
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* \since 13.8.0
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*/
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struct ast_dnsmgr_entry *external_signaling_address_refresher;
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/*!
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* Optional external signaling address information
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* \since 13.8.0
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*/
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struct ast_sockaddr external_signaling_address;
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/*!
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* DNS manager for refreshing the external media address
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* \since 13.18.0
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*/
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struct ast_dnsmgr_entry *external_media_address_refresher;
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/*!
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* Optional external signaling address information
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* \since 13.18.0
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*/
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struct ast_sockaddr external_media_address;
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/*!
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* Set when this transport is a flow of signaling to a target
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* \since 17.0.0
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*/
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int flow;
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/*!
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* The P-Preferred-Identity to use on traffic using this transport
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* \since 17.0.0
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*/
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char *preferred_identity;
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/*!
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* The Service Routes to use on traffic using this transport
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* \since 17.0.0
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*/
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struct ast_sip_service_route_vector *service_routes;
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/*!
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* Disregard RFC5922 7.2, and allow wildcard certs (TLS only)
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*/
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int allow_wildcard_certs;
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/*!
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* If true, fail if server certificate cannot verify (TLS only)
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*/
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int verify_server;
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#ifdef HAVE_PJSIP_TLS_TRANSPORT_RESTART
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/*!
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* The stats information for the certificate file, if configured
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*/
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struct stat cert_file_stat;
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/*!
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* The stats information for the private key file, if configured
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*/
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struct stat privkey_file_stat;
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#endif
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};
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#define ast_sip_transport_is_nonlocal(transport_state, addr) \
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(!transport_state->localnet || ast_apply_ha(transport_state->localnet, addr) == AST_SENSE_ALLOW)
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#define ast_sip_transport_is_local(transport_state, addr) \
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(transport_state->localnet && ast_apply_ha(transport_state->localnet, addr) != AST_SENSE_ALLOW)
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/*!
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* \brief Transport to bind to
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*/
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struct ast_sip_transport {
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/*! Sorcery object details */
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SORCERY_OBJECT(details);
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AST_DECLARE_STRING_FIELDS(
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/*! Certificate of authority list file */
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AST_STRING_FIELD(ca_list_file);
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/*! Certificate of authority list path */
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AST_STRING_FIELD(ca_list_path);
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/*! Public certificate file */
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AST_STRING_FIELD(cert_file);
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/*! Optional private key of the certificate file */
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AST_STRING_FIELD(privkey_file);
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/*! Password to open the private key */
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AST_STRING_FIELD(password);
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/*! External signaling address */
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AST_STRING_FIELD(external_signaling_address);
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/*! External media address */
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AST_STRING_FIELD(external_media_address);
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/*! Optional domain to use for messages if provided could not be found */
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AST_STRING_FIELD(domain);
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);
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/*! Type of transport */
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enum ast_transport type;
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/*!
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* \deprecated Moved to ast_sip_transport_state
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* \version 13.8.0 deprecated
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* Address and port to bind to
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*/
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pj_sockaddr host;
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/*! Number of simultaneous asynchronous operations */
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unsigned int async_operations;
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/*! Optional external port for signaling */
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unsigned int external_signaling_port;
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/*!
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* \deprecated Moved to ast_sip_transport_state
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* \version 13.7.1 deprecated
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* TLS settings
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*/
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pjsip_tls_setting tls;
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/*!
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* \deprecated Moved to ast_sip_transport_state
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* \version 13.7.1 deprecated
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* Configured TLS ciphers
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*/
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pj_ssl_cipher ciphers[SIP_TLS_MAX_CIPHERS];
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/*!
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* \deprecated Moved to ast_sip_transport_state
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* \version 13.7.1 deprecated
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* Optional local network information, used for NAT purposes
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*/
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struct ast_ha *localnet;
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/*!
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* \deprecated Moved to ast_sip_transport_state
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* \version 13.7.1 deprecated
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* DNS manager for refreshing the external address
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*/
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struct ast_dnsmgr_entry *external_address_refresher;
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/*!
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* \deprecated Moved to ast_sip_transport_state
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* \version 13.7.1 deprecated
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* Optional external address information
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*/
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struct ast_sockaddr external_address;
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/*!
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* \deprecated
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* \version 13.7.1 deprecated
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* Transport state information
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*/
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struct ast_sip_transport_state *state;
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/*! QOS DSCP TOS bits */
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unsigned int tos;
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/*! QOS COS value */
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unsigned int cos;
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/*! Write timeout */
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int write_timeout;
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/*! Allow reload */
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int allow_reload;
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/*! Automatically send requests out the same transport requests have come in on */
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int symmetric_transport;
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/*! This is a flow to another target */
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int flow;
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};
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#define SIP_SORCERY_DOMAIN_ALIAS_TYPE "domain_alias"
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/*!
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* Details about a SIP domain alias
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*/
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struct ast_sip_domain_alias {
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/*! Sorcery object details */
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SORCERY_OBJECT(details);
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AST_DECLARE_STRING_FIELDS(
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/*! Domain to be aliased to */
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AST_STRING_FIELD(domain);
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);
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};
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/*!
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* \brief Structure for SIP nat hook information
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*/
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struct ast_sip_nat_hook {
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/*! Sorcery object details */
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SORCERY_OBJECT(details);
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/*! Callback for when a message is going outside of our local network */
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void (*outgoing_external_message)(struct pjsip_tx_data *tdata, struct ast_sip_transport *transport);
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};
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/*! \brief Structure which contains information about a transport */
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struct ast_sip_request_transport_details {
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/*! \brief Type of transport */
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enum ast_transport type;
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/*! \brief Potential pointer to the transport itself, if UDP */
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pjsip_transport *transport;
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/*! \brief Potential pointer to the transport factory itself, if TCP/TLS */
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pjsip_tpfactory *factory;
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/*! \brief Local address for transport */
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pj_str_t local_address;
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/*! \brief Local port for transport */
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int local_port;
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};
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/*!
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* \brief The kind of security negotiation
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*/
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enum ast_sip_security_negotiation {
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/*! No security mechanism negotiation */
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AST_SIP_SECURITY_NEG_NONE = 0,
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/*! Use mediasec security mechanism negotiation */
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AST_SIP_SECURITY_NEG_MEDIASEC,
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/* Add RFC 3329 (sec-agree) mechanism negotiation in the future */
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};
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/*!
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* \brief The security mechanism type
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*/
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enum ast_sip_security_mechanism_type {
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AST_SIP_SECURITY_MECH_NONE = 0,
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/* Use msrp-tls as security mechanism */
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AST_SIP_SECURITY_MECH_MSRP_TLS,
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/* Use sdes-srtp as security mechanism */
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AST_SIP_SECURITY_MECH_SDES_SRTP,
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/* Use dtls-srtp as security mechanism */
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AST_SIP_SECURITY_MECH_DTLS_SRTP,
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/* Add RFC 3329 (sec-agree) mechanisms like tle, digest, ipsec-ike in the future */
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};
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/*!
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* \brief Structure representing a security mechanism as defined in RFC 3329
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*/
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struct ast_sip_security_mechanism {
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/* Used to determine which security mechanism to use. */
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enum ast_sip_security_mechanism_type type;
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/* The preference of this security mechanism. The higher the value, the more preferred. */
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float qvalue;
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/* Optional mechanism parameters. */
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struct ast_vector_string mechanism_parameters;
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};
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AST_VECTOR(ast_sip_security_mechanism_vector, struct ast_sip_security_mechanism *);
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/*!
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* \brief Contact associated with an address of record
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*/
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struct ast_sip_contact {
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/*! Sorcery object details, the id is the aor name plus a random string */
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SORCERY_OBJECT(details);
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AST_DECLARE_STRING_FIELDS(
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/*! Full URI of the contact */
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AST_STRING_FIELD(uri);
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/*! Outbound proxy to use for qualify */
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AST_STRING_FIELD(outbound_proxy);
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/*! Path information to place in Route headers */
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AST_STRING_FIELD(path);
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/*! Content of the User-Agent header in REGISTER request */
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AST_STRING_FIELD(user_agent);
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/*! The name of the aor this contact belongs to */
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AST_STRING_FIELD(aor);
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/*! Asterisk Server name */
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AST_STRING_FIELD(reg_server);
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/*! IP-address of the Via header in REGISTER request */
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AST_STRING_FIELD(via_addr);
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/*! Content of the Call-ID header in REGISTER request */
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AST_STRING_FIELD(call_id);
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/*! The name of the endpoint that added the contact */
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AST_STRING_FIELD(endpoint_name);
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);
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/*! Absolute time that this contact is no longer valid after */
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struct timeval expiration_time;
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/*! Frequency to send OPTIONS requests to contact. 0 is disabled. */
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unsigned int qualify_frequency;
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/*! If true authenticate the qualify challenge response if needed */
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int authenticate_qualify;
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/*! Qualify timeout. 0 is diabled. */
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double qualify_timeout;
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/*! Endpoint that added the contact, only available in observers */
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struct ast_sip_endpoint *endpoint;
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/*! Port of the Via header in REGISTER request */
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int via_port;
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/*! If true delete the contact on Asterisk restart/boot */
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int prune_on_boot;
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};
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/*!
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* \brief Status type for a contact.
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*/
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enum ast_sip_contact_status_type {
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/*! Frequency > 0, but no response from remote uri */
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UNAVAILABLE,
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/*! Frequency > 0, and got response from remote uri */
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AVAILABLE,
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/*! Default last status, and when a contact status object is not found */
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UNKNOWN,
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/*! Frequency == 0, has a contact, but don't know status (non-qualified) */
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CREATED,
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REMOVED,
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};
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/*!
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* \brief A contact's status.
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*
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* Maintains a contact's current status and round trip time if available.
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*/
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struct ast_sip_contact_status {
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AST_DECLARE_STRING_FIELDS(
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/*! The original contact's URI */
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AST_STRING_FIELD(uri);
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/*! The name of the aor this contact_status belongs to */
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AST_STRING_FIELD(aor);
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);
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/*! The round trip time in microseconds */
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int64_t rtt;
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/*!
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* The security mechanism list of the contact (RFC 3329).
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* Stores the values of Security-Server headers in 401/421/494 responses to an
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* in-dialog request or successful outbound registration which will be used to
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* set the Security-Verify headers of all subsequent requests to the contact.
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*/
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struct ast_sip_security_mechanism_vector security_mechanisms;
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/*! Current status for a contact (default - unavailable) */
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enum ast_sip_contact_status_type status;
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/*! Last status for a contact (default - unavailable) */
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enum ast_sip_contact_status_type last_status;
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/*! Name of the contact */
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char name[0];
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};
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/*!
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* \brief A SIP address of record
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*/
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struct ast_sip_aor {
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/*! Sorcery object details, the id is the AOR name */
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SORCERY_OBJECT(details);
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AST_DECLARE_STRING_FIELDS(
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/*! Voicemail boxes for this AOR */
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AST_STRING_FIELD(mailboxes);
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/*! Outbound proxy for OPTIONS requests */
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AST_STRING_FIELD(outbound_proxy);
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);
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/*! Minimum expiration time */
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unsigned int minimum_expiration;
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/*! Maximum expiration time */
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unsigned int maximum_expiration;
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/*! Default contact expiration if one is not provided in the contact */
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unsigned int default_expiration;
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/*! Frequency to send OPTIONS requests to AOR contacts. 0 is disabled. */
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unsigned int qualify_frequency;
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/*! If true authenticate the qualify challenge response if needed */
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int authenticate_qualify;
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/*! Maximum number of external contacts, 0 to disable */
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unsigned int max_contacts;
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/*! Whether to remove any existing contacts not related to an incoming REGISTER when it comes in */
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unsigned int remove_existing;
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/*! Any permanent configured contacts */
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struct ao2_container *permanent_contacts;
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/*! Determines whether SIP Path headers are supported */
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unsigned int support_path;
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/*! Qualify timeout. 0 is diabled. */
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double qualify_timeout;
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/*! Voicemail extension to set in Message-Account */
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char *voicemail_extension;
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/*! Whether to remove unavailable contacts over max_contacts at all or first if remove_existing is enabled */
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unsigned int remove_unavailable;
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};
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/*!
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* \brief A wrapper for contact that adds the aor_id and
|
|
* a consistent contact id. Used by ast_sip_for_each_contact.
|
|
*/
|
|
struct ast_sip_contact_wrapper {
|
|
/*! The id of the parent aor. */
|
|
char *aor_id;
|
|
/*! The id of contact in form of aor_id/contact_uri. */
|
|
char *contact_id;
|
|
/*! Pointer to the actual contact. */
|
|
struct ast_sip_contact *contact;
|
|
};
|
|
|
|
/*!
|
|
* \brief 100rel modes for SIP endpoints
|
|
*/
|
|
enum ast_sip_100rel_mode {
|
|
/*! Do not support 100rel. (no) */
|
|
AST_SIP_100REL_UNSUPPORTED = 0,
|
|
/*! As UAC, indicate 100rel support in Supported header. (yes) */
|
|
AST_SIP_100REL_SUPPORTED,
|
|
/*! As UAS, send 1xx responses reliably, if the UAC indicated its support. Otherwise same as AST_SIP_100REL_SUPPORTED. (peer_supported) */
|
|
AST_SIP_100REL_PEER_SUPPORTED,
|
|
/*! Require the use of 100rel. (required) */
|
|
AST_SIP_100REL_REQUIRED,
|
|
};
|
|
|
|
/*!
|
|
* \brief DTMF modes for SIP endpoints
|
|
*/
|
|
enum ast_sip_dtmf_mode {
|
|
/*! No DTMF to be used */
|
|
AST_SIP_DTMF_NONE,
|
|
/* XXX Should this be 2833 instead? */
|
|
/*! Use RFC 4733 events for DTMF */
|
|
AST_SIP_DTMF_RFC_4733,
|
|
/*! Use DTMF in the audio stream */
|
|
AST_SIP_DTMF_INBAND,
|
|
/*! Use SIP INFO DTMF (blech) */
|
|
AST_SIP_DTMF_INFO,
|
|
/*! Use SIP 4733 if supported by the other side or INBAND if not */
|
|
AST_SIP_DTMF_AUTO,
|
|
/*! Use SIP 4733 if supported by the other side or INFO DTMF (blech) if not */
|
|
AST_SIP_DTMF_AUTO_INFO,
|
|
};
|
|
|
|
/*!
|
|
* \brief Methods of storing SIP digest authentication credentials.
|
|
*
|
|
* Note that both methods result in MD5 digest authentication being
|
|
* used. The two methods simply alter how Asterisk determines the
|
|
* credentials for a SIP authentication
|
|
*/
|
|
enum ast_sip_auth_type {
|
|
/*! Credentials stored as a username and password combination */
|
|
AST_SIP_AUTH_TYPE_USER_PASS,
|
|
/*! Credentials stored as an MD5 sum */
|
|
AST_SIP_AUTH_TYPE_MD5,
|
|
/*! Google Oauth */
|
|
AST_SIP_AUTH_TYPE_GOOGLE_OAUTH,
|
|
/*! Credentials not stored this is a fake auth */
|
|
AST_SIP_AUTH_TYPE_ARTIFICIAL
|
|
};
|
|
|
|
#define SIP_SORCERY_AUTH_TYPE "auth"
|
|
|
|
struct ast_sip_auth {
|
|
/*! Sorcery ID of the auth is its name */
|
|
SORCERY_OBJECT(details);
|
|
AST_DECLARE_STRING_FIELDS(
|
|
/*! Identification for these credentials */
|
|
AST_STRING_FIELD(realm);
|
|
/*! Authentication username */
|
|
AST_STRING_FIELD(auth_user);
|
|
/*! Authentication password */
|
|
AST_STRING_FIELD(auth_pass);
|
|
/*! Authentication credentials in MD5 format (hash of user:realm:pass) */
|
|
AST_STRING_FIELD(md5_creds);
|
|
/*! Refresh token to use for OAuth authentication */
|
|
AST_STRING_FIELD(refresh_token);
|
|
/*! Client ID to use for OAuth authentication */
|
|
AST_STRING_FIELD(oauth_clientid);
|
|
/*! Secret to use for OAuth authentication */
|
|
AST_STRING_FIELD(oauth_secret);
|
|
);
|
|
/*! The time period (in seconds) that a nonce may be reused */
|
|
unsigned int nonce_lifetime;
|
|
/*! Used to determine what to use when authenticating */
|
|
enum ast_sip_auth_type type;
|
|
};
|
|
|
|
AST_VECTOR(ast_sip_auth_vector, const char *);
|
|
|
|
/*!
|
|
* \brief Different methods by which incoming requests can be matched to endpoints
|
|
*/
|
|
enum ast_sip_endpoint_identifier_type {
|
|
/*! Identify based on user name in From header */
|
|
AST_SIP_ENDPOINT_IDENTIFY_BY_USERNAME = (1 << 0),
|
|
/*! Identify based on user name in Auth header first, then From header */
|
|
AST_SIP_ENDPOINT_IDENTIFY_BY_AUTH_USERNAME = (1 << 1),
|
|
/*! Identify based on source IP address */
|
|
AST_SIP_ENDPOINT_IDENTIFY_BY_IP = (1 << 2),
|
|
/*! Identify based on arbitrary headers */
|
|
AST_SIP_ENDPOINT_IDENTIFY_BY_HEADER = (1 << 3),
|
|
};
|
|
AST_VECTOR(ast_sip_identify_by_vector, enum ast_sip_endpoint_identifier_type);
|
|
|
|
enum ast_sip_session_refresh_method {
|
|
/*! Use reinvite to negotiate direct media */
|
|
AST_SIP_SESSION_REFRESH_METHOD_INVITE,
|
|
/*! Use UPDATE to negotiate direct media */
|
|
AST_SIP_SESSION_REFRESH_METHOD_UPDATE,
|
|
};
|
|
|
|
enum ast_sip_direct_media_glare_mitigation {
|
|
/*! Take no special action to mitigate reinvite glare */
|
|
AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_NONE,
|
|
/*! Do not send an initial direct media session refresh on outgoing call legs
|
|
* Subsequent session refreshes will be sent no matter the session direction
|
|
*/
|
|
AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_OUTGOING,
|
|
/*! Do not send an initial direct media session refresh on incoming call legs
|
|
* Subsequent session refreshes will be sent no matter the session direction
|
|
*/
|
|
AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_INCOMING,
|
|
};
|
|
|
|
enum ast_sip_session_media_encryption {
|
|
/*! Invalid media encryption configuration */
|
|
AST_SIP_MEDIA_TRANSPORT_INVALID = 0,
|
|
/*! Do not allow any encryption of session media */
|
|
AST_SIP_MEDIA_ENCRYPT_NONE,
|
|
/*! Offer SDES-encrypted session media */
|
|
AST_SIP_MEDIA_ENCRYPT_SDES,
|
|
/*! Offer encrypted session media with datagram TLS key exchange */
|
|
AST_SIP_MEDIA_ENCRYPT_DTLS,
|
|
};
|
|
|
|
enum ast_sip_session_redirect {
|
|
/*! User portion of the target URI should be used as the target in the dialplan */
|
|
AST_SIP_REDIRECT_USER = 0,
|
|
/*! Target URI should be used as the target in the dialplan */
|
|
AST_SIP_REDIRECT_URI_CORE,
|
|
/*! Target URI should be used as the target within chan_pjsip itself */
|
|
AST_SIP_REDIRECT_URI_PJSIP,
|
|
};
|
|
|
|
/*!
|
|
* \brief Incoming/Outgoing call offer/answer joint codec preference.
|
|
*
|
|
* The default is INTERSECT ALL LOCAL.
|
|
*/
|
|
enum ast_sip_call_codec_pref {
|
|
/*! Two bits for merge */
|
|
/*! Intersection of local and remote */
|
|
AST_SIP_CALL_CODEC_PREF_INTERSECT = 1 << 0,
|
|
/*! Union of local and remote */
|
|
AST_SIP_CALL_CODEC_PREF_UNION = 1 << 1,
|
|
|
|
/*! Two bits for filter */
|
|
/*! No filter */
|
|
AST_SIP_CALL_CODEC_PREF_ALL = 1 << 2,
|
|
/*! Only the first */
|
|
AST_SIP_CALL_CODEC_PREF_FIRST = 1 << 3,
|
|
|
|
/*! Two bits for preference and sort */
|
|
/*! Prefer, and order by local values */
|
|
AST_SIP_CALL_CODEC_PREF_LOCAL = 1 << 4,
|
|
/*! Prefer, and order by remote values */
|
|
AST_SIP_CALL_CODEC_PREF_REMOTE = 1 << 5,
|
|
};
|
|
|
|
/*!
|
|
* \brief Returns true if the preference is set in the parameter
|
|
* \since 18.0.0
|
|
*
|
|
* \param __param A ast_flags struct with one or more of enum ast_sip_call_codec_pref set
|
|
* \param __codec_pref The last component of one of the enum values
|
|
* \retval 1 if the enum value is set
|
|
* \retval 0 if not
|
|
*/
|
|
#define ast_sip_call_codec_pref_test(__param, __codec_pref) (!!(ast_test_flag( &__param, AST_SIP_CALL_CODEC_PREF_ ## __codec_pref )))
|
|
|
|
/*!
|
|
* \brief Session timers options
|
|
*/
|
|
struct ast_sip_timer_options {
|
|
/*! Minimum session expiration period, in seconds */
|
|
unsigned int min_se;
|
|
/*! Session expiration period, in seconds */
|
|
unsigned int sess_expires;
|
|
};
|
|
|
|
/*!
|
|
* \brief Endpoint configuration for SIP extensions.
|
|
*
|
|
* SIP extensions, in this case refers to features
|
|
* indicated in Supported or Required headers.
|
|
*/
|
|
struct ast_sip_endpoint_extensions {
|
|
/*! Enabled SIP extensions */
|
|
unsigned int flags;
|
|
/*! Timer options */
|
|
struct ast_sip_timer_options timer;
|
|
};
|
|
|
|
/*!
|
|
* \brief Endpoint configuration for unsolicited MWI
|
|
*/
|
|
struct ast_sip_mwi_configuration {
|
|
AST_DECLARE_STRING_FIELDS(
|
|
/*! Configured voicemail boxes for this endpoint. Used for MWI */
|
|
AST_STRING_FIELD(mailboxes);
|
|
/*! Username to use when sending MWI NOTIFYs to this endpoint */
|
|
AST_STRING_FIELD(fromuser);
|
|
);
|
|
/*! Should mailbox states be combined into a single notification? */
|
|
unsigned int aggregate;
|
|
/*! Should a subscribe replace unsolicited notifies? */
|
|
unsigned int subscribe_replaces_unsolicited;
|
|
/*! Voicemail extension to set in Message-Account */
|
|
char *voicemail_extension;
|
|
};
|
|
|
|
/*!
|
|
* \brief Endpoint subscription configuration
|
|
*/
|
|
struct ast_sip_endpoint_subscription_configuration {
|
|
/*! Indicates if endpoint is allowed to initiate subscriptions */
|
|
unsigned int allow;
|
|
/*! The minimum allowed expiration for subscriptions from endpoint */
|
|
unsigned int minexpiry;
|
|
/*! Message waiting configuration */
|
|
struct ast_sip_mwi_configuration mwi;
|
|
/*! Context for SUBSCRIBE requests */
|
|
char context[AST_MAX_CONTEXT];
|
|
};
|
|
|
|
/*!
|
|
* \brief NAT configuration options for endpoints
|
|
*/
|
|
struct ast_sip_endpoint_nat_configuration {
|
|
/*! Whether to force using the source IP address/port for sending responses */
|
|
unsigned int force_rport;
|
|
/*! Whether to rewrite the Contact header with the source IP address/port or not */
|
|
unsigned int rewrite_contact;
|
|
};
|
|
|
|
/*!
|
|
* \brief Party identification options for endpoints
|
|
*
|
|
* This includes caller ID, connected line, and redirecting-related options
|
|
*/
|
|
struct ast_sip_endpoint_id_configuration {
|
|
struct ast_party_id self;
|
|
/*! Do we accept identification information from this endpoint */
|
|
unsigned int trust_inbound;
|
|
/*! Do we send private identification information to this endpoint? */
|
|
unsigned int trust_outbound;
|
|
/*! Do we send P-Asserted-Identity headers to this endpoint? */
|
|
unsigned int send_pai;
|
|
/*! Do we send Remote-Party-ID headers to this endpoint? */
|
|
unsigned int send_rpid;
|
|
/*! Do we send messages for connected line updates for unanswered incoming calls immediately to this endpoint? */
|
|
unsigned int rpid_immediate;
|
|
/*! Do we add Diversion headers to applicable outgoing requests/responses? */
|
|
unsigned int send_diversion;
|
|
/*! Do we accept connected line updates from this endpoint? */
|
|
unsigned int trust_connected_line;
|
|
/*! Do we send connected line updates to this endpoint? */
|
|
unsigned int send_connected_line;
|
|
/*! When performing connected line update, which method should be used */
|
|
enum ast_sip_session_refresh_method refresh_method;
|
|
/*! Do we add History-Info headers to applicable outgoing requests/responses? */
|
|
unsigned int send_history_info;
|
|
};
|
|
|
|
/*!
|
|
* \brief Call pickup configuration options for endpoints
|
|
*/
|
|
struct ast_sip_endpoint_pickup_configuration {
|
|
/*! Call group */
|
|
ast_group_t callgroup;
|
|
/*! Pickup group */
|
|
ast_group_t pickupgroup;
|
|
/*! Named call group */
|
|
struct ast_namedgroups *named_callgroups;
|
|
/*! Named pickup group */
|
|
struct ast_namedgroups *named_pickupgroups;
|
|
};
|
|
|
|
/*!
|
|
* \brief Configuration for one-touch INFO recording
|
|
*/
|
|
struct ast_sip_info_recording_configuration {
|
|
AST_DECLARE_STRING_FIELDS(
|
|
/*! Feature to enact when one-touch recording INFO with Record: On is received */
|
|
AST_STRING_FIELD(onfeature);
|
|
/*! Feature to enact when one-touch recording INFO with Record: Off is received */
|
|
AST_STRING_FIELD(offfeature);
|
|
);
|
|
/*! Is one-touch recording permitted? */
|
|
unsigned int enabled;
|
|
};
|
|
|
|
/*!
|
|
* \brief Endpoint configuration options for INFO packages
|
|
*/
|
|
struct ast_sip_endpoint_info_configuration {
|
|
/*! Configuration for one-touch recording */
|
|
struct ast_sip_info_recording_configuration recording;
|
|
};
|
|
|
|
/*!
|
|
* \brief RTP configuration for SIP endpoints
|
|
*/
|
|
struct ast_sip_media_rtp_configuration {
|
|
AST_DECLARE_STRING_FIELDS(
|
|
/*! Configured RTP engine for this endpoint. */
|
|
AST_STRING_FIELD(engine);
|
|
);
|
|
/*! Whether IPv6 RTP is enabled or not */
|
|
unsigned int ipv6;
|
|
/*! Whether symmetric RTP is enabled or not */
|
|
unsigned int symmetric;
|
|
/*! Whether ICE support is enabled or not */
|
|
unsigned int ice_support;
|
|
/*! Whether to use the "ptime" attribute received from the endpoint or not */
|
|
unsigned int use_ptime;
|
|
/*! Do we use AVPF exclusively for this endpoint? */
|
|
unsigned int use_avpf;
|
|
/*! Do we force AVP, AVPF, SAVP, or SAVPF even for DTLS media streams? */
|
|
unsigned int force_avp;
|
|
/*! Do we use the received media transport in our answer SDP */
|
|
unsigned int use_received_transport;
|
|
/*! \brief DTLS-SRTP configuration information */
|
|
struct ast_rtp_dtls_cfg dtls_cfg;
|
|
/*! Should SRTP use a 32 byte tag instead of an 80 byte tag? */
|
|
unsigned int srtp_tag_32;
|
|
/*! Do we use media encryption? what type? */
|
|
enum ast_sip_session_media_encryption encryption;
|
|
/*! Do we want to optimistically support encryption if possible? */
|
|
unsigned int encryption_optimistic;
|
|
/*! Number of seconds between RTP keepalive packets */
|
|
unsigned int keepalive;
|
|
/*! Number of seconds before terminating channel due to lack of RTP (when not on hold) */
|
|
unsigned int timeout;
|
|
/*! Number of seconds before terminating channel due to lack of RTP (when on hold) */
|
|
unsigned int timeout_hold;
|
|
/*! Follow forked media with a different To tag */
|
|
unsigned int follow_early_media_fork;
|
|
/*! Accept updated SDPs on non-100rel 18X and 2XX responses with the same To tag */
|
|
unsigned int accept_multiple_sdp_answers;
|
|
};
|
|
|
|
/*!
|
|
* \brief Direct media options for SIP endpoints
|
|
*/
|
|
struct ast_sip_direct_media_configuration {
|
|
/*! Boolean indicating if direct_media is permissible */
|
|
unsigned int enabled;
|
|
/*! When using direct media, which method should be used */
|
|
enum ast_sip_session_refresh_method method;
|
|
/*! Take steps to mitigate glare for direct media */
|
|
enum ast_sip_direct_media_glare_mitigation glare_mitigation;
|
|
/*! Do not attempt direct media session refreshes if a media NAT is detected */
|
|
unsigned int disable_on_nat;
|
|
};
|
|
|
|
struct ast_sip_t38_configuration {
|
|
/*! Whether T.38 UDPTL support is enabled or not */
|
|
unsigned int enabled;
|
|
/*! Error correction setting for T.38 UDPTL */
|
|
enum ast_t38_ec_modes error_correction;
|
|
/*! Explicit T.38 max datagram value, may be 0 to indicate the remote side can be trusted */
|
|
unsigned int maxdatagram;
|
|
/*! Whether NAT Support is enabled for T.38 UDPTL sessions or not */
|
|
unsigned int nat;
|
|
/*! Whether to use IPv6 for UDPTL or not */
|
|
unsigned int ipv6;
|
|
/*! Bind the UDPTL instance to the media_address */
|
|
unsigned int bind_udptl_to_media_address;
|
|
};
|
|
|
|
/*!
|
|
* \brief Media configuration for SIP endpoints
|
|
*/
|
|
struct ast_sip_endpoint_media_configuration {
|
|
AST_DECLARE_STRING_FIELDS(
|
|
/*! Optional media address to use in SDP */
|
|
AST_STRING_FIELD(address);
|
|
/*! SDP origin username */
|
|
AST_STRING_FIELD(sdpowner);
|
|
/*! SDP session name */
|
|
AST_STRING_FIELD(sdpsession);
|
|
);
|
|
/*! RTP media configuration */
|
|
struct ast_sip_media_rtp_configuration rtp;
|
|
/*! Direct media options */
|
|
struct ast_sip_direct_media_configuration direct_media;
|
|
/*! T.38 (FoIP) options */
|
|
struct ast_sip_t38_configuration t38;
|
|
/*! Configured codecs */
|
|
struct ast_format_cap *codecs;
|
|
/*! Capabilities in topology form */
|
|
struct ast_stream_topology *topology;
|
|
/*! DSCP TOS bits for audio streams */
|
|
unsigned int tos_audio;
|
|
/*! Priority for audio streams */
|
|
unsigned int cos_audio;
|
|
/*! DSCP TOS bits for video streams */
|
|
unsigned int tos_video;
|
|
/*! Priority for video streams */
|
|
unsigned int cos_video;
|
|
/*! Is g.726 packed in a non standard way */
|
|
unsigned int g726_non_standard;
|
|
/*! Bind the RTP instance to the media_address */
|
|
unsigned int bind_rtp_to_media_address;
|
|
/*! Use RTCP-MUX */
|
|
unsigned int rtcp_mux;
|
|
/*! Maximum number of audio streams to offer/accept */
|
|
unsigned int max_audio_streams;
|
|
/*! Maximum number of video streams to offer/accept */
|
|
unsigned int max_video_streams;
|
|
/*! Use BUNDLE */
|
|
unsigned int bundle;
|
|
/*! Enable webrtc settings and defaults */
|
|
unsigned int webrtc;
|
|
/*! Codec preference for an incoming offer */
|
|
struct ast_flags incoming_call_offer_pref;
|
|
/*! Codec preference for an outgoing offer */
|
|
struct ast_flags outgoing_call_offer_pref;
|
|
/*! Codec negotiation prefs for incoming offers */
|
|
struct ast_stream_codec_negotiation_prefs codec_prefs_incoming_offer;
|
|
/*! Codec negotiation prefs for outgoing offers */
|
|
struct ast_stream_codec_negotiation_prefs codec_prefs_outgoing_offer;
|
|
/*! Codec negotiation prefs for incoming answers */
|
|
struct ast_stream_codec_negotiation_prefs codec_prefs_incoming_answer;
|
|
/*! Codec negotiation prefs for outgoing answers */
|
|
struct ast_stream_codec_negotiation_prefs codec_prefs_outgoing_answer;
|
|
};
|
|
|
|
/*!
|
|
* \brief An entity with which Asterisk communicates
|
|
*/
|
|
struct ast_sip_endpoint {
|
|
SORCERY_OBJECT(details);
|
|
AST_DECLARE_STRING_FIELDS(
|
|
/*! Context to send incoming calls to */
|
|
AST_STRING_FIELD(context);
|
|
/*! Name of an explicit transport to use */
|
|
AST_STRING_FIELD(transport);
|
|
/*! Outbound proxy to use */
|
|
AST_STRING_FIELD(outbound_proxy);
|
|
/*! Explicit AORs to dial if none are specified */
|
|
AST_STRING_FIELD(aors);
|
|
/*! Musiconhold class to suggest that the other side use when placing on hold */
|
|
AST_STRING_FIELD(mohsuggest);
|
|
/*! Configured tone zone for this endpoint. */
|
|
AST_STRING_FIELD(zone);
|
|
/*! Configured language for this endpoint. */
|
|
AST_STRING_FIELD(language);
|
|
/*! Default username to place in From header */
|
|
AST_STRING_FIELD(fromuser);
|
|
/*! Domain to place in From header */
|
|
AST_STRING_FIELD(fromdomain);
|
|
/*! Context to route incoming MESSAGE requests to */
|
|
AST_STRING_FIELD(message_context);
|
|
/*! Accountcode to auto-set on channels */
|
|
AST_STRING_FIELD(accountcode);
|
|
/*! If set, we'll push incoming MWI NOTIFYs to stasis using this mailbox */
|
|
AST_STRING_FIELD(incoming_mwi_mailbox);
|
|
/*! STIR/SHAKEN profile to use */
|
|
AST_STRING_FIELD(stir_shaken_profile);
|
|
);
|
|
/*! Configuration for extensions */
|
|
struct ast_sip_endpoint_extensions extensions;
|
|
/*! Configuration relating to media */
|
|
struct ast_sip_endpoint_media_configuration media;
|
|
/*! SUBSCRIBE/NOTIFY configuration options */
|
|
struct ast_sip_endpoint_subscription_configuration subscription;
|
|
/*! NAT configuration */
|
|
struct ast_sip_endpoint_nat_configuration nat;
|
|
/*! Party identification options */
|
|
struct ast_sip_endpoint_id_configuration id;
|
|
/*! Configuration options for INFO packages */
|
|
struct ast_sip_endpoint_info_configuration info;
|
|
/*! Call pickup configuration */
|
|
struct ast_sip_endpoint_pickup_configuration pickup;
|
|
/*! Inbound authentication credentials */
|
|
struct ast_sip_auth_vector inbound_auths;
|
|
/*! Outbound authentication credentials */
|
|
struct ast_sip_auth_vector outbound_auths;
|
|
/*! DTMF mode to use with this endpoint */
|
|
enum ast_sip_dtmf_mode dtmf;
|
|
/*! Method(s) by which the endpoint should be identified. */
|
|
enum ast_sip_endpoint_identifier_type ident_method;
|
|
/*! Order of the method(s) by which the endpoint should be identified. */
|
|
struct ast_sip_identify_by_vector ident_method_order;
|
|
/*! Boolean indicating if ringing should be sent as inband progress */
|
|
unsigned int inband_progress;
|
|
/*! Pointer to the persistent Asterisk endpoint */
|
|
struct ast_endpoint *persistent;
|
|
/*! The number of channels at which busy device state is returned */
|
|
unsigned int devicestate_busy_at;
|
|
/*! Whether fax detection is enabled or not (CNG tone detection) */
|
|
unsigned int faxdetect;
|
|
/*! Determines if transfers (using REFER) are allowed by this endpoint */
|
|
unsigned int allowtransfer;
|
|
/*! Method used when handling redirects */
|
|
enum ast_sip_session_redirect redirect_method;
|
|
/*! Variables set on channel creation */
|
|
struct ast_variable *channel_vars;
|
|
/*! Whether to place a 'user=phone' parameter into the request URI if user is a number */
|
|
unsigned int usereqphone;
|
|
/*! Whether to pass through hold and unhold using re-invites with recvonly and sendrecv */
|
|
unsigned int moh_passthrough;
|
|
/*! Access control list */
|
|
struct ast_acl_list *acl;
|
|
/*! Restrict what IPs are allowed in the Contact header (for registration) */
|
|
struct ast_acl_list *contact_acl;
|
|
/*! The number of seconds into call to disable fax detection. (0 = disabled) */
|
|
unsigned int faxdetect_timeout;
|
|
/*! Override the user on the outgoing Contact header with this value. */
|
|
char *contact_user;
|
|
/*! Whether to response SDP offer with single most preferred codec. */
|
|
unsigned int preferred_codec_only;
|
|
/*! Do we allow an asymmetric RTP codec? */
|
|
unsigned int asymmetric_rtp_codec;
|
|
/*! Do we allow overlap dialling? */
|
|
unsigned int allow_overlap;
|
|
/*! Whether to notifies all the progress details on blind transfer */
|
|
unsigned int refer_blind_progress;
|
|
/*! Whether to notifies dialog-info 'early' on INUSE && RINGING state */
|
|
unsigned int notify_early_inuse_ringing;
|
|
/*! Suppress Q.850 Reason headers on this endpoint */
|
|
unsigned int suppress_q850_reason_headers;
|
|
/*! Ignore 183 if no SDP is present */
|
|
unsigned int ignore_183_without_sdp;
|
|
/*! Type of security negotiation to use (RFC 3329). */
|
|
enum ast_sip_security_negotiation security_negotiation;
|
|
/*! Client security mechanisms (RFC 3329). */
|
|
struct ast_sip_security_mechanism_vector security_mechanisms;
|
|
/*! Set which STIR/SHAKEN behaviors we want on this endpoint */
|
|
unsigned int stir_shaken;
|
|
/*! Should we authenticate OPTIONS requests per RFC 3261? */
|
|
unsigned int allow_unauthenticated_options;
|
|
/*! The name of the geoloc profile to apply when Asterisk receives a call from this endpoint */
|
|
AST_STRING_FIELD_EXTENDED(geoloc_incoming_call_profile);
|
|
/*! The name of the geoloc profile to apply when Asterisk sends a call to this endpoint */
|
|
AST_STRING_FIELD_EXTENDED(geoloc_outgoing_call_profile);
|
|
/*! The context to use for overlap dialing, if different from the endpoint's context */
|
|
AST_STRING_FIELD_EXTENDED(overlap_context);
|
|
/*! 100rel mode to use with this endpoint */
|
|
enum ast_sip_100rel_mode rel100;
|
|
/*! Send Advice-of-Charge messages */
|
|
unsigned int send_aoc;
|
|
};
|
|
|
|
/*! URI parameter for symmetric transport */
|
|
#define AST_SIP_X_AST_TXP "x-ast-txp"
|
|
#define AST_SIP_X_AST_TXP_LEN 9
|
|
|
|
/*! Common media types used throughout res_pjsip and pjproject */
|
|
extern pjsip_media_type pjsip_media_type_application_json;
|
|
extern pjsip_media_type pjsip_media_type_application_media_control_xml;
|
|
extern pjsip_media_type pjsip_media_type_application_pidf_xml;
|
|
extern pjsip_media_type pjsip_media_type_application_xpidf_xml;
|
|
extern pjsip_media_type pjsip_media_type_application_cpim_xpidf_xml;
|
|
extern pjsip_media_type pjsip_media_type_application_rlmi_xml;
|
|
extern pjsip_media_type pjsip_media_type_application_simple_message_summary;
|
|
extern pjsip_media_type pjsip_media_type_application_sdp;
|
|
extern pjsip_media_type pjsip_media_type_multipart_alternative;
|
|
extern pjsip_media_type pjsip_media_type_multipart_mixed;
|
|
extern pjsip_media_type pjsip_media_type_multipart_related;
|
|
extern pjsip_media_type pjsip_media_type_text_plain;
|
|
|
|
/*!
|
|
* \brief Compare pjsip media types
|
|
*
|
|
* \param a the first media type
|
|
* \param b the second media type
|
|
* \retval 1 Media types are equal
|
|
* \retval 0 Media types are not equal
|
|
*/
|
|
int ast_sip_are_media_types_equal(pjsip_media_type *a, pjsip_media_type *b);
|
|
|
|
/*!
|
|
* \brief Check if a media type is in a list of others
|
|
*
|
|
* \param a pjsip_media_type to search for
|
|
* \param ... one or more pointers to pjsip_media_types the last of which must be "SENTINEL"
|
|
* \retval 1 Media types are equal
|
|
* \retval 0 Media types are not equal
|
|
*/
|
|
int ast_sip_is_media_type_in(pjsip_media_type *a, ...) attribute_sentinel;
|
|
|
|
/*!
|
|
* \brief Add security headers to transmission data
|
|
*
|
|
* \param security_mechanisms Vector of security mechanisms.
|
|
* \param header_name The header name under which to add the security mechanisms.
|
|
* One of Security-Client, Security-Server, Security-Verify.
|
|
* \param add_qval If zero, don't add the q-value to the header.
|
|
* \param tdata The transmission data.
|
|
* \retval 0 Success
|
|
* \retval non-zero Failure
|
|
*/
|
|
int ast_sip_add_security_headers(struct ast_sip_security_mechanism_vector *security_mechanisms,
|
|
const char *header_name, int add_qval, pjsip_tx_data *tdata);
|
|
|
|
/*!
|
|
* \brief Append to security mechanism vector from SIP header
|
|
*
|
|
* \param hdr The header of the security mechanisms.
|
|
* \param security_mechanisms Vector of security mechanisms to append to.
|
|
* Header name must be one of Security-Client, Security-Server, Security-Verify.
|
|
*/
|
|
void ast_sip_header_to_security_mechanism(const pjsip_generic_string_hdr *hdr,
|
|
struct ast_sip_security_mechanism_vector *security_mechanisms);
|
|
|
|
/*!
|
|
* \brief Initialize security mechanism vector from string of security mechanisms.
|
|
*
|
|
* \param security_mechanism Pointer to vector of security mechanisms to initialize.
|
|
* \param value String of security mechanisms as defined in RFC 3329.
|
|
* \retval 0 Success
|
|
* \retval non-zero Failure
|
|
*/
|
|
int ast_sip_security_mechanism_vector_init(struct ast_sip_security_mechanism_vector *security_mechanism, const char *value);
|
|
|
|
/*!
|
|
* \brief Removes all headers of a specific name and value from a pjsip_msg.
|
|
*
|
|
* \param msg PJSIP message from which to remove headers.
|
|
* \param hdr_name Name of the header to remove.
|
|
* \param value Optional string value of the header to remove.
|
|
* If NULL, remove all headers of given hdr_name.
|
|
*/
|
|
void ast_sip_remove_headers_by_name_and_value(pjsip_msg *msg, const pj_str_t *hdr_name, const char* value);
|
|
|
|
/*!
|
|
* \brief Duplicate a security mechanism.
|
|
*
|
|
* \param dst Security mechanism to duplicate to.
|
|
* \param src Security mechanism to duplicate.
|
|
*/
|
|
void ast_sip_security_mechanisms_vector_copy(struct ast_sip_security_mechanism_vector *dst,
|
|
const struct ast_sip_security_mechanism_vector *src);
|
|
|
|
/*!
|
|
* \brief Free contents of a security mechanism vector.
|
|
*
|
|
* \param security_mechanisms Vector whose contents are to be freed
|
|
*/
|
|
void ast_sip_security_mechanisms_vector_destroy(struct ast_sip_security_mechanism_vector *security_mechanisms);
|
|
|
|
/*!
|
|
* \brief Allocate a security mechanism from a string.
|
|
*
|
|
* \param security_mechanism Pointer-pointer to the security mechanism to allocate.
|
|
* \param value The security mechanism string as defined in RFC 3329 (section 2.2)
|
|
* in the form <mechanism_name>;q=<q_value>;<mechanism_parameters>
|
|
* \retval 0 Success
|
|
* \retval non-zero Failure
|
|
*/
|
|
int ast_sip_str_to_security_mechanism(struct ast_sip_security_mechanism **security_mechanism, const char *value);
|
|
|
|
/*!
|
|
* \brief Writes the security mechanisms of an endpoint into a buffer as a string and returns the buffer.
|
|
*
|
|
* \note The buffer must be freed by the caller.
|
|
*
|
|
* \param endpoint Pointer to endpoint.
|
|
* \param add_qvalue If non-zero, the q-value is printed as well
|
|
* \param buf The buffer to write the string into
|
|
* \retval 0 Success
|
|
* \retval non-zero Failure
|
|
*/
|
|
int ast_sip_security_mechanisms_to_str(const struct ast_sip_security_mechanism_vector *security_mechanisms, int add_qvalue, char **buf);
|
|
|
|
/*!
|
|
* \brief Set the security negotiation based on a given string.
|
|
*
|
|
* \param security_negotiation Security negotiation enum to set.
|
|
* \param val String that represents a security_negotiation value.
|
|
* \retval 0 Success
|
|
* \retval non-zero Failure
|
|
*/
|
|
int ast_sip_set_security_negotiation(enum ast_sip_security_negotiation *security_negotiation, const char *val);
|
|
|
|
/*!
|
|
* \brief Initialize an auth vector with the configured values.
|
|
*
|
|
* \param vector Vector to initialize
|
|
* \param auth_names Comma-separated list of names to set in the array
|
|
* \retval 0 Success
|
|
* \retval non-zero Failure
|
|
*/
|
|
int ast_sip_auth_vector_init(struct ast_sip_auth_vector *vector, const char *auth_names);
|
|
|
|
/*!
|
|
* \brief Free contents of an auth vector.
|
|
*
|
|
* \param vector Vector whose contents are to be freed
|
|
*/
|
|
void ast_sip_auth_vector_destroy(struct ast_sip_auth_vector *vector);
|
|
|
|
/*!
|
|
* \brief Possible returns from ast_sip_check_authentication
|
|
*/
|
|
enum ast_sip_check_auth_result {
|
|
/*! Authentication needs to be challenged */
|
|
AST_SIP_AUTHENTICATION_CHALLENGE,
|
|
/*! Authentication succeeded */
|
|
AST_SIP_AUTHENTICATION_SUCCESS,
|
|
/*! Authentication failed */
|
|
AST_SIP_AUTHENTICATION_FAILED,
|
|
/*! Authentication encountered some internal error */
|
|
AST_SIP_AUTHENTICATION_ERROR,
|
|
};
|
|
|
|
/*!
|
|
* \brief An interchangeable way of handling digest authentication for SIP.
|
|
*
|
|
* An authenticator is responsible for filling in the callbacks provided below. Each is called from a publicly available
|
|
* function in res_sip. The authenticator can use configuration or other local policy to determine whether authentication
|
|
* should take place and what credentials should be used when challenging and authenticating a request.
|
|
*/
|
|
struct ast_sip_authenticator {
|
|
/*!
|
|
* \brief Check if a request requires authentication
|
|
* See ast_sip_requires_authentication for more details
|
|
*/
|
|
int (*requires_authentication)(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata);
|
|
/*!
|
|
* \brief Check that an incoming request passes authentication.
|
|
*
|
|
* The tdata parameter is useful for adding information such as digest challenges.
|
|
*
|
|
* \param endpoint The endpoint sending the incoming request
|
|
* \param rdata The incoming request
|
|
* \param tdata Tentative outgoing request.
|
|
*/
|
|
enum ast_sip_check_auth_result (*check_authentication)(struct ast_sip_endpoint *endpoint,
|
|
pjsip_rx_data *rdata, pjsip_tx_data *tdata);
|
|
};
|
|
|
|
/*!
|
|
* \brief an interchangeable way of responding to authentication challenges
|
|
*
|
|
* An outbound authenticator takes incoming challenges and formulates a new SIP request with
|
|
* credentials.
|
|
*/
|
|
struct ast_sip_outbound_authenticator {
|
|
/*!
|
|
* \brief Create a new request with authentication credentials
|
|
*
|
|
* \param auths A vector of IDs of auth sorcery objects
|
|
* \param challenge The SIP response with authentication challenge(s)
|
|
* \param old_request The request that received the auth challenge(s)
|
|
* \param new_request The new SIP request with challenge response(s)
|
|
* \retval 0 Successfully created new request
|
|
* \retval -1 Failed to create a new request
|
|
*/
|
|
int (*create_request_with_auth)(const struct ast_sip_auth_vector *auths, struct pjsip_rx_data *challenge,
|
|
struct pjsip_tx_data *old_request, struct pjsip_tx_data **new_request);
|
|
};
|
|
|
|
/*!
|
|
* \brief An entity responsible for identifying the source of a SIP message
|
|
*/
|
|
struct ast_sip_endpoint_identifier {
|
|
/*!
|
|
* \brief Callback used to identify the source of a message.
|
|
* See ast_sip_identify_endpoint for more details
|
|
*/
|
|
struct ast_sip_endpoint *(*identify_endpoint)(pjsip_rx_data *rdata);
|
|
};
|
|
|
|
/*!
|
|
* \brief Contact retrieval filtering flags
|
|
*/
|
|
enum ast_sip_contact_filter {
|
|
/*! \brief Default filter flags */
|
|
AST_SIP_CONTACT_FILTER_DEFAULT = 0,
|
|
|
|
/*! \brief Return only reachable or unknown contacts */
|
|
AST_SIP_CONTACT_FILTER_REACHABLE = (1 << 0),
|
|
};
|
|
|
|
/*!
|
|
* \brief Adds a Date header to the tdata, formatted like:
|
|
* Date: Wed, 01 Jan 2021 14:53:01 GMT
|
|
* \since 16.19.0
|
|
*
|
|
* \note There is no checking done to see if the header already exists
|
|
* before adding it. It's up to the caller of this function to determine
|
|
* if that needs to be done or not.
|
|
*/
|
|
void ast_sip_add_date_header(pjsip_tx_data *tdata);
|
|
|
|
/*!
|
|
* \brief Register a SIP service in Asterisk.
|
|
*
|
|
* This is more-or-less a wrapper around pjsip_endpt_register_module().
|
|
* Registering a service makes it so that PJSIP will call into the
|
|
* service at appropriate times. For more information about PJSIP module
|
|
* callbacks, see the PJSIP documentation. Asterisk modules that call
|
|
* this function will likely do so at module load time.
|
|
*
|
|
* \param module The module that is to be registered with PJSIP
|
|
* \retval 0 Success
|
|
* \retval -1 Failure
|
|
*/
|
|
int ast_sip_register_service(pjsip_module *module);
|
|
|
|
/*!
|
|
* This is the opposite of ast_sip_register_service(). Unregistering a
|
|
* service means that PJSIP will no longer call into the module any more.
|
|
* This will likely occur when an Asterisk module is unloaded.
|
|
*
|
|
* \param module The PJSIP module to unregister
|
|
*/
|
|
void ast_sip_unregister_service(pjsip_module *module);
|
|
|
|
/*!
|
|
* \brief Register a SIP authenticator
|
|
*
|
|
* An authenticator has three main purposes:
|
|
* 1) Determining if authentication should be performed on an incoming request
|
|
* 2) Gathering credentials necessary for issuing an authentication challenge
|
|
* 3) Authenticating a request that has credentials
|
|
*
|
|
* Asterisk provides a default authenticator, but it may be replaced by a
|
|
* custom one if desired.
|
|
*
|
|
* \param auth The authenticator to register
|
|
* \retval 0 Success
|
|
* \retval -1 Failure
|
|
*/
|
|
int ast_sip_register_authenticator(struct ast_sip_authenticator *auth);
|
|
|
|
/*!
|
|
* \brief Unregister a SIP authenticator
|
|
*
|
|
* When there is no authenticator registered, requests cannot be challenged
|
|
* or authenticated.
|
|
*
|
|
* \param auth The authenticator to unregister
|
|
*/
|
|
void ast_sip_unregister_authenticator(struct ast_sip_authenticator *auth);
|
|
|
|
/*!
|
|
* \brief Register an outbound SIP authenticator
|
|
*
|
|
* An outbound authenticator is responsible for creating responses to
|
|
* authentication challenges by remote endpoints.
|
|
*
|
|
* \param outbound_auth The authenticator to register
|
|
* \retval 0 Success
|
|
* \retval -1 Failure
|
|
*/
|
|
int ast_sip_register_outbound_authenticator(struct ast_sip_outbound_authenticator *outbound_auth);
|
|
|
|
/*!
|
|
* \brief Unregister an outbound SIP authenticator
|
|
*
|
|
* When there is no outbound authenticator registered, authentication challenges
|
|
* will be handled as any other final response would be.
|
|
*
|
|
* \param auth The authenticator to unregister
|
|
*/
|
|
void ast_sip_unregister_outbound_authenticator(struct ast_sip_outbound_authenticator *auth);
|
|
|
|
/*!
|
|
* \brief Register a SIP endpoint identifier with a name.
|
|
*
|
|
* An endpoint identifier's purpose is to determine which endpoint a given SIP
|
|
* message has come from.
|
|
*
|
|
* Multiple endpoint identifiers may be registered so that if an endpoint
|
|
* cannot be identified by one identifier, it may be identified by another.
|
|
*
|
|
* \param identifier The SIP endpoint identifier to register
|
|
* \param name The name of the endpoint identifier
|
|
* \retval 0 Success
|
|
* \retval -1 Failure
|
|
*/
|
|
int ast_sip_register_endpoint_identifier_with_name(struct ast_sip_endpoint_identifier *identifier,
|
|
const char *name);
|
|
|
|
/*!
|
|
* \brief Register a SIP endpoint identifier
|
|
*
|
|
* An endpoint identifier's purpose is to determine which endpoint a given SIP
|
|
* message has come from.
|
|
*
|
|
* Multiple endpoint identifiers may be registered so that if an endpoint
|
|
* cannot be identified by one identifier, it may be identified by another.
|
|
*
|
|
* Asterisk provides two endpoint identifiers. One identifies endpoints based
|
|
* on the user part of the From header URI. The other identifies endpoints based
|
|
* on the source IP address.
|
|
*
|
|
* If the order in which endpoint identifiers is run is important to you, then
|
|
* be sure to load individual endpoint identifier modules in the order you wish
|
|
* for them to be run in modules.conf
|
|
*
|
|
* \note endpoint identifiers registered using this method (no name specified)
|
|
* are placed at the front of the endpoint identifiers list ahead of any
|
|
* named identifiers.
|
|
*
|
|
* \param identifier The SIP endpoint identifier to register
|
|
* \retval 0 Success
|
|
* \retval -1 Failure
|
|
*/
|
|
int ast_sip_register_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier);
|
|
|
|
/*!
|
|
* \brief Unregister a SIP endpoint identifier
|
|
*
|
|
* This stops an endpoint identifier from being used.
|
|
*
|
|
* \param identifier The SIP endoint identifier to unregister
|
|
*/
|
|
void ast_sip_unregister_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier);
|
|
|
|
/*!
|
|
* \brief Allocate a new SIP endpoint
|
|
*
|
|
* This will return an endpoint with its refcount increased by one. This reference
|
|
* can be released using ao2_ref().
|
|
*
|
|
* \param name The name of the endpoint.
|
|
* \retval NULL Endpoint allocation failed
|
|
* \retval non-NULL The newly allocated endpoint
|
|
*/
|
|
void *ast_sip_endpoint_alloc(const char *name);
|
|
|
|
/*!
|
|
* \brief Change state of a persistent endpoint.
|
|
*
|
|
* \param endpoint_name The SIP endpoint name to change state.
|
|
* \param state The new state
|
|
* \retval 0 Success
|
|
* \retval -1 Endpoint not found
|
|
*/
|
|
int ast_sip_persistent_endpoint_update_state(const char *endpoint_name, enum ast_endpoint_state state);
|
|
|
|
/*!
|
|
* \brief Publish the change of state for a contact.
|
|
*
|
|
* \param endpoint_name The SIP endpoint name.
|
|
* \param contact_status The contact status.
|
|
*/
|
|
void ast_sip_persistent_endpoint_publish_contact_state(const char *endpoint_name, const struct ast_sip_contact_status *contact_status);
|
|
|
|
/*!
|
|
* \brief Retrieve the current status for a contact.
|
|
*
|
|
* \param contact The contact.
|
|
*
|
|
* \retval non-NULL Success
|
|
* \retval NULL Status information not found
|
|
*
|
|
* \note The returned contact status object is immutable.
|
|
*/
|
|
struct ast_sip_contact_status *ast_sip_get_contact_status(const struct ast_sip_contact *contact);
|
|
|
|
/*!
|
|
* \brief Get a pointer to the PJSIP endpoint.
|
|
*
|
|
* This is useful when modules have specific information they need
|
|
* to register with the PJSIP core.
|
|
* \retval NULL endpoint has not been created yet.
|
|
* \retval non-NULL PJSIP endpoint.
|
|
*/
|
|
pjsip_endpoint *ast_sip_get_pjsip_endpoint(void);
|
|
|
|
/*!
|
|
* \brief Get a pointer to the SIP sorcery structure.
|
|
*
|
|
* \retval NULL sorcery has not been initialized
|
|
* \retval non-NULL sorcery structure
|
|
*/
|
|
struct ast_sorcery *ast_sip_get_sorcery(void);
|
|
|
|
/*!
|
|
* \brief Retrieve a named AOR
|
|
*
|
|
* \param aor_name Name of the AOR
|
|
*
|
|
* \retval NULL if not found
|
|
* \retval non-NULL if found
|
|
*/
|
|
struct ast_sip_aor *ast_sip_location_retrieve_aor(const char *aor_name);
|
|
|
|
/*!
|
|
* \brief Retrieve the first bound contact for an AOR
|
|
*
|
|
* \param aor Pointer to the AOR
|
|
* \retval NULL if no contacts available
|
|
* \retval non-NULL if contacts available
|
|
*/
|
|
struct ast_sip_contact *ast_sip_location_retrieve_first_aor_contact(const struct ast_sip_aor *aor);
|
|
|
|
/*!
|
|
* \brief Retrieve the first bound contact for an AOR and filter based on flags
|
|
* \since 13.16.0
|
|
*
|
|
* \param aor Pointer to the AOR
|
|
* \param flags Filtering flags
|
|
* \retval NULL if no contacts available
|
|
* \retval non-NULL if contacts available
|
|
*/
|
|
struct ast_sip_contact *ast_sip_location_retrieve_first_aor_contact_filtered(const struct ast_sip_aor *aor,
|
|
unsigned int flags);
|
|
|
|
/*!
|
|
* \brief Retrieve all contacts currently available for an AOR
|
|
*
|
|
* \param aor Pointer to the AOR
|
|
*
|
|
* \retval NULL if no contacts available
|
|
* \retval non-NULL if contacts available
|
|
*
|
|
* \warning
|
|
* Since this function prunes expired contacts before returning, it holds a named write
|
|
* lock on the aor. If you already hold the lock, call ast_sip_location_retrieve_aor_contacts_nolock instead.
|
|
*/
|
|
struct ao2_container *ast_sip_location_retrieve_aor_contacts(const struct ast_sip_aor *aor);
|
|
|
|
/*!
|
|
* \brief Retrieve all contacts currently available for an AOR and filter based on flags
|
|
* \since 13.16.0
|
|
*
|
|
* \param aor Pointer to the AOR
|
|
* \param flags Filtering flags
|
|
*
|
|
* \retval NULL if no contacts available
|
|
* \retval non-NULL if contacts available
|
|
*
|
|
* \warning
|
|
* Since this function prunes expired contacts before returning, it holds a named write
|
|
* lock on the aor. If you already hold the lock, call ast_sip_location_retrieve_aor_contacts_nolock instead.
|
|
*/
|
|
struct ao2_container *ast_sip_location_retrieve_aor_contacts_filtered(const struct ast_sip_aor *aor,
|
|
unsigned int flags);
|
|
|
|
/*!
|
|
* \brief Retrieve all contacts currently available for an AOR without locking the AOR
|
|
* \since 13.9.0
|
|
*
|
|
* \param aor Pointer to the AOR
|
|
*
|
|
* \retval NULL if no contacts available
|
|
* \retval non-NULL if contacts available
|
|
*
|
|
* \warning
|
|
* This function should only be called if you already hold a named write lock on the aor.
|
|
*/
|
|
struct ao2_container *ast_sip_location_retrieve_aor_contacts_nolock(const struct ast_sip_aor *aor);
|
|
|
|
/*!
|
|
* \brief Retrieve all contacts currently available for an AOR without locking the AOR and filter based on flags
|
|
* \since 13.16.0
|
|
*
|
|
* \param aor Pointer to the AOR
|
|
* \param flags Filtering flags
|
|
*
|
|
* \retval NULL if no contacts available
|
|
* \retval non-NULL if contacts available
|
|
*
|
|
* \warning
|
|
* This function should only be called if you already hold a named write lock on the aor.
|
|
*/
|
|
struct ao2_container *ast_sip_location_retrieve_aor_contacts_nolock_filtered(const struct ast_sip_aor *aor,
|
|
unsigned int flags);
|
|
|
|
/*!
|
|
* \brief Retrieve the first bound contact from a list of AORs
|
|
*
|
|
* \param aor_list A comma-separated list of AOR names
|
|
* \retval NULL if no contacts available
|
|
* \retval non-NULL if contacts available
|
|
*/
|
|
struct ast_sip_contact *ast_sip_location_retrieve_contact_from_aor_list(const char *aor_list);
|
|
|
|
/*!
|
|
* \brief Retrieve all contacts from a list of AORs
|
|
*
|
|
* \param aor_list A comma-separated list of AOR names
|
|
* \retval NULL if no contacts available
|
|
* \retval non-NULL container (which must be freed) if contacts available
|
|
*/
|
|
struct ao2_container *ast_sip_location_retrieve_contacts_from_aor_list(const char *aor_list);
|
|
|
|
/*!
|
|
* \brief Retrieve the first bound contact AND the AOR chosen from a list of AORs
|
|
*
|
|
* \param aor_list A comma-separated list of AOR names
|
|
* \param aor The chosen AOR
|
|
* \param contact The chosen contact
|
|
*/
|
|
void ast_sip_location_retrieve_contact_and_aor_from_list(const char *aor_list, struct ast_sip_aor **aor,
|
|
struct ast_sip_contact **contact);
|
|
|
|
/*!
|
|
* \brief Retrieve the first bound contact AND the AOR chosen from a list of AORs and filter based on flags
|
|
* \since 13.16.0
|
|
*
|
|
* \param aor_list A comma-separated list of AOR names
|
|
* \param flags Filtering flags
|
|
* \param aor The chosen AOR
|
|
* \param contact The chosen contact
|
|
*/
|
|
void ast_sip_location_retrieve_contact_and_aor_from_list_filtered(const char *aor_list, unsigned int flags,
|
|
struct ast_sip_aor **aor, struct ast_sip_contact **contact);
|
|
|
|
/*!
|
|
* \brief Retrieve a named contact
|
|
*
|
|
* \param contact_name Name of the contact
|
|
*
|
|
* \retval NULL if not found
|
|
* \retval non-NULL if found
|
|
*/
|
|
struct ast_sip_contact *ast_sip_location_retrieve_contact(const char *contact_name);
|
|
|
|
/*!
|
|
* \brief Add a new contact to an AOR
|
|
*
|
|
* \param aor Pointer to the AOR
|
|
* \param uri Full contact URI
|
|
* \param expiration_time Optional expiration time of the contact
|
|
* \param path_info Path information
|
|
* \param user_agent User-Agent header from REGISTER request
|
|
* \param via_addr
|
|
* \param via_port
|
|
* \param call_id
|
|
* \param endpoint The endpoint that resulted in the contact being added
|
|
*
|
|
* \retval -1 failure
|
|
* \retval 0 success
|
|
*
|
|
* \warning
|
|
* This function holds a named write lock on the aor. If you already hold the lock
|
|
* you should call ast_sip_location_add_contact_nolock instead.
|
|
*/
|
|
int ast_sip_location_add_contact(struct ast_sip_aor *aor, const char *uri,
|
|
struct timeval expiration_time, const char *path_info, const char *user_agent,
|
|
const char *via_addr, int via_port, const char *call_id,
|
|
struct ast_sip_endpoint *endpoint);
|
|
|
|
/*!
|
|
* \brief Add a new contact to an AOR without locking the AOR
|
|
* \since 13.9.0
|
|
*
|
|
* \param aor Pointer to the AOR
|
|
* \param uri Full contact URI
|
|
* \param expiration_time Optional expiration time of the contact
|
|
* \param path_info Path information
|
|
* \param user_agent User-Agent header from REGISTER request
|
|
* \param via_addr
|
|
* \param via_port
|
|
* \param call_id
|
|
* \param endpoint The endpoint that resulted in the contact being added
|
|
*
|
|
* \retval -1 failure
|
|
* \retval 0 success
|
|
*
|
|
* \warning
|
|
* This function should only be called if you already hold a named write lock on the aor.
|
|
*/
|
|
int ast_sip_location_add_contact_nolock(struct ast_sip_aor *aor, const char *uri,
|
|
struct timeval expiration_time, const char *path_info, const char *user_agent,
|
|
const char *via_addr, int via_port, const char *call_id,
|
|
struct ast_sip_endpoint *endpoint);
|
|
|
|
/*!
|
|
* \brief Create a new contact for an AOR without locking the AOR
|
|
* \since 13.18.0
|
|
*
|
|
* \param aor Pointer to the AOR
|
|
* \param uri Full contact URI
|
|
* \param expiration_time Optional expiration time of the contact
|
|
* \param path_info Path information
|
|
* \param user_agent User-Agent header from REGISTER request
|
|
* \param via_addr
|
|
* \param via_port
|
|
* \param call_id
|
|
* \param prune_on_boot Non-zero if the contact cannot survive a restart/boot.
|
|
* \param endpoint The endpoint that resulted in the contact being added
|
|
*
|
|
* \return The created contact or NULL on failure.
|
|
*
|
|
* \warning
|
|
* This function should only be called if you already hold a named write lock on the aor.
|
|
*/
|
|
struct ast_sip_contact *ast_sip_location_create_contact(struct ast_sip_aor *aor,
|
|
const char *uri, struct timeval expiration_time, const char *path_info,
|
|
const char *user_agent, const char *via_addr, int via_port, const char *call_id,
|
|
int prune_on_boot, struct ast_sip_endpoint *endpoint);
|
|
|
|
/*!
|
|
* \brief Update a contact
|
|
*
|
|
* \param contact New contact object with details
|
|
*
|
|
* \retval -1 failure
|
|
* \retval 0 success
|
|
*/
|
|
int ast_sip_location_update_contact(struct ast_sip_contact *contact);
|
|
|
|
/*!
|
|
* \brief Delete a contact
|
|
*
|
|
* \param contact Contact object to delete
|
|
*
|
|
* \retval -1 failure
|
|
* \retval 0 success
|
|
*/
|
|
int ast_sip_location_delete_contact(struct ast_sip_contact *contact);
|
|
|
|
/*!
|
|
* \brief Prune the prune_on_boot contacts
|
|
* \since 13.18.0
|
|
*/
|
|
void ast_sip_location_prune_boot_contacts(void);
|
|
|
|
/*!
|
|
* \brief Callback called when an outbound request with authentication credentials is to be sent in dialog
|
|
*
|
|
* This callback will have the created request on it. The callback's purpose is to do any extra
|
|
* housekeeping that needs to be done as well as to send the request out.
|
|
*
|
|
* This callback is only necessary if working with a PJSIP API that sits between the application
|
|
* and the dialog layer.
|
|
*
|
|
* \param dlg The dialog to which the request belongs
|
|
* \param tdata The created request to be sent out
|
|
* \param user_data Data supplied with the callback
|
|
*
|
|
* \retval 0 Success
|
|
* \retval -1 Failure
|
|
*/
|
|
typedef int (*ast_sip_dialog_outbound_auth_cb)(pjsip_dialog *dlg, pjsip_tx_data *tdata, void *user_data);
|
|
|
|
/*!
|
|
* \brief Set up outbound authentication on a SIP dialog
|
|
*
|
|
* This sets up the infrastructure so that all requests associated with a created dialog
|
|
* can be re-sent with authentication credentials if the original request is challenged.
|
|
*
|
|
* \param dlg The dialog on which requests will be authenticated
|
|
* \param endpoint The endpoint whom this dialog pertains to
|
|
* \param cb Callback to call to send requests with authentication
|
|
* \param user_data Data to be provided to the callback when it is called
|
|
*
|
|
* \retval 0 Success
|
|
* \retval -1 Failure
|
|
*/
|
|
int ast_sip_dialog_setup_outbound_authentication(pjsip_dialog *dlg, const struct ast_sip_endpoint *endpoint,
|
|
ast_sip_dialog_outbound_auth_cb cb, void *user_data);
|
|
|
|
/*!
|
|
* \brief Retrieves a reference to the artificial auth.
|
|
*
|
|
* \retval The artificial auth
|
|
*/
|
|
struct ast_sip_auth *ast_sip_get_artificial_auth(void);
|
|
|
|
/*!
|
|
* \brief Retrieves a reference to the artificial endpoint.
|
|
*
|
|
* \retval The artificial endpoint
|
|
*/
|
|
struct ast_sip_endpoint *ast_sip_get_artificial_endpoint(void);
|
|
|
|
/*! \defgroup pjsip_threading PJSIP Threading Model
|
|
* @{
|
|
* \page PJSIP PJSIP Threading Model
|
|
*
|
|
* There are three major types of threads that SIP will have to deal with:
|
|
* \li Asterisk threads
|
|
* \li PJSIP threads
|
|
* \li SIP threadpool threads (a.k.a. "servants")
|
|
*
|
|
* \par Asterisk Threads
|
|
*
|
|
* Asterisk threads are those that originate from outside of SIP but within
|
|
* Asterisk. The most common of these threads are PBX (channel) threads and
|
|
* the autoservice thread. Most interaction with these threads will be through
|
|
* channel technology callbacks. Within these threads, it is fine to handle
|
|
* Asterisk data from outside of SIP, but any handling of SIP data should be
|
|
* left to servants, \b especially if you wish to call into PJSIP for anything.
|
|
* Asterisk threads are not registered with PJLIB, so attempting to call into
|
|
* PJSIP will cause an assertion to be triggered, thus causing the program to
|
|
* crash.
|
|
*
|
|
* \par PJSIP Threads
|
|
*
|
|
* PJSIP threads are those that originate from handling of PJSIP events, such
|
|
* as an incoming SIP request or response, or a transaction timeout. The role
|
|
* of these threads is to process information as quickly as possible so that
|
|
* the next item on the SIP socket(s) can be serviced. On incoming messages,
|
|
* Asterisk automatically will push the request to a servant thread. When your
|
|
* module callback is called, processing will already be in a servant. However,
|
|
* for other PJSIP events, such as transaction state changes due to timer
|
|
* expirations, your module will be called into from a PJSIP thread. If you
|
|
* are called into from a PJSIP thread, then you should push whatever processing
|
|
* is needed to a servant as soon as possible. You can discern if you are currently
|
|
* in a SIP servant thread using the \ref ast_sip_thread_is_servant function.
|
|
*
|
|
* \par Servants
|
|
*
|
|
* Servants are where the bulk of SIP work should be performed. These threads
|
|
* exist in order to do the work that Asterisk threads and PJSIP threads hand
|
|
* off to them. Servant threads register themselves with PJLIB, meaning that
|
|
* they are capable of calling PJSIP and PJLIB functions if they wish.
|
|
*
|
|
* \par Serializer
|
|
*
|
|
* Tasks are handed off to servant threads using the API call \ref ast_sip_push_task.
|
|
* The first parameter of this call is a serializer. If this pointer
|
|
* is NULL, then the work will be handed off to whatever servant can currently handle
|
|
* the task. If this pointer is non-NULL, then the task will not be executed until
|
|
* previous tasks pushed with the same serializer have completed. For more information
|
|
* on serializers and the benefits they provide, see \ref ast_threadpool_serializer
|
|
*
|
|
* \par Scheduler
|
|
*
|
|
* Some situations require that a task run periodically or at a future time. Normally
|
|
* the ast_sched functionality would be used but ast_sched only uses 1 thread for all
|
|
* tasks and that thread isn't registered with PJLIB and therefore can't do any PJSIP
|
|
* related work.
|
|
*
|
|
* ast_sip_sched uses ast_sched only as a scheduled queue. When a task is ready to run,
|
|
* it's pushed to a Serializer to be invoked asynchronously by a Servant. This ensures
|
|
* that the task is executed in a PJLIB registered thread and allows the ast_sched thread
|
|
* to immediately continue processing the queue. The Serializer used by ast_sip_sched
|
|
* is one of your choosing or a random one from the res_pjsip pool if you don't choose one.
|
|
*
|
|
* \note
|
|
*
|
|
* Do not make assumptions about individual threads based on a corresponding serializer.
|
|
* In other words, just because several tasks use the same serializer when being pushed
|
|
* to servants, it does not mean that the same thread is necessarily going to execute those
|
|
* tasks, even though they are all guaranteed to be executed in sequence.
|
|
*/
|
|
|
|
typedef int (*ast_sip_task)(void *user_data);
|
|
|
|
/*!
|
|
* \brief Create a new serializer for SIP tasks
|
|
* \since 13.8.0
|
|
*
|
|
* See \ref ast_threadpool_serializer for more information on serializers.
|
|
* SIP creates serializers so that tasks operating on similar data will run
|
|
* in sequence.
|
|
*
|
|
* \param name Name of the serializer. (must be unique)
|
|
*
|
|
* \retval NULL Failure
|
|
* \retval non-NULL Newly-created serializer
|
|
*/
|
|
struct ast_taskprocessor *ast_sip_create_serializer(const char *name);
|
|
|
|
struct ast_serializer_shutdown_group;
|
|
|
|
/*!
|
|
* \brief Create a new serializer for SIP tasks
|
|
* \since 13.8.0
|
|
*
|
|
* See \ref ast_threadpool_serializer for more information on serializers.
|
|
* SIP creates serializers so that tasks operating on similar data will run
|
|
* in sequence.
|
|
*
|
|
* \param name Name of the serializer. (must be unique)
|
|
* \param shutdown_group Group shutdown controller. (NULL if no group association)
|
|
*
|
|
* \retval NULL Failure
|
|
* \retval non-NULL Newly-created serializer
|
|
*/
|
|
struct ast_taskprocessor *ast_sip_create_serializer_group(const char *name, struct ast_serializer_shutdown_group *shutdown_group);
|
|
|
|
/*!
|
|
* \brief Determine the distributor serializer for the SIP message.
|
|
* \since 13.10.0
|
|
*
|
|
* \param rdata The incoming message.
|
|
*
|
|
* \retval Calculated distributor serializer on success.
|
|
* \retval NULL on error.
|
|
*/
|
|
struct ast_taskprocessor *ast_sip_get_distributor_serializer(pjsip_rx_data *rdata);
|
|
|
|
/*!
|
|
* \brief Set a serializer on a SIP dialog so requests and responses are automatically serialized
|
|
*
|
|
* Passing a NULL serializer is a way to remove a serializer from a dialog.
|
|
*
|
|
* \param dlg The SIP dialog itself
|
|
* \param serializer The serializer to use
|
|
*/
|
|
void ast_sip_dialog_set_serializer(pjsip_dialog *dlg, struct ast_taskprocessor *serializer);
|
|
|
|
/*!
|
|
* \brief Set an endpoint on a SIP dialog so in-dialog requests do not undergo endpoint lookup.
|
|
*
|
|
* \param dlg The SIP dialog itself
|
|
* \param endpoint The endpoint that this dialog is communicating with
|
|
*/
|
|
void ast_sip_dialog_set_endpoint(pjsip_dialog *dlg, struct ast_sip_endpoint *endpoint);
|
|
|
|
/*!
|
|
* \brief Get the endpoint associated with this dialog
|
|
*
|
|
* This function increases the refcount of the endpoint by one. Release
|
|
* the reference once you are finished with the endpoint.
|
|
*
|
|
* \param dlg The SIP dialog from which to retrieve the endpoint
|
|
* \retval NULL No endpoint associated with this dialog
|
|
* \retval non-NULL The endpoint.
|
|
*/
|
|
struct ast_sip_endpoint *ast_sip_dialog_get_endpoint(pjsip_dialog *dlg);
|
|
|
|
/*!
|
|
* \brief Pushes a task to SIP servants
|
|
*
|
|
* This uses the serializer provided to determine how to push the task.
|
|
* If the serializer is NULL, then the task will be pushed to the
|
|
* servants directly. If the serializer is non-NULL, then the task will be
|
|
* queued behind other tasks associated with the same serializer.
|
|
*
|
|
* \param serializer The serializer to which the task belongs. Can be NULL
|
|
* \param sip_task The task to execute
|
|
* \param task_data The parameter to pass to the task when it executes
|
|
* \retval 0 Success
|
|
* \retval -1 Failure
|
|
*/
|
|
int ast_sip_push_task(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data);
|
|
|
|
/*!
|
|
* \brief Push a task to SIP servants and wait for it to complete.
|
|
*
|
|
* Like \ref ast_sip_push_task except that it blocks until the task
|
|
* completes. If the current thread is a SIP servant thread then the
|
|
* task executes immediately. Otherwise, the specified serializer
|
|
* executes the task and the current thread waits for it to complete.
|
|
*
|
|
* \note PJPROJECT callbacks tend to have locks already held when
|
|
* called.
|
|
*
|
|
* \warning \b Never hold locks that may be acquired by a SIP servant
|
|
* thread when calling this function. Doing so may cause a deadlock
|
|
* if all SIP servant threads are blocked waiting to acquire the lock
|
|
* while the thread holding the lock is waiting for a free SIP servant
|
|
* thread.
|
|
*
|
|
* \warning \b Use of this function in an ao2 destructor callback is a
|
|
* bad idea. You don't have control over which thread executes the
|
|
* destructor. Attempting to shift execution to another thread with
|
|
* this function is likely to cause deadlock.
|
|
*
|
|
* \param serializer The SIP serializer to execute the task if the
|
|
* current thread is not a SIP servant. NULL if any of the default
|
|
* serializers can be used.
|
|
* \param sip_task The task to execute
|
|
* \param task_data The parameter to pass to the task when it executes
|
|
*
|
|
* \note The sip_task() return value may need to be distinguished from
|
|
* the failure to push the task.
|
|
*
|
|
* \return sip_task() return value on success.
|
|
* \retval -1 Failure to push the task.
|
|
*/
|
|
int ast_sip_push_task_wait_servant(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data);
|
|
|
|
/*!
|
|
* \brief Push a task to SIP servants and wait for it to complete.
|
|
* \deprecated Replaced with ast_sip_push_task_wait_servant().
|
|
*/
|
|
int ast_sip_push_task_synchronous(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data);
|
|
|
|
/*!
|
|
* \brief Push a task to the serializer and wait for it to complete.
|
|
*
|
|
* Like \ref ast_sip_push_task except that it blocks until the task is
|
|
* completed by the specified serializer. If the specified serializer
|
|
* is the current thread then the task executes immediately.
|
|
*
|
|
* \note PJPROJECT callbacks tend to have locks already held when
|
|
* called.
|
|
*
|
|
* \warning \b Never hold locks that may be acquired by a SIP servant
|
|
* thread when calling this function. Doing so may cause a deadlock
|
|
* if all SIP servant threads are blocked waiting to acquire the lock
|
|
* while the thread holding the lock is waiting for a free SIP servant
|
|
* thread for the serializer to execute in.
|
|
*
|
|
* \warning \b Never hold locks that may be acquired by the serializer
|
|
* when calling this function. Doing so will cause a deadlock.
|
|
*
|
|
* \warning \b Never use this function in the pjsip monitor thread (It
|
|
* is a SIP servant thread). This is likely to cause a deadlock.
|
|
*
|
|
* \warning \b Use of this function in an ao2 destructor callback is a
|
|
* bad idea. You don't have control over which thread executes the
|
|
* destructor. Attempting to shift execution to another thread with
|
|
* this function is likely to cause deadlock.
|
|
*
|
|
* \param serializer The SIP serializer to execute the task. NULL if
|
|
* any of the default serializers can be used.
|
|
* \param sip_task The task to execute
|
|
* \param task_data The parameter to pass to the task when it executes
|
|
*
|
|
* \note It is generally better to call
|
|
* ast_sip_push_task_wait_servant() if you pass NULL for the
|
|
* serializer parameter.
|
|
*
|
|
* \note The sip_task() return value may need to be distinguished from
|
|
* the failure to push the task.
|
|
*
|
|
* \return sip_task() return value on success.
|
|
* \retval -1 Failure to push the task.
|
|
*/
|
|
int ast_sip_push_task_wait_serializer(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data);
|
|
|
|
/*!
|
|
* \brief Determine if the current thread is a SIP servant thread
|
|
*
|
|
* \retval 0 This is not a SIP servant thread
|
|
* \retval 1 This is a SIP servant thread
|
|
*/
|
|
int ast_sip_thread_is_servant(void);
|
|
|
|
/*!
|
|
* \brief Task flags for the res_pjsip scheduler
|
|
*
|
|
* The default is AST_SIP_SCHED_TASK_FIXED
|
|
* | AST_SIP_SCHED_TASK_DATA_NOT_AO2
|
|
* | AST_SIP_SCHED_TASK_DATA_NO_CLEANUP
|
|
* | AST_SIP_SCHED_TASK_PERIODIC
|
|
*/
|
|
enum ast_sip_scheduler_task_flags {
|
|
/*!
|
|
* The defaults
|
|
*/
|
|
AST_SIP_SCHED_TASK_DEFAULTS = (0 << 0),
|
|
|
|
/*!
|
|
* Run at a fixed interval.
|
|
* Stop scheduling if the callback returns <= 0.
|
|
* Any other value is ignored.
|
|
*/
|
|
AST_SIP_SCHED_TASK_FIXED = (0 << 0),
|
|
/*!
|
|
* Run at a variable interval.
|
|
* Stop scheduling if the callback returns <= 0.
|
|
* Any other return value is used as the new interval.
|
|
*/
|
|
AST_SIP_SCHED_TASK_VARIABLE = (1 << 0),
|
|
|
|
/*!
|
|
* Run just once.
|
|
* Return values are ignored.
|
|
*/
|
|
AST_SIP_SCHED_TASK_ONESHOT = (1 << 6),
|
|
|
|
/*!
|
|
* The task data is not an AO2 object.
|
|
*/
|
|
AST_SIP_SCHED_TASK_DATA_NOT_AO2 = (0 << 1),
|
|
/*!
|
|
* The task data is an AO2 object.
|
|
* A reference count will be held by the scheduler until
|
|
* after the task has run for the final time (if ever).
|
|
*/
|
|
AST_SIP_SCHED_TASK_DATA_AO2 = (1 << 1),
|
|
|
|
/*!
|
|
* Don't take any cleanup action on the data
|
|
*/
|
|
AST_SIP_SCHED_TASK_DATA_NO_CLEANUP = (0 << 3),
|
|
/*!
|
|
* If AST_SIP_SCHED_TASK_DATA_AO2 is set, decrement the reference count
|
|
* otherwise call ast_free on it.
|
|
*/
|
|
AST_SIP_SCHED_TASK_DATA_FREE = ( 1 << 3 ),
|
|
|
|
/*!
|
|
* \brief The task is scheduled at multiples of interval
|
|
* \see Interval
|
|
*/
|
|
AST_SIP_SCHED_TASK_PERIODIC = (0 << 4),
|
|
/*!
|
|
* \brief The next invocation of the task is at last finish + interval
|
|
* \see Interval
|
|
*/
|
|
AST_SIP_SCHED_TASK_DELAY = (1 << 4),
|
|
/*!
|
|
* \brief The scheduled task's events are tracked in the debug log.
|
|
* \details
|
|
* Schedule events such as scheduling, running, rescheduling, canceling,
|
|
* and destroying are logged about the task.
|
|
*/
|
|
AST_SIP_SCHED_TASK_TRACK = (1 << 5),
|
|
};
|
|
|
|
/*!
|
|
* \brief Scheduler task data structure
|
|
*/
|
|
struct ast_sip_sched_task;
|
|
|
|
/*!
|
|
* \brief Schedule a task to run in the res_pjsip thread pool
|
|
* \since 13.9.0
|
|
*
|
|
* \param serializer The serializer to use. If NULL, don't use a serializer (see note below)
|
|
* \param interval The invocation interval in milliseconds (see note below)
|
|
* \param sip_task The task to invoke
|
|
* \param name An optional name to associate with the task
|
|
* \param task_data Optional data to pass to the task
|
|
* \param flags One of enum ast_sip_scheduler_task_type
|
|
*
|
|
* \returns Pointer to \ref ast_sip_sched_task ao2 object which must be dereferenced when done.
|
|
*
|
|
* \par Serialization
|
|
*
|
|
* Specifying a serializer guarantees serialized execution but NOT specifying a serializer
|
|
* may still result in tasks being effectively serialized if the thread pool is busy.
|
|
* The point of the serializer BTW is not to prevent parallel executions of the SAME task.
|
|
* That happens automatically (see below). It's to prevent the task from running at the same
|
|
* time as other work using the same serializer, whether or not it's being run by the scheduler.
|
|
*
|
|
* \par Interval
|
|
*
|
|
* The interval is used to calculate the next time the task should run. There are two models.
|
|
*
|
|
* \ref AST_SIP_SCHED_TASK_PERIODIC specifies that the invocations of the task occur at the
|
|
* specific interval. That is, every \p interval milliseconds, regardless of how long the task
|
|
* takes. If the task takes longer than \p interval, it will be scheduled at the next available
|
|
* multiple of \p interval. For example: If the task has an interval of 60 seconds and the task
|
|
* takes 70 seconds, the next invocation will happen at 120 seconds.
|
|
*
|
|
* \ref AST_SIP_SCHED_TASK_DELAY specifies that the next invocation of the task should start
|
|
* at \p interval milliseconds after the current invocation has finished.
|
|
*
|
|
*/
|
|
struct ast_sip_sched_task *ast_sip_schedule_task(struct ast_taskprocessor *serializer,
|
|
int interval, ast_sip_task sip_task, const char *name, void *task_data,
|
|
enum ast_sip_scheduler_task_flags flags);
|
|
|
|
/*!
|
|
* \brief Cancels the next invocation of a task
|
|
* \since 13.9.0
|
|
*
|
|
* \param schtd The task structure pointer
|
|
* \retval 0 Success
|
|
* \retval -1 Failure
|
|
* \note Only cancels future invocations not the currently running invocation.
|
|
*/
|
|
int ast_sip_sched_task_cancel(struct ast_sip_sched_task *schtd);
|
|
|
|
/*!
|
|
* \brief Cancels the next invocation of a task by name
|
|
* \since 13.9.0
|
|
*
|
|
* \param name The task name
|
|
* \retval 0 Success
|
|
* \retval -1 Failure
|
|
* \note Only cancels future invocations not the currently running invocation.
|
|
*/
|
|
int ast_sip_sched_task_cancel_by_name(const char *name);
|
|
|
|
/*!
|
|
* \brief Gets the last start and end times of the task
|
|
* \since 13.9.0
|
|
*
|
|
* \param schtd The task structure pointer
|
|
* \param[out] when_queued Pointer to a timeval structure to contain the time when queued
|
|
* \param[out] last_start Pointer to a timeval structure to contain the time when last started
|
|
* \param[out] last_end Pointer to a timeval structure to contain the time when last ended
|
|
* \retval 0 Success
|
|
* \retval -1 Failure
|
|
* \note Any of the pointers can be NULL if you don't need them.
|
|
*/
|
|
int ast_sip_sched_task_get_times(struct ast_sip_sched_task *schtd,
|
|
struct timeval *when_queued, struct timeval *last_start, struct timeval *last_end);
|
|
|
|
/*!
|
|
* \brief Gets the queued, last start, last_end, time left, interval, next run
|
|
* \since 16.15.0
|
|
* \since 18.1.0
|
|
*
|
|
* \param schtd The task structure pointer
|
|
* \param[out] when_queued Pointer to a timeval structure to contain the time when queued
|
|
* \param[out] last_start Pointer to a timeval structure to contain the time when last started
|
|
* \param[out] last_end Pointer to a timeval structure to contain the time when last ended
|
|
* \param[out] interval Pointer to an int to contain the interval in ms
|
|
* \param[out] time_left Pointer to an int to contain the ms left to the next run
|
|
* \param[out] next_start Pointer to a timeval structure to contain the next run time
|
|
* \retval 0 Success
|
|
* \retval -1 Failure
|
|
* \note Any of the pointers can be NULL if you don't need them.
|
|
*/
|
|
int ast_sip_sched_task_get_times2(struct ast_sip_sched_task *schtd,
|
|
struct timeval *when_queued, struct timeval *last_start, struct timeval *last_end,
|
|
int *interval, int *time_left, struct timeval *next_start);
|
|
|
|
/*!
|
|
* \brief Gets the last start and end times of the task by name
|
|
* \since 13.9.0
|
|
*
|
|
* \param name The task name
|
|
* \param[out] when_queued Pointer to a timeval structure to contain the time when queued
|
|
* \param[out] last_start Pointer to a timeval structure to contain the time when last started
|
|
* \param[out] last_end Pointer to a timeval structure to contain the time when last ended
|
|
* \retval 0 Success
|
|
* \retval -1 Failure
|
|
* \note Any of the pointers can be NULL if you don't need them.
|
|
*/
|
|
int ast_sip_sched_task_get_times_by_name(const char *name,
|
|
struct timeval *when_queued, struct timeval *last_start, struct timeval *last_end);
|
|
|
|
/*!
|
|
* \brief Gets the queued, last start, last_end, time left, interval, next run by task name
|
|
* \since 16.15.0
|
|
* \since 18.1.0
|
|
*
|
|
* \param name The task name
|
|
* \param[out] when_queued Pointer to a timeval structure to contain the time when queued
|
|
* \param[out] last_start Pointer to a timeval structure to contain the time when last started
|
|
* \param[out] last_end Pointer to a timeval structure to contain the time when last ended
|
|
* \param[out] interval Pointer to an int to contain the interval in ms
|
|
* \param[out] time_left Pointer to an int to contain the ms left to the next run
|
|
* \param[out] next_start Pointer to a timeval structure to contain the next run time
|
|
* \retval 0 Success
|
|
* \retval -1 Failure
|
|
* \note Any of the pointers can be NULL if you don't need them.
|
|
*/
|
|
int ast_sip_sched_task_get_times_by_name2(const char *name,
|
|
struct timeval *when_queued, struct timeval *last_start, struct timeval *last_end,
|
|
int *interval, int *time_left, struct timeval *next_start);
|
|
|
|
/*!
|
|
* \brief Gets the number of milliseconds until the next invocation
|
|
* \since 13.9.0
|
|
*
|
|
* \param schtd The task structure pointer
|
|
* \return The number of milliseconds until the next invocation or -1 if the task isn't scheduled
|
|
*/
|
|
int ast_sip_sched_task_get_next_run(struct ast_sip_sched_task *schtd);
|
|
|
|
/*!
|
|
* \brief Gets the number of milliseconds until the next invocation
|
|
* \since 13.9.0
|
|
*
|
|
* \param name The task name
|
|
* \return The number of milliseconds until the next invocation or -1 if the task isn't scheduled
|
|
*/
|
|
int ast_sip_sched_task_get_next_run_by_name(const char *name);
|
|
|
|
/*!
|
|
* \brief Checks if the task is currently running
|
|
* \since 13.9.0
|
|
*
|
|
* \param schtd The task structure pointer
|
|
* \retval 0 not running
|
|
* \retval 1 running
|
|
*/
|
|
int ast_sip_sched_is_task_running(struct ast_sip_sched_task *schtd);
|
|
|
|
/*!
|
|
* \brief Checks if the task is currently running
|
|
* \since 13.9.0
|
|
*
|
|
* \param name The task name
|
|
* \retval 0 not running or not found
|
|
* \retval 1 running
|
|
*/
|
|
int ast_sip_sched_is_task_running_by_name(const char *name);
|
|
|
|
/*!
|
|
* \brief Gets the task name
|
|
* \since 13.9.0
|
|
*
|
|
* \param schtd The task structure pointer
|
|
* \param name, maxlen
|
|
* \retval 0 success
|
|
* \retval 1 failure
|
|
*/
|
|
int ast_sip_sched_task_get_name(struct ast_sip_sched_task *schtd, char *name, size_t maxlen);
|
|
|
|
/*!
|
|
* @}
|
|
*/
|
|
|
|
/*!
|
|
* \brief SIP body description
|
|
*
|
|
* This contains a type and subtype that will be added as
|
|
* the "Content-Type" for the message as well as the body
|
|
* text.
|
|
*/
|
|
struct ast_sip_body {
|
|
/*! Type of the body, such as "application" */
|
|
const char *type;
|
|
/*! Subtype of the body, such as "sdp" */
|
|
const char *subtype;
|
|
/*! The text to go in the body */
|
|
const char *body_text;
|
|
};
|
|
|
|
/*!
|
|
* \brief General purpose method for creating a UAC dialog with an endpoint
|
|
*
|
|
* \param endpoint A pointer to the endpoint
|
|
* \param aor_name Optional name of the AOR to target, may even be an explicit SIP URI
|
|
* \param request_user Optional user to place into the target URI
|
|
*
|
|
* \retval non-NULL success
|
|
* \retval NULL failure
|
|
*/
|
|
pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint, const char *aor_name, const char *request_user);
|
|
|
|
/*!
|
|
* \brief General purpose method for creating a UAS dialog with an endpoint
|
|
*
|
|
* \deprecated This function is unsafe (due to the returned object not being locked nor
|
|
* having its reference incremented) and should no longer be used. Instead
|
|
* use ast_sip_create_dialog_uas_locked so a properly locked and referenced
|
|
* object is returned.
|
|
*
|
|
* \param endpoint A pointer to the endpoint
|
|
* \param rdata The request that is starting the dialog
|
|
* \param[out] status On failure, the reason for failure in creating the dialog
|
|
*/
|
|
pjsip_dialog *ast_sip_create_dialog_uas(const struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata, pj_status_t *status);
|
|
|
|
/*!
|
|
* \brief General purpose method for creating a UAS dialog with an endpoint
|
|
*
|
|
* This function creates and returns a locked, and referenced counted pjsip
|
|
* dialog object. The caller is thus responsible for freeing the allocated
|
|
* memory, decrementing the reference, and releasing the lock when done with
|
|
* the returned object.
|
|
*
|
|
* \note The safest way to unlock the object, and decrement its reference is by
|
|
* calling pjsip_dlg_dec_lock. Alternatively, pjsip_dlg_dec_session can be
|
|
* used to decrement the reference only.
|
|
*
|
|
* The dialog is returned locked and with a reference in order to ensure that the
|
|
* dialog object, and any of its associated objects (e.g. transaction) are not
|
|
* untimely destroyed. For instance, that could happen when a transport error
|
|
* occurs.
|
|
*
|
|
* As long as the caller maintains a reference to the dialog there should be no
|
|
* worry that it might unknowingly be destroyed. However, once the caller unlocks
|
|
* the dialog there is a danger that some of the dialog's internal objects could
|
|
* be lost and/or compromised. For example, when the aforementioned transport error
|
|
* occurs the dialog's associated transaction gets destroyed (see pjsip_dlg_on_tsx_state
|
|
* in sip_dialog.c, and mod_inv_on_tsx_state in sip_inv.c).
|
|
*
|
|
* In this case and before using the dialog again the caller should re-lock the
|
|
* dialog, check to make sure the dialog is still established, and the transaction
|
|
* still exists and has not been destroyed.
|
|
*
|
|
* \param endpoint A pointer to the endpoint
|
|
* \param rdata The request that is starting the dialog
|
|
* \param[out] status On failure, the reason for failure in creating the dialog
|
|
*
|
|
* \retval A locked, and reference counted pjsip_dialog object.
|
|
* \retval NULL on failure
|
|
*/
|
|
pjsip_dialog *ast_sip_create_dialog_uas_locked(const struct ast_sip_endpoint *endpoint,
|
|
pjsip_rx_data *rdata, pj_status_t *status);
|
|
|
|
/*!
|
|
* \brief General purpose method for creating an rdata structure using specific information
|
|
* \since 13.15.0
|
|
*
|
|
* \param[out] rdata The rdata structure that will be populated
|
|
* \param packet A SIP message
|
|
* \param src_name The source IP address of the message
|
|
* \param src_port The source port of the message
|
|
* \param transport_type The type of transport the message was received on
|
|
* \param local_name The local IP address the message was received on
|
|
* \param local_port The local port the message was received on
|
|
* \param contact_uri The contact URI of the message
|
|
*
|
|
* \retval 0 success
|
|
* \retval -1 failure
|
|
*/
|
|
int ast_sip_create_rdata_with_contact(pjsip_rx_data *rdata, char *packet,
|
|
const char *src_name, int src_port, char *transport_type, const char *local_name,
|
|
int local_port, const char *contact_uri);
|
|
|
|
/*!
|
|
* \brief General purpose method for creating an rdata structure using specific information
|
|
*
|
|
* \param[out] rdata The rdata structure that will be populated
|
|
* \param packet A SIP message
|
|
* \param src_name The source IP address of the message
|
|
* \param src_port The source port of the message
|
|
* \param transport_type The type of transport the message was received on
|
|
* \param local_name The local IP address the message was received on
|
|
* \param local_port The local port the message was received on
|
|
*
|
|
* \retval 0 success
|
|
* \retval -1 failure
|
|
*/
|
|
int ast_sip_create_rdata(pjsip_rx_data *rdata, char *packet, const char *src_name,
|
|
int src_port, char *transport_type, const char *local_name, int local_port);
|
|
|
|
/*!
|
|
* \brief General purpose method for creating a SIP request
|
|
*
|
|
* Its typical use would be to create one-off requests such as an out of dialog
|
|
* SIP MESSAGE.
|
|
*
|
|
* The request can either be in- or out-of-dialog. If in-dialog, the
|
|
* dlg parameter MUST be present. If out-of-dialog the endpoint parameter
|
|
* MUST be present. If both are present, then we will assume that the message
|
|
* is to be sent in-dialog.
|
|
*
|
|
* The uri parameter can be specified if the request should be sent to an explicit
|
|
* URI rather than one configured on the endpoint.
|
|
*
|
|
* \param method The method of the SIP request to send
|
|
* \param dlg Optional. If specified, the dialog on which to request the message.
|
|
* \param endpoint Optional. If specified, the request will be created out-of-dialog to the endpoint.
|
|
* \param uri Optional. If specified, the request will be sent to this URI rather
|
|
* than one configured for the endpoint.
|
|
* \param contact The contact with which this request is associated for out-of-dialog requests.
|
|
* \param[out] tdata The newly-created request
|
|
*
|
|
* The provided contact is attached to tdata with its reference bumped, but will
|
|
* not survive for the entire lifetime of tdata since the contact is cleaned up
|
|
* when all supplements have completed execution.
|
|
*
|
|
* \retval 0 Success
|
|
* \retval -1 Failure
|
|
*/
|
|
int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
|
|
struct ast_sip_endpoint *endpoint, const char *uri,
|
|
struct ast_sip_contact *contact, pjsip_tx_data **tdata);
|
|
|
|
/*!
|
|
* \brief General purpose method for sending a SIP request
|
|
*
|
|
* This is a companion function for \ref ast_sip_create_request. The request
|
|
* created there can be passed to this function, though any request may be
|
|
* passed in.
|
|
*
|
|
* This will automatically set up handling outbound authentication challenges if
|
|
* they arrive.
|
|
*
|
|
* \param tdata The request to send
|
|
* \param dlg Optional. The dialog in which the request is sent. Otherwise it is out-of-dialog.
|
|
* \param endpoint Optional. If specified, the out-of-dialog request is sent to the endpoint.
|
|
* \param token Data to be passed to the callback upon receipt of out-of-dialog response.
|
|
* \param callback Callback to be called upon receipt of out-of-dialog response.
|
|
*
|
|
* \retval 0 Success
|
|
* \retval -1 Failure (out-of-dialog callback will not be called.)
|
|
*/
|
|
int ast_sip_send_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg,
|
|
struct ast_sip_endpoint *endpoint, void *token,
|
|
void (*callback)(void *token, pjsip_event *e));
|
|
|
|
/*!
|
|
* \brief General purpose method for sending an Out-Of-Dialog SIP request
|
|
*
|
|
* This is a companion function for \ref ast_sip_create_request. The request
|
|
* created there can be passed to this function, though any request may be
|
|
* passed in.
|
|
*
|
|
* This will automatically set up handling outbound authentication challenges if
|
|
* they arrive.
|
|
*
|
|
* \param tdata The request to send
|
|
* \param endpoint Optional. If specified, the out-of-dialog request is sent to the endpoint.
|
|
* \param timeout If non-zero, after the timeout the transaction will be terminated
|
|
* and the callback will be called with the PJSIP_EVENT_TIMER type.
|
|
* \param token Data to be passed to the callback upon receipt of out-of-dialog response.
|
|
* \param callback Callback to be called upon receipt of out-of-dialog response.
|
|
*
|
|
* \retval 0 Success
|
|
* \retval -1 Failure (out-of-dialog callback will not be called.)
|
|
*
|
|
* \note Timeout processing:
|
|
* There are 2 timers associated with this request, PJSIP timer_b which is
|
|
* set globally in the "system" section of pjsip.conf, and the timeout specified
|
|
* on this call. The timer that expires first (before normal completion) will
|
|
* cause the callback to be run with e->body.tsx_state.type = PJSIP_EVENT_TIMER.
|
|
* The timer that expires second is simply ignored and the callback is not run again.
|
|
*/
|
|
int ast_sip_send_out_of_dialog_request(pjsip_tx_data *tdata,
|
|
struct ast_sip_endpoint *endpoint, int timeout, void *token,
|
|
void (*callback)(void *token, pjsip_event *e));
|
|
|
|
/*!
|
|
* \brief General purpose method for creating a SIP response
|
|
*
|
|
* Its typical use would be to create responses for out of dialog
|
|
* requests.
|
|
*
|
|
* \param rdata The rdata from the incoming request.
|
|
* \param st_code The response code to transmit.
|
|
* \param contact The contact with which this request is associated.
|
|
* \param[out] p_tdata The newly-created response
|
|
*
|
|
* The provided contact is attached to tdata with its reference bumped, but will
|
|
* not survive for the entire lifetime of tdata since the contact is cleaned up
|
|
* when all supplements have completed execution.
|
|
*
|
|
* \retval 0 Success
|
|
* \retval -1 Failure
|
|
*/
|
|
int ast_sip_create_response(const pjsip_rx_data *rdata, int st_code,
|
|
struct ast_sip_contact *contact, pjsip_tx_data **p_tdata);
|
|
|
|
/*!
|
|
* \brief Send a response to an out of dialog request
|
|
*
|
|
* Use this function sparingly, since this does not create a transaction
|
|
* within PJSIP. This means that if the request is retransmitted, it is
|
|
* your responsibility to detect this and not process the same request
|
|
* twice, and to send the same response for each retransmission.
|
|
*
|
|
* \param res_addr The response address for this response
|
|
* \param tdata The response to send
|
|
* \param sip_endpoint The ast_sip_endpoint associated with this response
|
|
*
|
|
* \retval 0 Success
|
|
* \retval -1 Failure
|
|
*/
|
|
int ast_sip_send_response(pjsip_response_addr *res_addr, pjsip_tx_data *tdata, struct ast_sip_endpoint *sip_endpoint);
|
|
|
|
/*!
|
|
* \brief Send a stateful response to an out of dialog request
|
|
*
|
|
* This creates a transaction within PJSIP, meaning that if the request
|
|
* that we are responding to is retransmitted, we will not attempt to
|
|
* re-handle the request.
|
|
*
|
|
* \param rdata The request that is being responded to
|
|
* \param tdata The response to send
|
|
* \param sip_endpoint The ast_sip_endpoint associated with this response
|
|
*
|
|
* \since 13.4.0
|
|
*
|
|
* \retval 0 Success
|
|
* \retval -1 Failure
|
|
*/
|
|
int ast_sip_send_stateful_response(pjsip_rx_data *rdata, pjsip_tx_data *tdata, struct ast_sip_endpoint *sip_endpoint);
|
|
|
|
/*!
|
|
* \brief Determine if an incoming request requires authentication
|
|
*
|
|
* This calls into the registered authenticator's requires_authentication callback
|
|
* in order to determine if the request requires authentication.
|
|
*
|
|
* If there is no registered authenticator, then authentication will be assumed
|
|
* not to be required.
|
|
*
|
|
* \param endpoint The endpoint from which the request originates
|
|
* \param rdata The incoming SIP request
|
|
* \retval non-zero The request requires authentication
|
|
* \retval 0 The request does not require authentication
|
|
*/
|
|
int ast_sip_requires_authentication(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata);
|
|
|
|
/*!
|
|
* \brief Method to determine authentication status of an incoming request
|
|
*
|
|
* This will call into a registered authenticator. The registered authenticator will
|
|
* do what is necessary to determine whether the incoming request passes authentication.
|
|
* A tentative response is passed into this function so that if, say, a digest authentication
|
|
* challenge should be sent in the ensuing response, it can be added to the response.
|
|
*
|
|
* \param endpoint The endpoint from the request was sent
|
|
* \param rdata The request to potentially authenticate
|
|
* \param tdata Tentative response to the request
|
|
* \return The result of checking authentication.
|
|
*/
|
|
enum ast_sip_check_auth_result ast_sip_check_authentication(struct ast_sip_endpoint *endpoint,
|
|
pjsip_rx_data *rdata, pjsip_tx_data *tdata);
|
|
|
|
/*!
|
|
* \brief Create a response to an authentication challenge
|
|
*
|
|
* This will call into an outbound authenticator's create_request_with_auth callback
|
|
* to create a new request with authentication credentials. See the create_request_with_auth
|
|
* callback in the \ref ast_sip_outbound_authenticator structure for details about
|
|
* the parameters and return values.
|
|
*/
|
|
int ast_sip_create_request_with_auth(const struct ast_sip_auth_vector *auths, pjsip_rx_data *challenge,
|
|
pjsip_tx_data *tdata, pjsip_tx_data **new_request);
|
|
|
|
/*!
|
|
* \brief Determine the endpoint that has sent a SIP message
|
|
*
|
|
* This will call into each of the registered endpoint identifiers'
|
|
* identify_endpoint() callbacks until one returns a non-NULL endpoint.
|
|
* This will return an ao2 object. Its reference count will need to be
|
|
* decremented when completed using the endpoint.
|
|
*
|
|
* \param rdata The inbound SIP message to use when identifying the endpoint.
|
|
* \retval NULL No matching endpoint
|
|
* \retval non-NULL The matching endpoint
|
|
*/
|
|
struct ast_sip_endpoint *ast_sip_identify_endpoint(pjsip_rx_data *rdata);
|
|
|
|
/*!
|
|
* \brief Get a specific header value from rdata
|
|
*
|
|
* \note The returned value does not need to be freed since it's from the rdata pool
|
|
*
|
|
* \param rdata The rdata
|
|
* \param str The header to find
|
|
*
|
|
* \retval NULL on failure
|
|
* \retval The header value on success
|
|
*/
|
|
char *ast_sip_rdata_get_header_value(pjsip_rx_data *rdata, const pj_str_t str);
|
|
|
|
/*!
|
|
* \brief Set the outbound proxy for an outbound SIP message
|
|
*
|
|
* \param tdata The message to set the outbound proxy on
|
|
* \param proxy SIP uri of the proxy
|
|
* \retval 0 Success
|
|
* \retval -1 Failure
|
|
*/
|
|
int ast_sip_set_outbound_proxy(pjsip_tx_data *tdata, const char *proxy);
|
|
|
|
/*!
|
|
* \brief Add a header to an outbound SIP message
|
|
*
|
|
* \param tdata The message to add the header to
|
|
* \param name The header name
|
|
* \param value The header value
|
|
* \retval 0 Success
|
|
* \retval -1 Failure
|
|
*/
|
|
int ast_sip_add_header(pjsip_tx_data *tdata, const char *name, const char *value);
|
|
|
|
/*!
|
|
* \brief Add a header to an outbound SIP message, returning a pointer to the header
|
|
*
|
|
* \param tdata The message to add the header to
|
|
* \param name The header name
|
|
* \param value The header value
|
|
* \return The pjsip_generic_string_hdr * added.
|
|
*/
|
|
pjsip_generic_string_hdr *ast_sip_add_header2(pjsip_tx_data *tdata,
|
|
const char *name, const char *value);
|
|
|
|
/*!
|
|
* \brief Add a body to an outbound SIP message
|
|
*
|
|
* If this is called multiple times, the latest body will replace the current
|
|
* body.
|
|
*
|
|
* \param tdata The message to add the body to
|
|
* \param body The message body to add
|
|
* \retval 0 Success
|
|
* \retval -1 Failure
|
|
*/
|
|
int ast_sip_add_body(pjsip_tx_data *tdata, const struct ast_sip_body *body);
|
|
|
|
/*!
|
|
* \brief Add a multipart body to an outbound SIP message
|
|
*
|
|
* This will treat each part of the input vector as part of a multipart body and
|
|
* add each part to the SIP message.
|
|
*
|
|
* \param tdata The message to add the body to
|
|
* \param bodies The message bodies to add
|
|
* \param num_bodies The parts of the body to add
|
|
* \retval 0 Success
|
|
* \retval -1 Failure
|
|
*/
|
|
int ast_sip_add_body_multipart(pjsip_tx_data *tdata, const struct ast_sip_body *bodies[], int num_bodies);
|
|
|
|
/*!
|
|
* \brief Append body data to a SIP message
|
|
*
|
|
* This acts mostly the same as ast_sip_add_body, except that rather than replacing
|
|
* a body if it currently exists, it appends data to an existing body.
|
|
*
|
|
* \param tdata The message to append the body to
|
|
* \param body_text The string to append to the end of the current body
|
|
* \retval 0 Success
|
|
* \retval -1 Failure
|
|
*/
|
|
int ast_sip_append_body(pjsip_tx_data *tdata, const char *body_text);
|
|
|
|
/*!
|
|
* \brief Copy a pj_str_t into a standard character buffer.
|
|
*
|
|
* pj_str_t is not NULL-terminated. Any place that expects a NULL-
|
|
* terminated string needs to have the pj_str_t copied into a separate
|
|
* buffer.
|
|
*
|
|
* This method copies the pj_str_t contents into the destination buffer
|
|
* and NULL-terminates the buffer.
|
|
*
|
|
* \param dest The destination buffer
|
|
* \param src The pj_str_t to copy
|
|
* \param size The size of the destination buffer.
|
|
*/
|
|
void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size);
|
|
|
|
/*!
|
|
* \brief Create and copy a pj_str_t into a standard character buffer.
|
|
*
|
|
* pj_str_t is not NULL-terminated. Any place that expects a NULL-
|
|
* terminated string needs to have the pj_str_t copied into a separate
|
|
* buffer.
|
|
*
|
|
* Copies the pj_str_t contents into a newly allocated buffer pointed to
|
|
* by dest. NULL-terminates the buffer.
|
|
*
|
|
* \note Caller is responsible for freeing the allocated memory.
|
|
*
|
|
* \param[out] dest The destination buffer
|
|
* \param src The pj_str_t to copy
|
|
* \return Number of characters copied or negative value on error
|
|
*/
|
|
int ast_copy_pj_str2(char **dest, const pj_str_t *src);
|
|
|
|
/*!
|
|
* \brief Get the looked-up endpoint on an out-of dialog request or response
|
|
*
|
|
* The function may ONLY be called on out-of-dialog requests or responses. For
|
|
* in-dialog requests and responses, it is required that the user of the dialog
|
|
* has the looked-up endpoint stored locally.
|
|
*
|
|
* This function should never return NULL if the message is out-of-dialog. It will
|
|
* always return NULL if the message is in-dialog.
|
|
*
|
|
* This function will increase the reference count of the returned endpoint by one.
|
|
* Release your reference using the ao2_ref function when finished.
|
|
*
|
|
* \param rdata Out-of-dialog request or response
|
|
* \return The looked up endpoint
|
|
*/
|
|
struct ast_sip_endpoint *ast_pjsip_rdata_get_endpoint(pjsip_rx_data *rdata);
|
|
|
|
/*!
|
|
* \brief Add 'user=phone' parameter to URI if enabled and user is a phone number.
|
|
*
|
|
* \param endpoint The endpoint to use for configuration
|
|
* \param pool The memory pool to allocate the parameter from
|
|
* \param uri The URI to check for user and to add parameter to
|
|
*/
|
|
void ast_sip_add_usereqphone(const struct ast_sip_endpoint *endpoint, pj_pool_t *pool, pjsip_uri *uri);
|
|
|
|
/*!
|
|
* \brief Retrieve any endpoints available to sorcery.
|
|
*
|
|
* \retval Endpoints available to sorcery, NULL if no endpoints found.
|
|
*/
|
|
struct ao2_container *ast_sip_get_endpoints(void);
|
|
|
|
/*!
|
|
* \brief Retrieve the default outbound endpoint.
|
|
*
|
|
* \retval The default outbound endpoint, NULL if not found.
|
|
*/
|
|
struct ast_sip_endpoint *ast_sip_default_outbound_endpoint(void);
|
|
|
|
/*!
|
|
* \brief Retrieve relevant SIP auth structures from sorcery
|
|
*
|
|
* \param auths Vector of sorcery IDs of auth credentials to retrieve
|
|
* \param[out] out The retrieved auths are stored here
|
|
*/
|
|
int ast_sip_retrieve_auths(const struct ast_sip_auth_vector *auths, struct ast_sip_auth **out);
|
|
|
|
/*!
|
|
* \brief Clean up retrieved auth structures from memory
|
|
*
|
|
* Call this function once you have completed operating on auths
|
|
* retrieved from \ref ast_sip_retrieve_auths
|
|
*
|
|
* \param auths An array of auth object pointers to clean up
|
|
* \param num_auths The number of auths in the array
|
|
*/
|
|
void ast_sip_cleanup_auths(struct ast_sip_auth *auths[], size_t num_auths);
|
|
|
|
AST_VECTOR(ast_sip_auth_objects_vector, struct ast_sip_auth *);
|
|
/*!
|
|
* \brief Retrieve relevant SIP auth structures from sorcery as a vector
|
|
*
|
|
* \param auth_ids Vector of sorcery IDs of auth credentials to retrieve
|
|
* \param[out] auth_objects A pointer ast_sip_auth_objects_vector to hold the objects
|
|
*
|
|
* \retval 0 Success
|
|
* \retval -1 Number of auth objects found is less than the number of names supplied.
|
|
*
|
|
* \warning The number of auth objects retrieved may be less than the
|
|
* number of auth ids supplied if auth objects couldn't be found for
|
|
* some of them.
|
|
*
|
|
* \note Since the ref count on all auth objects returned has been
|
|
* bumped, you must call ast_sip_cleanup_auth_objects_vector() to decrement
|
|
* the ref count on all of the auth objects in the vector,
|
|
* then call AST_VECTOR_FREE() on the vector itself.
|
|
*
|
|
*/
|
|
int ast_sip_retrieve_auths_vector(const struct ast_sip_auth_vector *auth_ids,
|
|
struct ast_sip_auth_objects_vector *auth_objects);
|
|
|
|
/*!
|
|
* \brief Clean up retrieved auth objects in vector
|
|
*
|
|
* Call this function once you have completed operating on auths
|
|
* retrieved from \ref ast_sip_retrieve_auths_vector. All
|
|
* auth objects will have their reference counts decremented and
|
|
* the vector size will be reset to 0. You must still call
|
|
* AST_VECTOR_FREE() on the vector itself.
|
|
*
|
|
* \param auth_objects A vector of auth structures to clean up
|
|
*/
|
|
#define ast_sip_cleanup_auth_objects_vector(auth_objects) AST_VECTOR_RESET(auth_objects, ao2_cleanup)
|
|
|
|
/*!
|
|
* \brief Checks if the given content type matches type/subtype.
|
|
*
|
|
* Compares the pjsip_media_type with the passed type and subtype and
|
|
* returns the result of that comparison. The media type parameters are
|
|
* ignored.
|
|
*
|
|
* \param content_type The pjsip_media_type structure to compare
|
|
* \param type The media type to compare
|
|
* \param subtype The media subtype to compare
|
|
* \retval 0 No match
|
|
* \retval -1 Match
|
|
*/
|
|
int ast_sip_is_content_type(pjsip_media_type *content_type, char *type, char *subtype);
|
|
|
|
/*!
|
|
* \brief Send a security event notification for when an invalid endpoint is requested
|
|
*
|
|
* \param name Name of the endpoint requested
|
|
* \param rdata Received message
|
|
*/
|
|
void ast_sip_report_invalid_endpoint(const char *name, pjsip_rx_data *rdata);
|
|
|
|
/*!
|
|
* \brief Send a security event notification for when an ACL check fails
|
|
*
|
|
* \param endpoint Pointer to the endpoint in use
|
|
* \param rdata Received message
|
|
* \param name Name of the ACL
|
|
*/
|
|
void ast_sip_report_failed_acl(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata, const char *name);
|
|
|
|
/*!
|
|
* \brief Send a security event notification for when a challenge response has failed
|
|
*
|
|
* \param endpoint Pointer to the endpoint in use
|
|
* \param rdata Received message
|
|
*/
|
|
void ast_sip_report_auth_failed_challenge_response(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata);
|
|
|
|
/*!
|
|
* \brief Send a security event notification for when authentication succeeds
|
|
*
|
|
* \param endpoint Pointer to the endpoint in use
|
|
* \param rdata Received message
|
|
*/
|
|
void ast_sip_report_auth_success(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata);
|
|
|
|
/*!
|
|
* \brief Send a security event notification for when an authentication challenge is sent
|
|
*
|
|
* \param endpoint Pointer to the endpoint in use
|
|
* \param rdata Received message
|
|
* \param tdata Sent message
|
|
*/
|
|
void ast_sip_report_auth_challenge_sent(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata, pjsip_tx_data *tdata);
|
|
|
|
/*!
|
|
* \brief Send a security event notification for when a request is not supported
|
|
*
|
|
* \param endpoint Pointer to the endpoint in use
|
|
* \param rdata Received message
|
|
* \param req_type the type of request
|
|
*/
|
|
void ast_sip_report_req_no_support(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata,
|
|
const char* req_type);
|
|
|
|
/*!
|
|
* \brief Send a security event notification for when a memory limit is hit.
|
|
*
|
|
* \param endpoint Pointer to the endpoint in use
|
|
* \param rdata Received message
|
|
*/
|
|
void ast_sip_report_mem_limit(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata);
|
|
|
|
int ast_sip_add_global_request_header(const char *name, const char *value, int replace);
|
|
int ast_sip_add_global_response_header(const char *name, const char *value, int replace);
|
|
|
|
/*!
|
|
* \brief Retrieves the value associated with the given key.
|
|
*
|
|
* \param ht the hash table/dictionary to search
|
|
* \param key the key to find
|
|
*
|
|
* \retval the value associated with the key, NULL otherwise.
|
|
*/
|
|
void *ast_sip_dict_get(void *ht, const char *key);
|
|
|
|
/*!
|
|
* \brief Using the dictionary stored in mod_data array at a given id,
|
|
* retrieve the value associated with the given key.
|
|
*
|
|
* \param mod_data a module data array
|
|
* \param id the mod_data array index
|
|
* \param key the key to find
|
|
*
|
|
* \retval the value associated with the key, NULL otherwise.
|
|
*/
|
|
#define ast_sip_mod_data_get(mod_data, id, key) \
|
|
ast_sip_dict_get(mod_data[id], key)
|
|
|
|
/*!
|
|
* \brief Set the value for the given key.
|
|
*
|
|
* Note - if the hash table does not exist one is created first, the key/value
|
|
* pair is set, and the hash table returned.
|
|
*
|
|
* \param pool the pool to allocate memory in
|
|
* \param ht the hash table/dictionary in which to store the key/value pair
|
|
* \param key the key to associate a value with
|
|
* \param val the value to associate with a key
|
|
*
|
|
* \retval the given, or newly created, hash table.
|
|
*/
|
|
void *ast_sip_dict_set(pj_pool_t* pool, void *ht,
|
|
const char *key, void *val);
|
|
|
|
/*!
|
|
* \brief Utilizing a mod_data array for a given id, set the value
|
|
* associated with the given key.
|
|
*
|
|
* For a given structure's mod_data array set the element indexed by id to
|
|
* be a dictionary containing the key/val pair.
|
|
*
|
|
* \param pool a memory allocation pool
|
|
* \param mod_data a module data array
|
|
* \param id the mod_data array index
|
|
* \param key the key to find
|
|
* \param val the value to associate with a key
|
|
*/
|
|
#define ast_sip_mod_data_set(pool, mod_data, id, key, val) \
|
|
mod_data[id] = ast_sip_dict_set(pool, mod_data[id], key, val)
|
|
|
|
/*!
|
|
* \brief For every contact on an AOR call the given 'on_contact' handler.
|
|
*
|
|
* \param aor the aor containing a list of contacts to iterate
|
|
* \param on_contact callback on each contact on an AOR. The object
|
|
* received by the callback will be a ast_sip_contact_wrapper structure.
|
|
* \param arg user data passed to handler
|
|
* \retval 0 Success, non-zero on failure
|
|
*/
|
|
int ast_sip_for_each_contact(const struct ast_sip_aor *aor,
|
|
ao2_callback_fn on_contact, void *arg);
|
|
|
|
/*!
|
|
* \brief Handler used to convert a contact to a string.
|
|
*
|
|
* \param object the ast_sip_aor_contact_pair containing a list of contacts to iterate and the contact
|
|
* \param arg user data passed to handler
|
|
* \param flags
|
|
* \retval 0 Success, non-zero on failure
|
|
*/
|
|
int ast_sip_contact_to_str(void *object, void *arg, int flags);
|
|
|
|
/*!
|
|
* \brief For every aor in the comma separated aors string call the
|
|
* given 'on_aor' handler.
|
|
*
|
|
* \param aors a comma separated list of aors
|
|
* \param on_aor callback for each aor
|
|
* \param arg user data passed to handler
|
|
* \retval 0 Success, non-zero on failure
|
|
*/
|
|
int ast_sip_for_each_aor(const char *aors, ao2_callback_fn on_aor, void *arg);
|
|
|
|
/*!
|
|
* \brief For every auth in the array call the given 'on_auth' handler.
|
|
*
|
|
* \param array an array of auths
|
|
* \param on_auth callback for each auth
|
|
* \param arg user data passed to handler
|
|
* \retval 0 Success, non-zero on failure
|
|
*/
|
|
int ast_sip_for_each_auth(const struct ast_sip_auth_vector *array,
|
|
ao2_callback_fn on_auth, void *arg);
|
|
|
|
/*!
|
|
* \brief Converts the given auth type to a string
|
|
*
|
|
* \param type the auth type to convert
|
|
* \retval a string representative of the auth type
|
|
*/
|
|
const char *ast_sip_auth_type_to_str(enum ast_sip_auth_type type);
|
|
|
|
/*!
|
|
* \brief Converts an auths array to a string of comma separated values
|
|
*
|
|
* \param auths an auth array
|
|
* \param buf the string buffer to write the object data
|
|
* \retval 0 Success, non-zero on failure
|
|
*/
|
|
int ast_sip_auths_to_str(const struct ast_sip_auth_vector *auths, char **buf);
|
|
|
|
/*!
|
|
* \brief AMI variable container
|
|
*/
|
|
struct ast_sip_ami {
|
|
/*! Manager session */
|
|
struct mansession *s;
|
|
/*! Manager message */
|
|
const struct message *m;
|
|
/*! Manager Action ID */
|
|
const char *action_id;
|
|
/*! user specified argument data */
|
|
void *arg;
|
|
/*! count of objects */
|
|
int count;
|
|
};
|
|
|
|
/*!
|
|
* \brief Creates a string to store AMI event data in.
|
|
*
|
|
* \param event the event to set
|
|
* \param ami AMI session and message container
|
|
* \retval an initialized ast_str or NULL on error.
|
|
*/
|
|
struct ast_str *ast_sip_create_ami_event(const char *event,
|
|
struct ast_sip_ami *ami);
|
|
|
|
/*!
|
|
* \brief An entity responsible formatting endpoint information.
|
|
*/
|
|
struct ast_sip_endpoint_formatter {
|
|
/*!
|
|
* \brief Callback used to format endpoint information over AMI.
|
|
*/
|
|
int (*format_ami)(const struct ast_sip_endpoint *endpoint,
|
|
struct ast_sip_ami *ami);
|
|
AST_RWLIST_ENTRY(ast_sip_endpoint_formatter) next;
|
|
};
|
|
|
|
/*!
|
|
* \brief Register an endpoint formatter.
|
|
*
|
|
* \param obj the formatter to register
|
|
*/
|
|
void ast_sip_register_endpoint_formatter(struct ast_sip_endpoint_formatter *obj);
|
|
|
|
/*!
|
|
* \brief Unregister an endpoint formatter.
|
|
*
|
|
* \param obj the formatter to unregister
|
|
*/
|
|
void ast_sip_unregister_endpoint_formatter(struct ast_sip_endpoint_formatter *obj);
|
|
|
|
/*!
|
|
* \brief Converts a sorcery object to a string of object properties.
|
|
*
|
|
* \param obj the sorcery object to convert
|
|
* \param buf the string buffer to write the object data
|
|
* \retval 0 Success, non-zero on failure
|
|
*/
|
|
int ast_sip_sorcery_object_to_ami(const void *obj, struct ast_str **buf);
|
|
|
|
/*!
|
|
* \brief Formats the endpoint and sends over AMI.
|
|
*
|
|
* \param endpoint the endpoint to format and send
|
|
* \param ami AMI variable container
|
|
* \param count the number of formatters operated on
|
|
* \retval 0 Success, otherwise non-zero on error
|
|
*/
|
|
int ast_sip_format_endpoint_ami(struct ast_sip_endpoint *endpoint,
|
|
struct ast_sip_ami *ami, int *count);
|
|
|
|
/*!
|
|
* \brief Formats the contact and sends over AMI.
|
|
*
|
|
* \param obj a pointer an ast_sip_contact_wrapper structure
|
|
* \param arg a pointer to an ast_sip_ami structure
|
|
* \param flags ignored
|
|
* \retval 0 Success, otherwise non-zero on error
|
|
*/
|
|
int ast_sip_format_contact_ami(void *obj, void *arg, int flags);
|
|
|
|
/*!
|
|
* \brief Format auth details for AMI.
|
|
*
|
|
* \param auths an auth array
|
|
* \param ami ami variable container
|
|
* \retval 0 Success, non-zero on failure
|
|
*/
|
|
int ast_sip_format_auths_ami(const struct ast_sip_auth_vector *auths,
|
|
struct ast_sip_ami *ami);
|
|
|
|
/*!
|
|
* \brief Retrieve the endpoint snapshot for an endpoint
|
|
*
|
|
* \param endpoint The endpoint whose snapshot is to be retrieved.
|
|
* \retval The endpoint snapshot
|
|
*/
|
|
struct ast_endpoint_snapshot *ast_sip_get_endpoint_snapshot(
|
|
const struct ast_sip_endpoint *endpoint);
|
|
|
|
/*!
|
|
* \brief Retrieve the device state for an endpoint.
|
|
*
|
|
* \param endpoint The endpoint whose state is to be retrieved.
|
|
* \retval The device state.
|
|
*/
|
|
const char *ast_sip_get_device_state(const struct ast_sip_endpoint *endpoint);
|
|
|
|
/*!
|
|
* \brief For every channel snapshot on an endpoint snapshot call the given
|
|
* 'on_channel_snapshot' handler.
|
|
*
|
|
* \param endpoint_snapshot snapshot of an endpoint
|
|
* \param on_channel_snapshot callback for each channel snapshot
|
|
* \param arg user data passed to handler
|
|
* \retval 0 Success, non-zero on failure
|
|
*/
|
|
int ast_sip_for_each_channel_snapshot(const struct ast_endpoint_snapshot *endpoint_snapshot,
|
|
ao2_callback_fn on_channel_snapshot,
|
|
void *arg);
|
|
|
|
/*!
|
|
* \brief For every channel snapshot on an endpoint all the given
|
|
* 'on_channel_snapshot' handler.
|
|
*
|
|
* \param endpoint endpoint
|
|
* \param on_channel_snapshot callback for each channel snapshot
|
|
* \param arg user data passed to handler
|
|
* \retval 0 Success, non-zero on failure
|
|
*/
|
|
int ast_sip_for_each_channel(const struct ast_sip_endpoint *endpoint,
|
|
ao2_callback_fn on_channel_snapshot,
|
|
void *arg);
|
|
|
|
enum ast_sip_supplement_priority {
|
|
/*! Top priority. Supplements with this priority are those that need to run before any others */
|
|
AST_SIP_SUPPLEMENT_PRIORITY_FIRST = 0,
|
|
/*! Channel creation priority.
|
|
* chan_pjsip creates a channel at this priority. If your supplement depends on being run before
|
|
* or after channel creation, then set your priority to be lower or higher than this value.
|
|
*/
|
|
AST_SIP_SUPPLEMENT_PRIORITY_CHANNEL = 1000000,
|
|
/*! Lowest priority. Supplements with this priority should be run after all other supplements */
|
|
AST_SIP_SUPPLEMENT_PRIORITY_LAST = INT_MAX,
|
|
};
|
|
|
|
/*!
|
|
* \brief A supplement to SIP message processing
|
|
*
|
|
* These can be registered by any module in order to add
|
|
* processing to incoming and outgoing SIP out of dialog
|
|
* requests and responses
|
|
*/
|
|
struct ast_sip_supplement {
|
|
/*! Method on which to call the callbacks. If NULL, call on all methods */
|
|
const char *method;
|
|
/*! Priority for this supplement. Lower numbers are visited before higher numbers */
|
|
enum ast_sip_supplement_priority priority;
|
|
/*!
|
|
* \brief Called on incoming SIP request
|
|
* This method can indicate a failure in processing in its return. If there
|
|
* is a failure, it is required that this method sends a response to the request.
|
|
* This method is always called from a SIP servant thread.
|
|
*
|
|
* \note
|
|
* The following PJSIP methods will not work properly:
|
|
* pjsip_rdata_get_dlg()
|
|
* pjsip_rdata_get_tsx()
|
|
* The reason is that the rdata passed into this function is a cloned rdata structure,
|
|
* and its module data is not copied during the cloning operation.
|
|
* If you need to get the dialog, you can get it via session->inv_session->dlg.
|
|
*
|
|
* \note
|
|
* There is no guarantee that a channel will be present on the session when this is called.
|
|
*/
|
|
int (*incoming_request)(struct ast_sip_endpoint *endpoint, struct pjsip_rx_data *rdata);
|
|
/*!
|
|
* \brief Called on an incoming SIP response
|
|
* This method is always called from a SIP servant thread.
|
|
*
|
|
* \note
|
|
* The following PJSIP methods will not work properly:
|
|
* pjsip_rdata_get_dlg()
|
|
* pjsip_rdata_get_tsx()
|
|
* The reason is that the rdata passed into this function is a cloned rdata structure,
|
|
* and its module data is not copied during the cloning operation.
|
|
* If you need to get the dialog, you can get it via session->inv_session->dlg.
|
|
*
|
|
* \note
|
|
* There is no guarantee that a channel will be present on the session when this is called.
|
|
*/
|
|
void (*incoming_response)(struct ast_sip_endpoint *endpoint, struct pjsip_rx_data *rdata);
|
|
/*!
|
|
* \brief Called on an outgoing SIP request
|
|
* This method is always called from a SIP servant thread.
|
|
*/
|
|
void (*outgoing_request)(struct ast_sip_endpoint *endpoint, struct ast_sip_contact *contact, struct pjsip_tx_data *tdata);
|
|
/*!
|
|
* \brief Called on an outgoing SIP response
|
|
* This method is always called from a SIP servant thread.
|
|
*/
|
|
void (*outgoing_response)(struct ast_sip_endpoint *endpoint, struct ast_sip_contact *contact, struct pjsip_tx_data *tdata);
|
|
/*! Next item in the list */
|
|
AST_LIST_ENTRY(ast_sip_supplement) next;
|
|
};
|
|
|
|
/*!
|
|
* \brief Register a supplement to SIP out of dialog processing
|
|
*
|
|
* This allows for someone to insert themselves in the processing of out
|
|
* of dialog SIP requests and responses. This, for example could allow for
|
|
* a module to set channel data based on headers in an incoming message.
|
|
* Similarly, a module could reject an incoming request if desired.
|
|
*
|
|
* \param supplement The supplement to register
|
|
*/
|
|
void ast_sip_register_supplement(struct ast_sip_supplement *supplement);
|
|
|
|
/*!
|
|
* \brief Unregister a an supplement to SIP out of dialog processing
|
|
*
|
|
* \param supplement The supplement to unregister
|
|
*/
|
|
void ast_sip_unregister_supplement(struct ast_sip_supplement *supplement);
|
|
|
|
/*!
|
|
* \brief Retrieve the global MWI taskprocessor high water alert trigger level.
|
|
*
|
|
* \since 13.12.0
|
|
*
|
|
* \retval the system MWI taskprocessor high water alert trigger level
|
|
*/
|
|
unsigned int ast_sip_get_mwi_tps_queue_high(void);
|
|
|
|
/*!
|
|
* \brief Retrieve the global MWI taskprocessor low water clear alert level.
|
|
*
|
|
* \since 13.12.0
|
|
*
|
|
* \retval the system MWI taskprocessor low water clear alert level
|
|
*/
|
|
int ast_sip_get_mwi_tps_queue_low(void);
|
|
|
|
/*!
|
|
* \brief Retrieve the global setting 'disable sending unsolicited mwi on startup'.
|
|
* \since 13.12.0
|
|
*
|
|
* \retval non zero if disable.
|
|
*/
|
|
unsigned int ast_sip_get_mwi_disable_initial_unsolicited(void);
|
|
|
|
/*!
|
|
* \brief Retrieve the global setting 'allow_sending_180_after_183'.
|
|
*
|
|
* \retval non zero if disable.
|
|
*/
|
|
unsigned int ast_sip_get_allow_sending_180_after_183(void);
|
|
|
|
/*!
|
|
* \brief Retrieve the global setting 'use_callerid_contact'.
|
|
* \since 13.24.0
|
|
*
|
|
* \retval non zero if CALLERID(num) is to be used as the default username in the contact
|
|
*/
|
|
unsigned int ast_sip_get_use_callerid_contact(void);
|
|
|
|
/*!
|
|
* \brief Retrieve the global setting 'norefersub'.
|
|
*
|
|
* \retval non zero if norefersub is to be sent in "Supported" Headers
|
|
*/
|
|
unsigned int ast_sip_get_norefersub(void);
|
|
|
|
/*!
|
|
* \brief Retrieve the global setting 'ignore_uri_user_options'.
|
|
* \since 13.12.0
|
|
*
|
|
* \retval non zero if ignore the user field options.
|
|
*/
|
|
unsigned int ast_sip_get_ignore_uri_user_options(void);
|
|
|
|
/*!
|
|
* \brief Retrieve the global setting 'send_contact_status_on_update_registration'.
|
|
* \since 16.2.0
|
|
*
|
|
* \retval non zero if need to send AMI ContactStatus event when a contact is updated.
|
|
*/
|
|
unsigned int ast_sip_get_send_contact_status_on_update_registration(void);
|
|
|
|
|
|
/*!
|
|
* \brief Truncate the URI user field options string if enabled.
|
|
* \since 13.12.0
|
|
*
|
|
* \param str URI user field string to truncate if enabled
|
|
*
|
|
* \details
|
|
* We need to be able to handle URI's looking like
|
|
* "sip:1235557890;phone-context=national@x.x.x.x;user=phone"
|
|
*
|
|
* Where the URI user field is:
|
|
* "1235557890;phone-context=national"
|
|
*
|
|
* When truncated the string will become:
|
|
* "1235557890"
|
|
*/
|
|
#define AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(str) \
|
|
do { \
|
|
char *__semi = strchr((str), ';'); \
|
|
if (__semi && ast_sip_get_ignore_uri_user_options()) { \
|
|
*__semi = '\0'; \
|
|
} \
|
|
} while (0)
|
|
|
|
/*!
|
|
* \brief Retrieve the system debug setting (yes|no|host).
|
|
*
|
|
* \note returned string needs to be de-allocated by caller.
|
|
*
|
|
* \retval the system debug setting.
|
|
*/
|
|
char *ast_sip_get_debug(void);
|
|
|
|
/*!
|
|
* \brief Retrieve the global regcontext setting.
|
|
*
|
|
* \since 13.8.0
|
|
*
|
|
* \note returned string needs to be de-allocated by caller.
|
|
*
|
|
* \retval the global regcontext setting
|
|
*/
|
|
char *ast_sip_get_regcontext(void);
|
|
|
|
/*!
|
|
* \brief Retrieve the global endpoint_identifier_order setting.
|
|
*
|
|
* Specifies the order by which endpoint identifiers should be regarded.
|
|
*
|
|
* \retval the global endpoint_identifier_order value
|
|
*/
|
|
char *ast_sip_get_endpoint_identifier_order(void);
|
|
|
|
/*!
|
|
* \brief Retrieve the default voicemail extension.
|
|
* \since 13.9.0
|
|
*
|
|
* \note returned string needs to be de-allocated by caller.
|
|
*
|
|
* \retval the default voicemail extension
|
|
*/
|
|
char *ast_sip_get_default_voicemail_extension(void);
|
|
|
|
/*!
|
|
* \brief Retrieve the global default realm.
|
|
*
|
|
* This is the value placed in outbound challenges' realm if there
|
|
* is no better option (such as an auth-configured realm).
|
|
*
|
|
* \param[out] realm The default realm
|
|
* \param size The buffer size of realm
|
|
*/
|
|
void ast_sip_get_default_realm(char *realm, size_t size);
|
|
|
|
/*!
|
|
* \brief Retrieve the global default from user.
|
|
*
|
|
* This is the value placed in outbound requests' From header if there
|
|
* is no better option (such as an endpoint-configured from_user or
|
|
* caller ID number).
|
|
*
|
|
* \param[out] from_user The default from user
|
|
* \param size The buffer size of from_user
|
|
*/
|
|
void ast_sip_get_default_from_user(char *from_user, size_t size);
|
|
|
|
/*!
|
|
* \brief Retrieve the system keep alive interval setting.
|
|
*
|
|
* \retval the keep alive interval.
|
|
*/
|
|
unsigned int ast_sip_get_keep_alive_interval(void);
|
|
|
|
/*!
|
|
* \brief Retrieve the system contact expiration check interval setting.
|
|
*
|
|
* \retval the contact expiration check interval.
|
|
*/
|
|
unsigned int ast_sip_get_contact_expiration_check_interval(void);
|
|
|
|
/*!
|
|
* \brief Retrieve the system setting 'disable multi domain'.
|
|
* \since 13.9.0
|
|
*
|
|
* \retval non zero if disable multi domain.
|
|
*/
|
|
unsigned int ast_sip_get_disable_multi_domain(void);
|
|
|
|
/*!
|
|
* \brief Retrieve the system max initial qualify time.
|
|
*
|
|
* \retval the maximum initial qualify time.
|
|
*/
|
|
unsigned int ast_sip_get_max_initial_qualify_time(void);
|
|
|
|
/*!
|
|
* \brief translate ast_sip_contact_status_type to character string.
|
|
*
|
|
* \retval the character string equivalent.
|
|
*/
|
|
|
|
const char *ast_sip_get_contact_status_label(const enum ast_sip_contact_status_type status);
|
|
const char *ast_sip_get_contact_short_status_label(const enum ast_sip_contact_status_type status);
|
|
|
|
/*!
|
|
* \brief Set a request to use the next value in the list of resolved addresses.
|
|
*
|
|
* \param tdata the tx data from the original request
|
|
* \retval 0 No more addresses to try
|
|
* \retval 1 The request was successfully re-intialized
|
|
*/
|
|
int ast_sip_failover_request(pjsip_tx_data *tdata);
|
|
|
|
/*!
|
|
* \brief Retrieve the local host address in IP form
|
|
*
|
|
* \param af The address family to retrieve
|
|
* \param addr A place to store the local host address
|
|
*
|
|
* \retval 0 success
|
|
* \retval -1 failure
|
|
*
|
|
* \since 13.6.0
|
|
*/
|
|
int ast_sip_get_host_ip(int af, pj_sockaddr *addr);
|
|
|
|
/*!
|
|
* \brief Retrieve the local host address in string form
|
|
*
|
|
* \param af The address family to retrieve
|
|
*
|
|
* \retval non-NULL success
|
|
* \retval NULL failure
|
|
*
|
|
* \since 13.6.0
|
|
*
|
|
* \note An empty string may be returned if the address family is valid but no local address exists
|
|
*/
|
|
const char *ast_sip_get_host_ip_string(int af);
|
|
|
|
/*!
|
|
* \brief Return the size of the SIP threadpool's task queue
|
|
* \since 13.7.0
|
|
*/
|
|
long ast_sip_threadpool_queue_size(void);
|
|
|
|
/*!
|
|
* \brief Retrieve the SIP threadpool object
|
|
*/
|
|
struct ast_threadpool *ast_sip_threadpool(void);
|
|
|
|
/*!
|
|
* \brief Retrieve transport state
|
|
* \since 13.7.1
|
|
*
|
|
* \param transport_id
|
|
* \retval transport_state
|
|
*
|
|
* \note ao2_cleanup(...) or ao2_ref(..., -1) must be called on the returned object
|
|
*/
|
|
struct ast_sip_transport_state *ast_sip_get_transport_state(const char *transport_id);
|
|
|
|
/*!
|
|
* \brief Return the SIP URI of the Contact header
|
|
*
|
|
* \param tdata
|
|
* \retval Pointer to SIP URI of Contact
|
|
* \retval NULL if Contact header not found or not a SIP(S) URI
|
|
*
|
|
* \note Do not free the returned object.
|
|
*/
|
|
pjsip_sip_uri *ast_sip_get_contact_sip_uri(pjsip_tx_data *tdata);
|
|
|
|
/*!
|
|
* \brief Returns the transport state currently in use based on request transport details
|
|
*
|
|
* \param details
|
|
* \retval transport_state
|
|
*
|
|
* \note ao2_cleanup(...) or ao2_ref(..., -1) must be called on the returned object
|
|
*/
|
|
struct ast_sip_transport_state *ast_sip_find_transport_state_in_use(struct ast_sip_request_transport_details *details);
|
|
|
|
/*!
|
|
* \brief Sets request transport details based on tdata
|
|
*
|
|
* \param details pre-allocated request transport details to set
|
|
* \param tdata
|
|
* \param use_ipv6 if non-zero, ipv6 transports will be considered
|
|
* \retval 0 success
|
|
* \retval -1 failure
|
|
*/
|
|
int ast_sip_set_request_transport_details(struct ast_sip_request_transport_details *details, pjsip_tx_data *tdata, int use_ipv6);
|
|
|
|
/*!
|
|
* \brief Replace domain and port of SIP URI to point to (external) signaling address of this Asterisk instance
|
|
*
|
|
* \param uri
|
|
* \param tdata
|
|
*
|
|
* \retval 0 success
|
|
* \retval -1 failure
|
|
*
|
|
* \note Uses domain and port in Contact header if it exists, otherwise the local URI of the dialog is used if the
|
|
* message is sent within the context of a dialog. Further, NAT settings are considered - i.e. if the target
|
|
* is not in the localnet, the external_signaling_address and port are used.
|
|
*/
|
|
int ast_sip_rewrite_uri_to_local(pjsip_sip_uri *uri, pjsip_tx_data *tdata);
|
|
|
|
/*!
|
|
* \brief Retrieves all transport states
|
|
* \since 13.7.1
|
|
*
|
|
* \retval ao2_container
|
|
*
|
|
* \note ao2_cleanup(...) or ao2_ref(..., -1) must be called on the returned object
|
|
*/
|
|
struct ao2_container *ast_sip_get_transport_states(void);
|
|
|
|
/*!
|
|
* \brief Sets pjsip_tpselector from ast_sip_transport
|
|
* \since 13.8.0
|
|
*
|
|
* \param transport The transport to be used
|
|
* \param selector The selector to be populated
|
|
* \retval 0 success
|
|
* \retval -1 failure
|
|
*
|
|
* \note The transport selector must be unreffed using ast_sip_tpselector_unref
|
|
*/
|
|
int ast_sip_set_tpselector_from_transport(const struct ast_sip_transport *transport, pjsip_tpselector *selector);
|
|
|
|
/*!
|
|
* \brief Sets pjsip_tpselector from ast_sip_transport
|
|
* \since 13.8.0
|
|
*
|
|
* \param transport_name The name of the transport to be used
|
|
* \param selector The selector to be populated
|
|
* \retval 0 success
|
|
* \retval -1 failure
|
|
*
|
|
* \note The transport selector must be unreffed using ast_sip_tpselector_unref
|
|
*/
|
|
int ast_sip_set_tpselector_from_transport_name(const char *transport_name, pjsip_tpselector *selector);
|
|
|
|
/*!
|
|
* \brief Unreference a pjsip_tpselector
|
|
* \since 17.0.0
|
|
*
|
|
* \param selector The selector to be unreffed
|
|
*/
|
|
void ast_sip_tpselector_unref(pjsip_tpselector *selector);
|
|
|
|
/*!
|
|
* \brief Sets the PJSIP transport on a child transport
|
|
* \since 17.0.0
|
|
*
|
|
* \param transport_name The name of the transport to be updated
|
|
* \param transport The PJSIP transport
|
|
* \retval 0 success
|
|
* \retval -1 failure
|
|
*/
|
|
int ast_sip_transport_state_set_transport(const char *transport_name, pjsip_transport *transport);
|
|
|
|
/*!
|
|
* \brief Sets the P-Preferred-Identity on a child transport
|
|
* \since 17.0.0
|
|
*
|
|
* \param transport_name The name of the transport to be set on
|
|
* \param identity The P-Preferred-Identity to use on requests on this transport
|
|
* \retval 0 success
|
|
* \retval -1 failure
|
|
*/
|
|
int ast_sip_transport_state_set_preferred_identity(const char *transport_name, const char *identity);
|
|
|
|
/*!
|
|
* \brief Sets the service routes on a child transport
|
|
* \since 17.0.0
|
|
*
|
|
* \param transport_name The name of the transport to be set on
|
|
* \param service_routes A vector of service routes
|
|
* \retval 0 success
|
|
* \retval -1 failure
|
|
*
|
|
* \note This assumes ownership of the service routes in both success and failure scenarios
|
|
*/
|
|
int ast_sip_transport_state_set_service_routes(const char *transport_name, struct ast_sip_service_route_vector *service_routes);
|
|
|
|
/*!
|
|
* \brief Apply the configuration for a transport to an outgoing message
|
|
* \since 17.0.0
|
|
*
|
|
* \param transport_name The name of the transport to apply configuration from
|
|
* \param tdata The SIP message
|
|
*/
|
|
void ast_sip_message_apply_transport(const char *transport_name, pjsip_tx_data *tdata);
|
|
|
|
/*!
|
|
* \brief Allocate a vector of service routes
|
|
* \since 17.0.0
|
|
*
|
|
* \retval non-NULL success
|
|
* \retval NULL failure
|
|
*/
|
|
struct ast_sip_service_route_vector *ast_sip_service_route_vector_alloc(void);
|
|
|
|
/*!
|
|
* \brief Destroy a vector of service routes
|
|
* \since 17.0.0
|
|
*
|
|
* \param service_routes A vector of service routes
|
|
*/
|
|
void ast_sip_service_route_vector_destroy(struct ast_sip_service_route_vector *service_routes);
|
|
|
|
/*!
|
|
* \brief Set the ID for a connected line update
|
|
*
|
|
* \retval -1 on failure, 0 on success
|
|
*/
|
|
int ast_sip_set_id_connected_line(struct pjsip_rx_data *rdata, struct ast_party_id *id);
|
|
|
|
/*!
|
|
* \brief Set the ID from an INVITE
|
|
*
|
|
* \param rdata
|
|
* \param id ID structure to fill
|
|
* \param default_id Default ID structure with data to use (for non-trusted endpoints)
|
|
* \param trust_inbound Whether or not the endpoint is trusted (controls whether PAI or RPID can be used)
|
|
*
|
|
* \retval -1 on failure, 0 on success
|
|
*/
|
|
int ast_sip_set_id_from_invite(struct pjsip_rx_data *rdata, struct ast_party_id *id, struct ast_party_id *default_id, int trust_inbound);
|
|
|
|
/*!
|
|
* \brief Set name and number information on an identity header.
|
|
*
|
|
* \param pool Memory pool to use for string duplication
|
|
* \param id_hdr A From, P-Asserted-Identity, or Remote-Party-ID header to modify
|
|
* \param id The identity information to apply to the header
|
|
*/
|
|
void ast_sip_modify_id_header(pj_pool_t *pool, pjsip_fromto_hdr *id_hdr,
|
|
const struct ast_party_id *id);
|
|
|
|
/*!
|
|
* \brief Retrieves an endpoint and URI from the "to" string.
|
|
*
|
|
* This URI is used as the Request URI.
|
|
*
|
|
* Expects the given 'to' to be in one of the following formats:
|
|
* Why we allow so many is a mystery.
|
|
*
|
|
* Basic:
|
|
*
|
|
* endpoint : We'll get URI from the default aor/contact
|
|
* endpoint/aor : We'll get the URI from the specific aor/contact
|
|
* endpoint@domain : We toss the domain part and just use the endpoint
|
|
*
|
|
* These all use the endpoint and specified URI:
|
|
* \verbatim
|
|
endpoint/<sip[s]:host>
|
|
endpoint/<sip[s]:user@host>
|
|
endpoint/"Bob" <sip[s]:host>
|
|
endpoint/"Bob" <sip[s]:user@host>
|
|
endpoint/sip[s]:host
|
|
endpoint/sip[s]:user@host
|
|
endpoint/host
|
|
endpoint/user@host
|
|
\endverbatim
|
|
*
|
|
* These all use the default endpoint and specified URI:
|
|
* \verbatim
|
|
<sip[s]:host>
|
|
<sip[s]:user@host>
|
|
"Bob" <sip[s]:host>
|
|
"Bob" <sip[s]:user@host>
|
|
sip[s]:host
|
|
sip[s]:user@host
|
|
\endverbatim
|
|
*
|
|
* These use the default endpoint and specified host:
|
|
* \verbatim
|
|
host
|
|
user@host
|
|
\endverbatim
|
|
*
|
|
* This form is similar to a dialstring:
|
|
* \verbatim
|
|
PJSIP/user@endpoint
|
|
\endverbatim
|
|
*
|
|
* In this case, the user will be added to the endpoint contact's URI.
|
|
* If the contact URI already has a user, it will be replaced.
|
|
*
|
|
* The ones that have the sip[s] scheme are the easiest to parse.
|
|
* The rest all have some issue.
|
|
*
|
|
* endpoint vs host : We have to test for endpoint first
|
|
* endpoint/aor vs endpoint/host : We have to test for aor first
|
|
* What if there's an aor with the same
|
|
* name as the host?
|
|
* endpoint@domain vs user@host : We have to test for endpoint first.
|
|
* What if there's an endpoint with the
|
|
* same name as the user?
|
|
*
|
|
* \param to 'To' field with possible endpoint
|
|
* \param get_default_outbound If nonzero, try to retrieve the default
|
|
* outbound endpoint if no endpoint was found.
|
|
* Otherwise, return NULL if no endpoint was found.
|
|
* \param uri Pointer to a char* which will be set to the URI.
|
|
* Always must be ast_free'd by the caller - even if the return value is NULL!
|
|
*
|
|
* \note The logic below could probably be condensed but then it wouldn't be
|
|
* as clear.
|
|
*/
|
|
struct ast_sip_endpoint *ast_sip_get_endpoint(const char *to, int get_default_outbound, char **uri);
|
|
|
|
/*!
|
|
* \brief Replace the To URI in the tdata with the supplied one
|
|
*
|
|
* \param tdata the outbound message data structure
|
|
* \param to URI to replace the To URI with. Must be a valid SIP URI.
|
|
*
|
|
* \retval 0: success, -1: failure
|
|
*/
|
|
int ast_sip_update_to_uri(pjsip_tx_data *tdata, const char *to);
|
|
|
|
/*!
|
|
* \brief Overwrite fields in the outbound 'From' header
|
|
*
|
|
* The outbound 'From' header is created/added in ast_sip_create_request with
|
|
* default data. If available that data may be info specified in the 'from_user'
|
|
* and 'from_domain' options found on the endpoint. That information will be
|
|
* overwritten with data in the given 'from' parameter.
|
|
*
|
|
* \param tdata the outbound message data structure
|
|
* \param from info to copy into the header.
|
|
* Can be either a SIP URI, or in the format user[@domain]
|
|
*
|
|
* \retval 0: success, -1: failure
|
|
*/
|
|
int ast_sip_update_from(pjsip_tx_data *tdata, char *from);
|
|
|
|
/*!
|
|
* \brief Retrieve the unidentified request security event thresholds
|
|
* \since 13.8.0
|
|
*
|
|
* \param count The maximum number of unidentified requests per source ip to accumulate before emitting a security event
|
|
* \param period The period in seconds over which to accumulate unidentified requests
|
|
* \param prune_interval The interval in seconds at which expired entries will be pruned
|
|
*/
|
|
void ast_sip_get_unidentified_request_thresholds(unsigned int *count, unsigned int *period,
|
|
unsigned int *prune_interval);
|
|
|
|
/*!
|
|
* \brief Get the transport name from an endpoint or request uri
|
|
* \since 13.15.0
|
|
*
|
|
* \param endpoint
|
|
* \param sip_uri
|
|
* \param buf Buffer to receive transport name
|
|
* \param buf_len Buffer length
|
|
*
|
|
* \retval 0 Success
|
|
* \retval -1 Failure
|
|
*
|
|
* \note
|
|
* If endpoint->transport is not NULL, it is returned in buf.
|
|
* Otherwise if sip_uri has an 'x-ast-txp' parameter AND the sip_uri host is
|
|
* an ip4 or ip6 address, its value is returned,
|
|
*/
|
|
int ast_sip_get_transport_name(const struct ast_sip_endpoint *endpoint,
|
|
pjsip_sip_uri *sip_uri, char *buf, size_t buf_len);
|
|
|
|
/*!
|
|
* \brief Sets pjsip_tpselector from an endpoint or uri
|
|
* \since 13.15.0
|
|
*
|
|
* \param endpoint If endpoint->transport is set, it's used
|
|
* \param sip_uri If sip_uri contains a x-ast-txp parameter, it's used
|
|
* \param selector The selector to be populated
|
|
*
|
|
* \retval 0 success
|
|
* \retval -1 failure
|
|
*/
|
|
int ast_sip_set_tpselector_from_ep_or_uri(const struct ast_sip_endpoint *endpoint,
|
|
pjsip_sip_uri *sip_uri, pjsip_tpselector *selector);
|
|
|
|
/*!
|
|
* \brief Set the transport on a dialog
|
|
* \since 13.15.0
|
|
*
|
|
* \param endpoint
|
|
* \param dlg
|
|
* \param selector (optional)
|
|
*
|
|
* \note
|
|
* This API calls ast_sip_get_transport_name(endpoint, dlg->target) and if the result is
|
|
* non-NULL, calls pjsip_dlg_set_transport. If 'selector' is non-NULL, it is updated with
|
|
* the selector used.
|
|
*
|
|
* \note
|
|
* It is the responsibility of the caller to unref the passed in selector if one is provided.
|
|
*/
|
|
int ast_sip_dlg_set_transport(const struct ast_sip_endpoint *endpoint, pjsip_dialog *dlg,
|
|
pjsip_tpselector *selector);
|
|
|
|
/*!
|
|
* \brief Convert the DTMF mode enum value into a string
|
|
* \since 13.18.0
|
|
*
|
|
* \param dtmf the dtmf mode
|
|
* \param buf Buffer to receive dtmf mode string
|
|
* \param buf_len Buffer length
|
|
*
|
|
* \retval 0 Success
|
|
* \retval -1 Failure
|
|
*
|
|
*/
|
|
int ast_sip_dtmf_to_str(const enum ast_sip_dtmf_mode dtmf,
|
|
char *buf, size_t buf_len);
|
|
|
|
/*!
|
|
* \brief Convert the DTMF mode name into an enum
|
|
* \since 13.18.0
|
|
*
|
|
* \param dtmf_mode dtmf mode as a string
|
|
*
|
|
* \retval >= 0 The enum value
|
|
* \retval -1 Failure
|
|
*
|
|
*/
|
|
int ast_sip_str_to_dtmf(const char *dtmf_mode);
|
|
|
|
/*!
|
|
* \brief Convert the call codec preference flags to a string
|
|
* \since 18.0.0
|
|
*
|
|
* \param pref the call codec preference setting
|
|
*
|
|
* \returns a constant string with either the setting value or 'unknown'
|
|
* \note Don't try to free the string!
|
|
*
|
|
*/
|
|
const char *ast_sip_call_codec_pref_to_str(struct ast_flags pref);
|
|
|
|
/*!
|
|
* \brief Convert a call codec preference string to preference flags
|
|
* \since 18.0.0
|
|
*
|
|
* \param pref A pointer to an ast_flags structure to receive the preference flags
|
|
* \param pref_str The call codec preference setting string
|
|
* \param is_outgoing Is for outgoing calls?
|
|
*
|
|
* \retval 0 The string was parsed successfully
|
|
* \retval -1 The string option was invalid
|
|
*/
|
|
int ast_sip_call_codec_str_to_pref(struct ast_flags *pref, const char *pref_str, int is_outgoing);
|
|
|
|
/*!
|
|
* \brief Transport shutdown monitor callback.
|
|
* \since 13.18.0
|
|
*
|
|
* \param data User data to know what to do when transport shuts down.
|
|
*
|
|
* \note The callback does not need to care that data is an ao2 object.
|
|
*/
|
|
typedef void (*ast_transport_monitor_shutdown_cb)(void *data);
|
|
|
|
/*!
|
|
* \brief Transport shutdown monitor data matcher
|
|
* \since 13.20.0
|
|
*
|
|
* \param a User data to compare.
|
|
* \param b User data to compare.
|
|
*
|
|
* \retval 1 The data objects match
|
|
* \retval 0 The data objects don't match
|
|
*/
|
|
typedef int (*ast_transport_monitor_data_matcher)(void *a, void *b);
|
|
|
|
enum ast_transport_monitor_reg {
|
|
/*! \brief Successfully registered the transport monitor */
|
|
AST_TRANSPORT_MONITOR_REG_SUCCESS,
|
|
/*! \brief Replaced the already existing transport monitor with new one. */
|
|
AST_TRANSPORT_MONITOR_REG_REPLACED,
|
|
/*!
|
|
* \brief Transport not found to monitor.
|
|
* \note Transport is either already shutdown or is not reliable.
|
|
*/
|
|
AST_TRANSPORT_MONITOR_REG_NOT_FOUND,
|
|
/*! \brief Error while registering transport monitor. */
|
|
AST_TRANSPORT_MONITOR_REG_FAILED,
|
|
};
|
|
|
|
/*!
|
|
* \brief Register a reliable transport shutdown monitor callback.
|
|
* \deprecated Replaced with ast_sip_transport_monitor_register_key().
|
|
* \since 13.20.0
|
|
*
|
|
* \param transport Transport to monitor for shutdown.
|
|
* \param cb Who to call when transport is shutdown.
|
|
* \param ao2_data Data to pass with the callback.
|
|
*
|
|
* \note The data object passed will have its reference count automatically
|
|
* incremented by this call and automatically decremented after the callback
|
|
* runs or when the callback is unregistered.
|
|
*
|
|
* There is no checking for duplicate registrations.
|
|
*
|
|
* \return enum ast_transport_monitor_reg
|
|
*/
|
|
enum ast_transport_monitor_reg ast_sip_transport_monitor_register(pjsip_transport *transport,
|
|
ast_transport_monitor_shutdown_cb cb, void *ao2_data);
|
|
|
|
/*!
|
|
* \brief Register a reliable transport shutdown monitor callback.
|
|
*
|
|
* \param transport_key Key for the transport to monitor for shutdown.
|
|
* Create the key with AST_SIP_MAKE_REMOTE_IPADDR_PORT_STR.
|
|
* \param cb Who to call when transport is shutdown.
|
|
* \param ao2_data Data to pass with the callback.
|
|
*
|
|
* \note The data object passed will have its reference count automatically
|
|
* incremented by this call and automatically decremented after the callback
|
|
* runs or when the callback is unregistered.
|
|
*
|
|
* There is no checking for duplicate registrations.
|
|
*
|
|
* \return enum ast_transport_monitor_reg
|
|
*/
|
|
enum ast_transport_monitor_reg ast_sip_transport_monitor_register_key(
|
|
const char *transport_key, ast_transport_monitor_shutdown_cb cb,
|
|
void *ao2_data);
|
|
|
|
/*!
|
|
* \brief Register a reliable transport shutdown monitor callback replacing any duplicate.
|
|
* \deprecated Replaced with ast_sip_transport_monitor_register_replace_key().
|
|
* \since 13.26.0
|
|
* \since 16.3.0
|
|
*
|
|
* \param transport Transport to monitor for shutdown.
|
|
* \param cb Who to call when transport is shutdown.
|
|
* \param ao2_data Data to pass with the callback.
|
|
* \param matches Matcher function that returns true if data matches a previously
|
|
* registered data object
|
|
*
|
|
* \note The data object passed will have its reference count automatically
|
|
* incremented by this call and automatically decremented after the callback
|
|
* runs or when the callback is unregistered.
|
|
*
|
|
* This function checks for duplicates, and overwrites/replaces the old monitor
|
|
* with the given one.
|
|
*
|
|
* \return enum ast_transport_monitor_reg
|
|
*/
|
|
enum ast_transport_monitor_reg ast_sip_transport_monitor_register_replace(pjsip_transport *transport,
|
|
ast_transport_monitor_shutdown_cb cb, void *ao2_data, ast_transport_monitor_data_matcher matches);
|
|
|
|
/*!
|
|
* \brief Register a reliable transport shutdown monitor callback replacing any duplicate.
|
|
*
|
|
* \param transport_key Key for the transport to monitor for shutdown.
|
|
* Create the key with AST_SIP_MAKE_REMOTE_IPADDR_PORT_STR.
|
|
* \param cb Who to call when transport is shutdown.
|
|
* \param ao2_data Data to pass with the callback.
|
|
* \param matches Matcher function that returns true if data matches a previously
|
|
* registered data object
|
|
*
|
|
* \note The data object passed will have its reference count automatically
|
|
* incremented by this call and automatically decremented after the callback
|
|
* runs or when the callback is unregistered.
|
|
*
|
|
* This function checks for duplicates, and overwrites/replaces the old monitor
|
|
* with the given one.
|
|
*
|
|
* \return enum ast_transport_monitor_reg
|
|
*/
|
|
enum ast_transport_monitor_reg ast_sip_transport_monitor_register_replace_key(
|
|
const char *transport_key, ast_transport_monitor_shutdown_cb cb,
|
|
void *ao2_data, ast_transport_monitor_data_matcher matches);
|
|
|
|
/*!
|
|
* \brief Unregister a reliable transport shutdown monitor
|
|
* \deprecated Replaced with ast_sip_transport_monitor_unregister_key().
|
|
* \since 13.20.0
|
|
*
|
|
* \param transport Transport to monitor for shutdown.
|
|
* \param cb The callback that was used for the original register.
|
|
* \param data Data to pass to the matcher. May be NULL and does NOT need to be an ao2 object.
|
|
* If NULL, all monitors with the provided callback are unregistered.
|
|
* \param matches Matcher function that returns true if data matches the previously
|
|
* registered data object. If NULL, a simple pointer comparison is done.
|
|
*
|
|
* \note The data object passed into the original register will have its reference count
|
|
* automatically decremented.
|
|
*/
|
|
void ast_sip_transport_monitor_unregister(pjsip_transport *transport,
|
|
ast_transport_monitor_shutdown_cb cb, void *data, ast_transport_monitor_data_matcher matches);
|
|
|
|
/*!
|
|
* \brief Unregister a reliable transport shutdown monitor
|
|
*
|
|
* \param transport_key Key for the transport to monitor for shutdown.
|
|
* Create the key with AST_SIP_MAKE_REMOTE_IPADDR_PORT_STR.
|
|
* \param cb The callback that was used for the original register.
|
|
* \param data Data to pass to the matcher. May be NULL and does NOT need to be an ao2 object.
|
|
* If NULL, all monitors with the provided callback are unregistered.
|
|
* \param matches Matcher function that returns true if data matches the previously
|
|
* registered data object. If NULL, a simple pointer comparison is done.
|
|
*
|
|
* \note The data object passed into the original register will have its reference count
|
|
* automatically decremented.
|
|
*/
|
|
void ast_sip_transport_monitor_unregister_key(const char *transport_key,
|
|
ast_transport_monitor_shutdown_cb cb, void *data, ast_transport_monitor_data_matcher matches);
|
|
|
|
/*!
|
|
* \brief Unregister a transport shutdown monitor from all reliable transports
|
|
* \since 13.20.0
|
|
*
|
|
* \param cb The callback that was used for the original register.
|
|
* \param data Data to pass to the matcher. May be NULL and does NOT need to be an ao2 object.
|
|
* If NULL, all monitors with the provided callback are unregistered.
|
|
* \param matches Matcher function that returns true if ao2_data matches the previously
|
|
* registered data object. If NULL, a simple pointer comparison is done.
|
|
*
|
|
* \note The data object passed into the original register will have its reference count
|
|
* automatically decremented.
|
|
*/
|
|
void ast_sip_transport_monitor_unregister_all(ast_transport_monitor_shutdown_cb cb,
|
|
void *data, ast_transport_monitor_data_matcher matches);
|
|
|
|
/*! Transport state notification registration element. */
|
|
struct ast_sip_tpmgr_state_callback {
|
|
/*! PJPROJECT transport state notification callback */
|
|
pjsip_tp_state_callback cb;
|
|
AST_LIST_ENTRY(ast_sip_tpmgr_state_callback) node;
|
|
};
|
|
|
|
/*!
|
|
* \brief Register a transport state notification callback element.
|
|
* \since 13.18.0
|
|
*
|
|
* \param element What we are registering.
|
|
*/
|
|
void ast_sip_transport_state_register(struct ast_sip_tpmgr_state_callback *element);
|
|
|
|
/*!
|
|
* \brief Unregister a transport state notification callback element.
|
|
* \since 13.18.0
|
|
*
|
|
* \param element What we are unregistering.
|
|
*/
|
|
void ast_sip_transport_state_unregister(struct ast_sip_tpmgr_state_callback *element);
|
|
|
|
/*!
|
|
* \brief Check whether a pjsip_uri is SIP/SIPS or not
|
|
* \since 16.28.0
|
|
*
|
|
* \param uri The pjsip_uri to check
|
|
*
|
|
* \retval 1 if true
|
|
* \retval 0 if false
|
|
*/
|
|
int ast_sip_is_uri_sip_sips(pjsip_uri *uri);
|
|
|
|
/*!
|
|
* \brief Check whether a pjsip_uri is allowed or not
|
|
* \since 16.28.0
|
|
*
|
|
* \param uri The pjsip_uri to check
|
|
*
|
|
* \retval 1 if allowed
|
|
* \retval 0 if not allowed
|
|
*/
|
|
int ast_sip_is_allowed_uri(pjsip_uri *uri);
|
|
|
|
/*!
|
|
* \brief Get the user portion of the pjsip_uri
|
|
* \since 16.28.0
|
|
*
|
|
* \param uri The pjsip_uri to get the user from
|
|
*
|
|
* \note This function will check what kind of URI it receives and return
|
|
* the user based off of that
|
|
*
|
|
* \return User string or empty string if not present
|
|
*/
|
|
const pj_str_t *ast_sip_pjsip_uri_get_username(pjsip_uri *uri);
|
|
|
|
/*!
|
|
* \brief Get the host portion of the pjsip_uri
|
|
* \since 16.28.0
|
|
*
|
|
* \param uri The pjsip_uri to get the host from
|
|
*
|
|
* \note This function will check what kind of URI it receives and return
|
|
* the host based off of that
|
|
*
|
|
* \return Host string or empty string if not present
|
|
*/
|
|
const pj_str_t *ast_sip_pjsip_uri_get_hostname(pjsip_uri *uri);
|
|
|
|
/*!
|
|
* \brief Find an 'other' SIP/SIPS URI parameter by name
|
|
* \since 16.28.0
|
|
*
|
|
* A convenience function to find a named parameter from a SIP/SIPS URI. This
|
|
* function will not find the following standard SIP/SIPS URI parameters which
|
|
* are stored separately by PJSIP:
|
|
*
|
|
* \li `user`
|
|
* \li `method`
|
|
* \li `transport`
|
|
* \li `ttl`
|
|
* \li `lr`
|
|
* \li `maddr`
|
|
*
|
|
* \param uri The pjsip_uri to get the parameter from
|
|
* \param param_str The name of the parameter to find
|
|
*
|
|
* \note This function will check what kind of URI it receives and return
|
|
* the parameter based off of that
|
|
*
|
|
* \return Find parameter or NULL if not present
|
|
*/
|
|
struct pjsip_param *ast_sip_pjsip_uri_get_other_param(pjsip_uri *uri, const pj_str_t *param_str);
|
|
|
|
/*!
|
|
* \brief Retrieve the system setting 'all_codecs_on_empty_reinvite'.
|
|
*
|
|
* \retval non zero if we should return all codecs on empty re-INVITE
|
|
*/
|
|
unsigned int ast_sip_get_all_codecs_on_empty_reinvite(void);
|
|
|
|
|
|
/*!
|
|
* \brief Convert SIP hangup causes to Asterisk hangup causes
|
|
*
|
|
* \param cause SIP cause
|
|
*
|
|
* \retval matched cause code from causes.h
|
|
*/
|
|
const int ast_sip_hangup_sip2cause(int cause);
|
|
|
|
/*!
|
|
* \brief Convert name to SIP response code
|
|
*
|
|
* \param name SIP response code name matching one of the
|
|
* enum names defined in "enum pjsip_status_code"
|
|
* defined in sip_msg.h. May be specified with or
|
|
* without the PJSIP_SC_ prefix.
|
|
*
|
|
* \retval SIP response code
|
|
* \retval -1 if matching code not found
|
|
*/
|
|
int ast_sip_str2rc(const char *name);
|
|
|
|
#endif /* _RES_PJSIP_H */
|