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2699 lines
90 KiB
2699 lines
90 KiB
/*
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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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#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 pj_sockaddr */
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#include <pjlib.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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/* 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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/*!
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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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* \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 address
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* \since 13.8.0
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*/
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struct ast_dnsmgr_entry *external_address_refresher;
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/*!
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* Optional external address information
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* \since 13.8.0
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*/
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struct ast_sockaddr external_address;
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};
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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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};
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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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/*!
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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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);
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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 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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/*! Asterisk Server name */
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AST_STRING_FIELD_EXTENDED(reg_server);
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/*! IP-address of the Via header in REGISTER request */
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AST_STRING_FIELD_EXTENDED(via_addr);
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/* Port of the Via header in REGISTER request */
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int via_port;
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/*! Content of the Call-ID header in REGISTER request */
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AST_STRING_FIELD_EXTENDED(call_id);
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/*! The name of the endpoint that added the contact */
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AST_STRING_FIELD_EXTENDED(endpoint_name);
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};
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#define CONTACT_STATUS "contact_status"
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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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UNAVAILABLE,
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AVAILABLE,
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UNKNOWN,
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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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* \detail Maintains a contact's current status and round trip time
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* if available.
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*/
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struct ast_sip_contact_status {
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SORCERY_OBJECT(details);
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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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/*! Current status for a contact (default - unavailable) */
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enum ast_sip_contact_status_type status;
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/*! The round trip start time set before sending a qualify request */
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struct timeval rtt_start;
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/*! The round trip time in microseconds */
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int64_t rtt;
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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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/*! TRUE if the contact was refreshed. e.g., re-registered */
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unsigned int refresh:1;
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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 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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};
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/*!
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* \brief A wrapper for contact that adds the aor_id and
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* a consistent contact id. Used by ast_sip_for_each_contact.
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*/
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struct ast_sip_contact_wrapper {
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/*! The id of the parent aor. */
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char *aor_id;
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/*! The id of contact in form of aor_id/contact_uri. */
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char *contact_id;
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/*! Pointer to the actual contact. */
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struct ast_sip_contact *contact;
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};
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/*!
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* \brief DTMF modes for SIP endpoints
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*/
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enum ast_sip_dtmf_mode {
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/*! No DTMF to be used */
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AST_SIP_DTMF_NONE,
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/* XXX Should this be 2833 instead? */
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/*! Use RFC 4733 events for DTMF */
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AST_SIP_DTMF_RFC_4733,
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/*! Use DTMF in the audio stream */
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AST_SIP_DTMF_INBAND,
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/*! Use SIP INFO DTMF (blech) */
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AST_SIP_DTMF_INFO,
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/*! Use SIP 4733 if supported by the other side or INBAND if not */
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AST_SIP_DTMF_AUTO,
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};
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/*!
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* \brief Methods of storing SIP digest authentication credentials.
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*
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* Note that both methods result in MD5 digest authentication being
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* used. The two methods simply alter how Asterisk determines the
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* credentials for a SIP authentication
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*/
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enum ast_sip_auth_type {
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/*! Credentials stored as a username and password combination */
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AST_SIP_AUTH_TYPE_USER_PASS,
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/*! Credentials stored as an MD5 sum */
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AST_SIP_AUTH_TYPE_MD5,
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/*! Credentials not stored this is a fake auth */
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AST_SIP_AUTH_TYPE_ARTIFICIAL
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};
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#define SIP_SORCERY_AUTH_TYPE "auth"
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struct ast_sip_auth {
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/*! Sorcery ID of the auth is its name */
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SORCERY_OBJECT(details);
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AST_DECLARE_STRING_FIELDS(
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/*! Identification for these credentials */
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AST_STRING_FIELD(realm);
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/*! Authentication username */
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AST_STRING_FIELD(auth_user);
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/*! Authentication password */
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AST_STRING_FIELD(auth_pass);
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/*! Authentication credentials in MD5 format (hash of user:realm:pass) */
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AST_STRING_FIELD(md5_creds);
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);
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/*! The time period (in seconds) that a nonce may be reused */
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unsigned int nonce_lifetime;
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/*! Used to determine what to use when authenticating */
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enum ast_sip_auth_type type;
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};
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AST_VECTOR(ast_sip_auth_vector, const char *);
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/*!
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* \brief Different methods by which incoming requests can be matched to endpoints
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*/
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enum ast_sip_endpoint_identifier_type {
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/*! Identify based on user name in From header */
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AST_SIP_ENDPOINT_IDENTIFY_BY_USERNAME = (1 << 0),
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/*! Identify based on user name in Auth header first, then From header */
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AST_SIP_ENDPOINT_IDENTIFY_BY_AUTH_USERNAME = (1 << 1),
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};
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AST_VECTOR(ast_sip_identify_by_vector, enum ast_sip_endpoint_identifier_type);
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enum ast_sip_session_refresh_method {
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/*! Use reinvite to negotiate direct media */
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AST_SIP_SESSION_REFRESH_METHOD_INVITE,
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/*! Use UPDATE to negotiate direct media */
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AST_SIP_SESSION_REFRESH_METHOD_UPDATE,
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};
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enum ast_sip_direct_media_glare_mitigation {
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/*! Take no special action to mitigate reinvite glare */
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AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_NONE,
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/*! Do not send an initial direct media session refresh on outgoing call legs
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* Subsequent session refreshes will be sent no matter the session direction
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*/
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AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_OUTGOING,
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/*! Do not send an initial direct media session refresh on incoming call legs
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* Subsequent session refreshes will be sent no matter the session direction
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*/
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AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_INCOMING,
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};
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enum ast_sip_session_media_encryption {
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/*! Invalid media encryption configuration */
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AST_SIP_MEDIA_TRANSPORT_INVALID = 0,
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/*! Do not allow any encryption of session media */
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AST_SIP_MEDIA_ENCRYPT_NONE,
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/*! Offer SDES-encrypted session media */
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AST_SIP_MEDIA_ENCRYPT_SDES,
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/*! Offer encrypted session media with datagram TLS key exchange */
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AST_SIP_MEDIA_ENCRYPT_DTLS,
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};
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enum ast_sip_session_redirect {
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/*! User portion of the target URI should be used as the target in the dialplan */
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AST_SIP_REDIRECT_USER = 0,
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/*! Target URI should be used as the target in the dialplan */
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AST_SIP_REDIRECT_URI_CORE,
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/*! Target URI should be used as the target within chan_pjsip itself */
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AST_SIP_REDIRECT_URI_PJSIP,
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};
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/*!
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* \brief Session timers options
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*/
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struct ast_sip_timer_options {
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/*! Minimum session expiration period, in seconds */
|
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unsigned int min_se;
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/*! Session expiration period, in seconds */
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|
unsigned int sess_expires;
|
|
};
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|
|
|
/*!
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* \brief Endpoint configuration for SIP extensions.
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*
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* SIP extensions, in this case refers to features
|
|
* indicated in Supported or Required headers.
|
|
*/
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|
struct ast_sip_endpoint_extensions {
|
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/*! Enabled SIP extensions */
|
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unsigned int flags;
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/*! Timer options */
|
|
struct ast_sip_timer_options timer;
|
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};
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|
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/*!
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* \brief Endpoint configuration for unsolicited MWI
|
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*/
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struct ast_sip_mwi_configuration {
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AST_DECLARE_STRING_FIELDS(
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/*! Configured voicemail boxes for this endpoint. Used for MWI */
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AST_STRING_FIELD(mailboxes);
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/*! Username to use when sending MWI NOTIFYs to this endpoint */
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AST_STRING_FIELD(fromuser);
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);
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|
/* Should mailbox states be combined into a single notification? */
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|
unsigned int aggregate;
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|
/* Should a subscribe replace unsolicited notifies? */
|
|
unsigned int subscribe_replaces_unsolicited;
|
|
/* Voicemail extension to set in Message-Account */
|
|
char *voicemail_extension;
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};
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|
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/*!
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* \brief Endpoint subscription configuration
|
|
*/
|
|
struct ast_sip_endpoint_subscription_configuration {
|
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/*! 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];
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|
};
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|
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/*!
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* \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;
|
|
/*! When performing connected line update, which method should be used */
|
|
enum ast_sip_session_refresh_method refresh_method;
|
|
};
|
|
|
|
/*!
|
|
* \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;
|
|
};
|
|
|
|
/*!
|
|
* \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;
|
|
};
|
|
|
|
/*!
|
|
* \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;
|
|
/*! 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;
|
|
};
|
|
|
|
/*!
|
|
* \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);
|
|
);
|
|
/*! 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;
|
|
/*! Do we allow an asymmetric RTP codec? */
|
|
unsigned int asymmetric_rtp_codec;
|
|
};
|
|
|
|
/*!
|
|
* \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 array 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 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 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 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 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 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 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 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 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 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 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 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 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 PSJIP 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.
|
|
*
|
|
* \warning \b Never use this function in a SIP servant thread. This can potentially
|
|
* cause a deadlock. If you are in a SIP servant thread, just call your function
|
|
* in-line.
|
|
*
|
|
* \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.
|
|
*
|
|
* \param serializer The SIP serializer to which the task belongs. May 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_synchronous(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),
|
|
|
|
/*!
|
|
* 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 AST_SIP_SCHED_TASK_PERIODIC
|
|
* The task is scheduled at multiples of interval
|
|
* \see Interval
|
|
*/
|
|
AST_SIP_SCHED_TASK_PERIODIC = (0 << 4),
|
|
/*! \brief AST_SIP_SCHED_TASK_DELAY
|
|
* The next invocation of the task is at last finish + interval
|
|
* \see Interval
|
|
*/
|
|
AST_SIP_SCHED_TASK_DELAY = (1 << 4),
|
|
};
|
|
|
|
/*!
|
|
* \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.
|
|
*
|
|
* \paragraph 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.
|
|
*
|
|
* \paragraph 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 \ref "interval" milliseconds, regardless of how long the task
|
|
* takes. If the task takes longer than \ref interval, it will be scheduled at the next available
|
|
* multiple of \ref interval. For exmaple: 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 \ref 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, 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 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 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
|
|
* \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
|
|
*
|
|
* \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 an rdata structure using specific information
|
|
*
|
|
* \param rdata[out] 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] 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 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 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 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 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 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 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 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 vector of auth structures to clean up
|
|
* \param num_auths The number of auths in the vector
|
|
*/
|
|
void ast_sip_cleanup_auths(struct ast_sip_auth *auths[], size_t num_auths);
|
|
|
|
/*!
|
|
* \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
|
|
* \retval 0 Success
|
|
* \retval -1 Failure
|
|
*/
|
|
int 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 str 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 endpoint 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 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 retreieved.
|
|
* \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
|
|
* \retval 0 Success
|
|
* \retval -1 Failure
|
|
*/
|
|
int 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 '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 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
|
|
* \return nothing
|
|
*/
|
|
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
|
|
* \return nothing
|
|
*/
|
|
void ast_sip_get_default_from_user(char *from_user, size_t size);
|
|
|
|
/*! \brief Determines whether the res_pjsip module is loaded */
|
|
#define CHECK_PJSIP_MODULE_LOADED() \
|
|
do { \
|
|
if (!ast_module_check("res_pjsip.so") \
|
|
|| !ast_sip_get_pjsip_endpoint()) { \
|
|
return AST_MODULE_LOAD_DECLINE; \
|
|
} \
|
|
} while(0)
|
|
|
|
/*!
|
|
* \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 transport state
|
|
* \since 13.7.1
|
|
*
|
|
* @param transport_id
|
|
* @returns 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 Retrieves all transport states
|
|
* \since 13.7.1
|
|
*
|
|
* @returns 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
|
|
*/
|
|
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
|
|
*/
|
|
int ast_sip_set_tpselector_from_transport_name(const char *transport_name, pjsip_tpselector *selector);
|
|
|
|
/*!
|
|
* \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 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);
|
|
|
|
#endif /* _RES_PJSIP_H */
|