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asterisk/include/asterisk/res_pjsip.h
Alexei Gradinari 6fa3ed0679 res_pjsip: improve realtime performance #2 
The patch removes updating all Endpoints' status on startup.
Instead, only non-qualified aors with static contact
and non-qualified non-expired contacts are retrieved from the realtime to
update the endpoint status to ONLINE.
The endpoint name was added to the contact object to simply find the endpoint
that created this contact.

The status of endpoints with qualified aors will be updated by 'qualify'
functions.

ASTERISK-26061 #close

Change-Id: Id324c1776fa55d3741e0c5457ecac0304cb1a0df
2016-06-22 15:29:50 -04:00

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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2013, Digium, Inc.
*
* Mark Michelson <mmichelson@digium.com>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
#ifndef _RES_PJSIP_H
#define _RES_PJSIP_H
#include <pjsip.h>
/* Needed for SUBSCRIBE, NOTIFY, and PUBLISH method definitions */
#include <pjsip_simple.h>
#include <pjsip/sip_transaction.h>
#include <pj/timer.h>
#include <pjlib.h>
#include "asterisk/stringfields.h"
/* Needed for struct ast_sockaddr */
#include "asterisk/netsock2.h"
/* Needed for linked list macros */
#include "asterisk/linkedlists.h"
/* Needed for ast_party_id */
#include "asterisk/channel.h"
/* Needed for ast_sorcery */
#include "asterisk/sorcery.h"
/* Needed for ast_dnsmgr */
#include "asterisk/dnsmgr.h"
/* Needed for ast_endpoint */
#include "asterisk/endpoints.h"
/* Needed for ast_t38_ec_modes */
#include "asterisk/udptl.h"
/* Needed for pj_sockaddr */
#include <pjlib.h>
/* Needed for ast_rtp_dtls_cfg struct */
#include "asterisk/rtp_engine.h"
/* Needed for AST_VECTOR macro */
#include "asterisk/vector.h"
/* Needed for ast_sip_for_each_channel_snapshot struct */
#include "asterisk/stasis_channels.h"
#include "asterisk/stasis_endpoints.h"
/* Forward declarations of PJSIP stuff */
struct pjsip_rx_data;
struct pjsip_module;
struct pjsip_tx_data;
struct pjsip_dialog;
struct pjsip_transport;
struct pjsip_tpfactory;
struct pjsip_tls_setting;
struct pjsip_tpselector;
/*! \brief Maximum number of ciphers supported for a TLS transport */
#define SIP_TLS_MAX_CIPHERS 64
/*!
* \brief Structure for SIP transport information
*/
struct ast_sip_transport_state {
/*! \brief Transport itself */
struct pjsip_transport *transport;
/*! \brief Transport factory */
struct pjsip_tpfactory *factory;
/*!
* Transport id
* \since 13.8.0
*/
char *id;
/*!
* Transport type
* \since 13.8.0
*/
enum ast_transport type;
/*!
* Address and port to bind to
* \since 13.8.0
*/
pj_sockaddr host;
/*!
* TLS settings
* \since 13.8.0
*/
pjsip_tls_setting tls;
/*!
* Configured TLS ciphers
* \since 13.8.0
*/
pj_ssl_cipher ciphers[SIP_TLS_MAX_CIPHERS];
/*!
* Optional local network information, used for NAT purposes
* \since 13.8.0
*/
struct ast_ha *localnet;
/*!
* DNS manager for refreshing the external address
* \since 13.8.0
*/
struct ast_dnsmgr_entry *external_address_refresher;
/*!
* Optional external address information
* \since 13.8.0
*/
struct ast_sockaddr external_address;
};
/*
* \brief Transport to bind to
*/
struct ast_sip_transport {
/*! Sorcery object details */
SORCERY_OBJECT(details);
AST_DECLARE_STRING_FIELDS(
/*! Certificate of authority list file */
AST_STRING_FIELD(ca_list_file);
/*! Certificate of authority list path */
AST_STRING_FIELD(ca_list_path);
/*! Public certificate file */
AST_STRING_FIELD(cert_file);
/*! Optional private key of the certificate file */
AST_STRING_FIELD(privkey_file);
/*! Password to open the private key */
AST_STRING_FIELD(password);
/*! External signaling address */
AST_STRING_FIELD(external_signaling_address);
/*! External media address */
AST_STRING_FIELD(external_media_address);
/*! Optional domain to use for messages if provided could not be found */
AST_STRING_FIELD(domain);
);
/*! Type of transport */
enum ast_transport type;
/*!
* \deprecated Moved to ast_sip_transport_state
* \version 13.8.0 deprecated
* Address and port to bind to
*/
pj_sockaddr host;
/*! Number of simultaneous asynchronous operations */
unsigned int async_operations;
/*! Optional external port for signaling */
unsigned int external_signaling_port;
/*!
* \deprecated Moved to ast_sip_transport_state
* \version 13.7.1 deprecated
* TLS settings
*/
pjsip_tls_setting tls;
/*!
* \deprecated Moved to ast_sip_transport_state
* \version 13.7.1 deprecated
* Configured TLS ciphers
*/
pj_ssl_cipher ciphers[SIP_TLS_MAX_CIPHERS];
/*!
* \deprecated Moved to ast_sip_transport_state
* \version 13.7.1 deprecated
* Optional local network information, used for NAT purposes
*/
struct ast_ha *localnet;
/*!
* \deprecated Moved to ast_sip_transport_state
* \version 13.7.1 deprecated
* DNS manager for refreshing the external address
*/
struct ast_dnsmgr_entry *external_address_refresher;
/*!
* \deprecated Moved to ast_sip_transport_state
* \version 13.7.1 deprecated
* Optional external address information
*/
struct ast_sockaddr external_address;
/*!
* \deprecated
* \version 13.7.1 deprecated
* Transport state information
*/
struct ast_sip_transport_state *state;
/*! QOS DSCP TOS bits */
unsigned int tos;
/*! QOS COS value */
unsigned int cos;
/*! Write timeout */
int write_timeout;
/*! Allow reload */
int allow_reload;
};
#define SIP_SORCERY_DOMAIN_ALIAS_TYPE "domain_alias"
/*!
* Details about a SIP domain alias
*/
struct ast_sip_domain_alias {
/*! Sorcery object details */
SORCERY_OBJECT(details);
AST_DECLARE_STRING_FIELDS(
/*! Domain to be aliased to */
AST_STRING_FIELD(domain);
);
};
/*!
* \brief Structure for SIP nat hook information
*/
struct ast_sip_nat_hook {
/*! Sorcery object details */
SORCERY_OBJECT(details);
/*! Callback for when a message is going outside of our local network */
void (*outgoing_external_message)(struct pjsip_tx_data *tdata, struct ast_sip_transport *transport);
};
/*!
* \brief Contact associated with an address of record
*/
struct ast_sip_contact {
/*! Sorcery object details, the id is the aor name plus a random string */
SORCERY_OBJECT(details);
AST_DECLARE_STRING_FIELDS(
/*! Full URI of the contact */
AST_STRING_FIELD(uri);
/*! Outbound proxy to use for qualify */
AST_STRING_FIELD(outbound_proxy);
/*! Path information to place in Route headers */
AST_STRING_FIELD(path);
/*! Content of the User-Agent header in REGISTER request */
AST_STRING_FIELD(user_agent);
/*! The name of the aor this contact belongs to */
AST_STRING_FIELD(aor);
);
/*! Absolute time that this contact is no longer valid after */
struct timeval expiration_time;
/*! Frequency to send OPTIONS requests to contact. 0 is disabled. */
unsigned int qualify_frequency;
/*! If true authenticate the qualify if needed */
int authenticate_qualify;
/*! Qualify timeout. 0 is diabled. */
double qualify_timeout;
/*! Endpoint that added the contact, only available in observers */
struct ast_sip_endpoint *endpoint;
/*! Asterisk Server name */
AST_STRING_FIELD_EXTENDED(reg_server);
/*! IP-address of the Via header in REGISTER request */
AST_STRING_FIELD_EXTENDED(via_addr);
/* Port of the Via header in REGISTER request */
int via_port;
/*! Content of the Call-ID header in REGISTER request */
AST_STRING_FIELD_EXTENDED(call_id);
/*! The name of the endpoint that added the contact */
AST_STRING_FIELD_EXTENDED(endpoint_name);
};
#define CONTACT_STATUS "contact_status"
/*!
* \brief Status type for a contact.
*/
enum ast_sip_contact_status_type {
UNAVAILABLE,
AVAILABLE,
UNKNOWN,
CREATED,
REMOVED,
UPDATED,
};
/*!
* \brief A contact's status.
*
* \detail Maintains a contact's current status and round trip time
* if available.
*/
struct ast_sip_contact_status {
SORCERY_OBJECT(details);
AST_DECLARE_STRING_FIELDS(
/*! The original contact's URI */
AST_STRING_FIELD(uri);
/*! The name of the aor this contact_status belongs to */
AST_STRING_FIELD(aor);
);
/*! Current status for a contact (default - unavailable) */
enum ast_sip_contact_status_type status;
/*! The round trip start time set before sending a qualify request */
struct timeval rtt_start;
/*! The round trip time in microseconds */
int64_t rtt;
/*! Last status for a contact (default - unavailable) */
enum ast_sip_contact_status_type last_status;
};
/*!
* \brief A SIP address of record
*/
struct ast_sip_aor {
/*! Sorcery object details, the id is the AOR name */
SORCERY_OBJECT(details);
AST_DECLARE_STRING_FIELDS(
/*! Voicemail boxes for this AOR */
AST_STRING_FIELD(mailboxes);
/*! Outbound proxy for OPTIONS requests */
AST_STRING_FIELD(outbound_proxy);
);
/*! Minimum expiration time */
unsigned int minimum_expiration;
/*! Maximum expiration time */
unsigned int maximum_expiration;
/*! Default contact expiration if one is not provided in the contact */
unsigned int default_expiration;
/*! Frequency to send OPTIONS requests to AOR contacts. 0 is disabled. */
unsigned int qualify_frequency;
/*! If true authenticate the qualify if needed */
int authenticate_qualify;
/*! Maximum number of external contacts, 0 to disable */
unsigned int max_contacts;
/*! Whether to remove any existing contacts not related to an incoming REGISTER when it comes in */
unsigned int remove_existing;
/*! Any permanent configured contacts */
struct ao2_container *permanent_contacts;
/*! Determines whether SIP Path headers are supported */
unsigned int support_path;
/*! Qualify timeout. 0 is diabled. */
double qualify_timeout;
/* Voicemail extension to set in Message-Account */
char *voicemail_extension;
};
/*!
* \brief A wrapper for contact that adds the aor_id and
* a consistent contact id. Used by ast_sip_for_each_contact.
*/
struct ast_sip_contact_wrapper {
/*! The id of the parent aor. */
char *aor_id;
/*! The id of contact in form of aor_id/contact_uri. */
char *contact_id;
/*! Pointer to the actual contact. */
struct ast_sip_contact *contact;
};
/*!
* \brief DTMF modes for SIP endpoints
*/
enum ast_sip_dtmf_mode {
/*! No DTMF to be used */
AST_SIP_DTMF_NONE,
/* XXX Should this be 2833 instead? */
/*! Use RFC 4733 events for DTMF */
AST_SIP_DTMF_RFC_4733,
/*! Use DTMF in the audio stream */
AST_SIP_DTMF_INBAND,
/*! Use SIP INFO DTMF (blech) */
AST_SIP_DTMF_INFO,
/*! Use SIP 4733 if supported by the other side or INBAND if not */
AST_SIP_DTMF_AUTO,
};
/*!
* \brief Methods of storing SIP digest authentication credentials.
*
* Note that both methods result in MD5 digest authentication being
* used. The two methods simply alter how Asterisk determines the
* credentials for a SIP authentication
*/
enum ast_sip_auth_type {
/*! Credentials stored as a username and password combination */
AST_SIP_AUTH_TYPE_USER_PASS,
/*! Credentials stored as an MD5 sum */
AST_SIP_AUTH_TYPE_MD5,
/*! Credentials not stored this is a fake auth */
AST_SIP_AUTH_TYPE_ARTIFICIAL
};
#define SIP_SORCERY_AUTH_TYPE "auth"
struct ast_sip_auth {
/*! Sorcery ID of the auth is its name */
SORCERY_OBJECT(details);
AST_DECLARE_STRING_FIELDS(
/*! Identification for these credentials */
AST_STRING_FIELD(realm);
/*! Authentication username */
AST_STRING_FIELD(auth_user);
/*! Authentication password */
AST_STRING_FIELD(auth_pass);
/*! Authentication credentials in MD5 format (hash of user:realm:pass) */
AST_STRING_FIELD(md5_creds);
);
/*! The time period (in seconds) that a nonce may be reused */
unsigned int nonce_lifetime;
/*! Used to determine what to use when authenticating */
enum ast_sip_auth_type type;
};
AST_VECTOR(ast_sip_auth_vector, const char *);
/*!
* \brief Different methods by which incoming requests can be matched to endpoints
*/
enum ast_sip_endpoint_identifier_type {
/*! Identify based on user name in From header */
AST_SIP_ENDPOINT_IDENTIFY_BY_USERNAME = (1 << 0),
/*! Identify based on user name in Auth header first, then From header */
AST_SIP_ENDPOINT_IDENTIFY_BY_AUTH_USERNAME = (1 << 1),
};
AST_VECTOR(ast_sip_identify_by_vector, enum ast_sip_endpoint_identifier_type);
enum ast_sip_session_refresh_method {
/*! Use reinvite to negotiate direct media */
AST_SIP_SESSION_REFRESH_METHOD_INVITE,
/*! Use UPDATE to negotiate direct media */
AST_SIP_SESSION_REFRESH_METHOD_UPDATE,
};
enum ast_sip_direct_media_glare_mitigation {
/*! Take no special action to mitigate reinvite glare */
AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_NONE,
/*! Do not send an initial direct media session refresh on outgoing call legs
* Subsequent session refreshes will be sent no matter the session direction
*/
AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_OUTGOING,
/*! Do not send an initial direct media session refresh on incoming call legs
* Subsequent session refreshes will be sent no matter the session direction
*/
AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_INCOMING,
};
enum ast_sip_session_media_encryption {
/*! Invalid media encryption configuration */
AST_SIP_MEDIA_TRANSPORT_INVALID = 0,
/*! Do not allow any encryption of session media */
AST_SIP_MEDIA_ENCRYPT_NONE,
/*! Offer SDES-encrypted session media */
AST_SIP_MEDIA_ENCRYPT_SDES,
/*! Offer encrypted session media with datagram TLS key exchange */
AST_SIP_MEDIA_ENCRYPT_DTLS,
};
enum ast_sip_session_redirect {
/*! User portion of the target URI should be used as the target in the dialplan */
AST_SIP_REDIRECT_USER = 0,
/*! Target URI should be used as the target in the dialplan */
AST_SIP_REDIRECT_URI_CORE,
/*! Target URI should be used as the target within chan_pjsip itself */
AST_SIP_REDIRECT_URI_PJSIP,
};
/*!
* \brief Session timers options
*/
struct ast_sip_timer_options {
/*! Minimum session expiration period, in seconds */
unsigned int min_se;
/*! Session expiration period, in seconds */
unsigned int sess_expires;
};
/*!
* \brief Endpoint configuration for SIP extensions.
*
* SIP extensions, in this case refers to features
* indicated in Supported or Required headers.
*/
struct ast_sip_endpoint_extensions {
/*! Enabled SIP extensions */
unsigned int flags;
/*! Timer options */
struct ast_sip_timer_options timer;
};
/*!
* \brief Endpoint configuration for unsolicited MWI
*/
struct ast_sip_mwi_configuration {
AST_DECLARE_STRING_FIELDS(
/*! Configured voicemail boxes for this endpoint. Used for MWI */
AST_STRING_FIELD(mailboxes);
/*! Username to use when sending MWI NOTIFYs to this endpoint */
AST_STRING_FIELD(fromuser);
);
/* Should mailbox states be combined into a single notification? */
unsigned int aggregate;
/* Should a subscribe replace unsolicited notifies? */
unsigned int subscribe_replaces_unsolicited;
/* Voicemail extension to set in Message-Account */
char *voicemail_extension;
};
/*!
* \brief Endpoint subscription configuration
*/
struct ast_sip_endpoint_subscription_configuration {
/*! Indicates if endpoint is allowed to initiate subscriptions */
unsigned int allow;
/*! The minimum allowed expiration for subscriptions from endpoint */
unsigned int minexpiry;
/*! Message waiting configuration */
struct ast_sip_mwi_configuration mwi;
};
/*!
* \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;
};
/*!
* \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 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 */
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