/* * Asterisk -- An open source telephony toolkit. * * Copyright (C) 2013, Digium, Inc. * * Mark Michelson * * 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 /* Needed for SUBSCRIBE, NOTIFY, and PUBLISH method definitions */ #include #include #include #include #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 /* 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. * "deny" (set) means that it's in the local network. Use the * ast_sip_transport_is_nonlocal and ast_sip_transport_is_local * macro's. * \since 13.8.0 */ struct ast_ha *localnet; /*! * DNS manager for refreshing the external signaling address * \since 13.8.0 */ struct ast_dnsmgr_entry *external_signaling_address_refresher; /*! * Optional external signaling address information * \since 13.8.0 */ struct ast_sockaddr external_signaling_address; /*! * DNS manager for refreshing the external media address * \since 13.18.0 */ struct ast_dnsmgr_entry *external_media_address_refresher; /*! * Optional external signaling address information * \since 13.18.0 */ struct ast_sockaddr external_media_address; }; #define ast_sip_transport_is_nonlocal(transport_state, addr) \ (!transport_state->localnet || ast_apply_ha(transport_state->localnet, addr) == AST_SENSE_ALLOW) #define ast_sip_transport_is_local(transport_state, addr) \ (transport_state->localnet && ast_apply_ha(transport_state->localnet, addr) != AST_SENSE_ALLOW) /* * \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; /*! Automatically send requests out the same transport requests have come in on */ int symmetric_transport; }; #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 challenge response 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); /*! If true delete the contact on Asterisk restart/boot */ int prune_on_boot; }; #define CONTACT_STATUS "contact_status" /*! * \brief Status type for a contact. */ enum ast_sip_contact_status_type { /*! Frequency > 0, but no response from remote uri */ UNAVAILABLE, /*! Frequency > 0, and got response from remote uri */ AVAILABLE, /*! Default last status, and when a contact status object is not found */ UNKNOWN, /*! Frequency == 0, has a contact, but don't know status (non-qualified) */ CREATED, REMOVED, }; /*! * \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; /*! TRUE if the contact was refreshed. e.g., re-registered */ unsigned int refresh:1; }; /*! * \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 challenge response 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, /*! Use SIP 4733 if supported by the other side or INFO DTMF (blech) if not */ AST_SIP_DTMF_AUTO_INFO, }; /*! * \brief Methods of storing SIP digest authentication credentials. * * Note that both methods result in MD5 digest authentication being * used. The two methods simply alter how Asterisk determines the * credentials for a SIP authentication */ enum ast_sip_auth_type { /*! Credentials stored as a username and password combination */ AST_SIP_AUTH_TYPE_USER_PASS, /*! Credentials stored as an MD5 sum */ AST_SIP_AUTH_TYPE_MD5, /*! 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), /*! Identify based on source IP address */ AST_SIP_ENDPOINT_IDENTIFY_BY_IP = (1 << 2), /*! Identify based on arbitrary headers */ AST_SIP_ENDPOINT_IDENTIFY_BY_HEADER = (1 << 3), }; AST_VECTOR(ast_sip_identify_by_vector, enum ast_sip_endpoint_identifier_type); enum ast_sip_session_refresh_method { /*! Use reinvite to negotiate direct media */ AST_SIP_SESSION_REFRESH_METHOD_INVITE, /*! Use UPDATE to negotiate direct media */ AST_SIP_SESSION_REFRESH_METHOD_UPDATE, }; enum ast_sip_direct_media_glare_mitigation { /*! Take no special action to mitigate reinvite glare */ AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_NONE, /*! Do not send an initial direct media session refresh on outgoing call legs * Subsequent session refreshes will be sent no matter the session direction */ AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_OUTGOING, /*! Do not send an initial direct media session refresh on incoming call legs * Subsequent session refreshes will be sent no matter the session direction */ AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_INCOMING, }; enum ast_sip_session_media_encryption { /*! Invalid media encryption configuration */ AST_SIP_MEDIA_TRANSPORT_INVALID = 0, /*! Do not allow any encryption of session media */ AST_SIP_MEDIA_ENCRYPT_NONE, /*! Offer SDES-encrypted session media */ AST_SIP_MEDIA_ENCRYPT_SDES, /*! Offer encrypted session media with datagram TLS key exchange */ AST_SIP_MEDIA_ENCRYPT_DTLS, }; enum ast_sip_session_redirect { /*! User portion of the target URI should be used as the target in the dialplan */ AST_SIP_REDIRECT_USER = 0, /*! Target URI should be used as the target in the dialplan */ AST_SIP_REDIRECT_URI_CORE, /*! Target URI should be used as the target within chan_pjsip itself */ AST_SIP_REDIRECT_URI_PJSIP, }; /*! * \brief Session timers options */ struct ast_sip_timer_options { /*! Minimum session expiration period, in seconds */ unsigned int min_se; /*! Session expiration period, in seconds */ unsigned int sess_expires; }; /*! * \brief Endpoint configuration for SIP extensions. * * SIP extensions, in this case refers to features * indicated in Supported or Required headers. */ struct ast_sip_endpoint_extensions { /*! Enabled SIP extensions */ unsigned int flags; /*! Timer options */ struct ast_sip_timer_options timer; }; /*! * \brief Endpoint configuration for unsolicited MWI */ struct ast_sip_mwi_configuration { AST_DECLARE_STRING_FIELDS( /*! Configured voicemail boxes for this endpoint. Used for MWI */ AST_STRING_FIELD(mailboxes); /*! Username to use when sending MWI NOTIFYs to this endpoint */ AST_STRING_FIELD(fromuser); ); /* Should mailbox states be combined into a single notification? */ unsigned int aggregate; /* Should a subscribe replace unsolicited notifies? */ unsigned int subscribe_replaces_unsolicited; /* Voicemail extension to set in Message-Account */ char *voicemail_extension; }; /*! * \brief Endpoint subscription configuration */ struct ast_sip_endpoint_subscription_configuration { /*! Indicates if endpoint is allowed to initiate subscriptions */ unsigned int allow; /*! The minimum allowed expiration for subscriptions from endpoint */ unsigned int minexpiry; /*! Message waiting configuration */ struct ast_sip_mwi_configuration mwi; /* Context for SUBSCRIBE requests */ char context[AST_MAX_CONTEXT]; }; /*! * \brief NAT configuration options for endpoints */ struct ast_sip_endpoint_nat_configuration { /*! Whether to force using the source IP address/port for sending responses */ unsigned int force_rport; /*! Whether to rewrite the Contact header with the source IP address/port or not */ unsigned int rewrite_contact; }; /*! * \brief Party identification options for endpoints * * This includes caller ID, connected line, and redirecting-related options */ struct ast_sip_endpoint_id_configuration { struct ast_party_id self; /*! Do we accept identification information from this endpoint */ unsigned int trust_inbound; /*! Do we send private identification information to this endpoint? */ unsigned int trust_outbound; /*! Do we send P-Asserted-Identity headers to this endpoint? */ unsigned int send_pai; /*! Do we send Remote-Party-ID headers to this endpoint? */ unsigned int send_rpid; /*! Do we send messages for connected line updates for unanswered incoming calls immediately to this endpoint? */ unsigned int rpid_immediate; /*! Do we add Diversion headers to applicable outgoing requests/responses? */ unsigned int send_diversion; /*! 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; /*! Capabilities in topology form */ struct ast_stream_topology *topology; /*! DSCP TOS bits for audio streams */ unsigned int tos_audio; /*! Priority for audio streams */ unsigned int cos_audio; /*! DSCP TOS bits for video streams */ unsigned int tos_video; /*! Priority for video streams */ unsigned int cos_video; /*! Is g.726 packed in a non standard way */ unsigned int g726_non_standard; /*! Bind the RTP instance to the media_address */ unsigned int bind_rtp_to_media_address; /*! Use RTCP-MUX */ unsigned int rtcp_mux; /*! Maximum number of audio streams to offer/accept */ unsigned int max_audio_streams; /*! Maximum number of video streams to offer/accept */ unsigned int max_video_streams; /*! Use BUNDLE */ unsigned int bundle; /*! Enable webrtc settings and defaults */ unsigned int webrtc; }; /*! * \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; /*! Whether to response SDP offer with single most preferred codec. */ unsigned int preferred_codec_only; /*! Do we allow an asymmetric RTP codec? */ unsigned int asymmetric_rtp_codec; /*! Do we allow overlap dialling? */ unsigned int allow_overlap; /*! Whether to notifies all the progress details on blind transfer */ unsigned int refer_blind_progress; /*! Whether to notifies dialog-info 'early' on INUSE && RINGING state */ unsigned int notify_early_inuse_ringing; /*! If set, we'll push incoming MWI NOTIFYs to stasis using this mailbox */ AST_STRING_FIELD_EXTENDED(incoming_mwi_mailbox); }; /*! URI parameter for symmetric transport */ #define AST_SIP_X_AST_TXP "x-ast-txp" #define AST_SIP_X_AST_TXP_LEN 9 /*! * \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 Contact retrieval filtering flags */ enum ast_sip_contact_filter { /*! \brief Default filter flags */ AST_SIP_CONTACT_FILTER_DEFAULT = 0, /*! \brief Return only reachable or unknown contacts */ AST_SIP_CONTACT_FILTER_REACHABLE = (1 << 0), }; /*! * \brief 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 the first bound contact for an AOR and filter based on flags * \since 13.16.0 * * \param aor Pointer to the AOR * \param flags Filtering flags * \retval NULL if no contacts available * \retval non-NULL if contacts available */ struct ast_sip_contact *ast_sip_location_retrieve_first_aor_contact_filtered(const struct ast_sip_aor *aor, unsigned int flags); /*! * \brief Retrieve all contacts currently available for an AOR * * \param aor Pointer to the AOR * * \retval NULL if no contacts available * \retval non-NULL if contacts available * * \warning * Since this function prunes expired contacts before returning, it holds a named write * lock on the aor. If you already hold the lock, call ast_sip_location_retrieve_aor_contacts_nolock instead. */ struct ao2_container *ast_sip_location_retrieve_aor_contacts(const struct ast_sip_aor *aor); /*! * \brief Retrieve all contacts currently available for an AOR and filter based on flags * \since 13.16.0 * * \param aor Pointer to the AOR * \param flags Filtering flags * * \retval NULL if no contacts available * \retval non-NULL if contacts available * * \warning * Since this function prunes expired contacts before returning, it holds a named write * lock on the aor. If you already hold the lock, call ast_sip_location_retrieve_aor_contacts_nolock instead. */ struct ao2_container *ast_sip_location_retrieve_aor_contacts_filtered(const struct ast_sip_aor *aor, unsigned int flags); /*! * \brief Retrieve all contacts currently available for an AOR without locking the AOR * \since 13.9.0 * * \param aor Pointer to the AOR * * \retval NULL if no contacts available * \retval non-NULL if contacts available * * \warning * This function should only be called if you already hold a named write lock on the aor. */ struct ao2_container *ast_sip_location_retrieve_aor_contacts_nolock(const struct ast_sip_aor *aor); /*! * \brief Retrieve all contacts currently available for an AOR without locking the AOR and filter based on flags * \since 13.16.0 * * \param aor Pointer to the AOR * \param flags Filtering flags * * \retval NULL if no contacts available * \retval non-NULL if contacts available * * \warning * This function should only be called if you already hold a named write lock on the aor. */ struct ao2_container *ast_sip_location_retrieve_aor_contacts_nolock_filtered(const struct ast_sip_aor *aor, unsigned int flags); /*! * \brief Retrieve the first bound contact from a list of AORs * * \param aor_list A comma-separated list of AOR names * \retval NULL if no contacts available * \retval non-NULL if contacts available */ struct ast_sip_contact *ast_sip_location_retrieve_contact_from_aor_list(const char *aor_list); /*! * \brief Retrieve all contacts from a list of AORs * * \param aor_list A comma-separated list of AOR names * \retval NULL if no contacts available * \retval non-NULL container (which must be freed) if contacts available */ struct ao2_container *ast_sip_location_retrieve_contacts_from_aor_list(const char *aor_list); /*! * \brief Retrieve the first bound contact AND the AOR chosen from a list of AORs * * \param aor_list A comma-separated list of AOR names * \param aor The chosen AOR * \param contact The chosen contact */ void ast_sip_location_retrieve_contact_and_aor_from_list(const char *aor_list, struct ast_sip_aor **aor, struct ast_sip_contact **contact); /*! * \brief Retrieve the first bound contact AND the AOR chosen from a list of AORs and filter based on flags * \since 13.16.0 * * \param aor_list A comma-separated list of AOR names * \param flags Filtering flags * \param aor The chosen AOR * \param contact The chosen contact */ void ast_sip_location_retrieve_contact_and_aor_from_list_filtered(const char *aor_list, unsigned int flags, struct ast_sip_aor **aor, struct ast_sip_contact **contact); /*! * \brief Retrieve a named contact * * \param contact_name Name of the contact * * \retval NULL if not found * \retval non-NULL if found */ struct ast_sip_contact *ast_sip_location_retrieve_contact(const char *contact_name); /*! * \brief Add a new contact to an AOR * * \param aor Pointer to the AOR * \param uri Full contact URI * \param expiration_time Optional expiration time of the contact * \param path_info Path information * \param user_agent User-Agent header from REGISTER request * \param via_addr * \param via_port * \param call_id * \param endpoint The endpoint that resulted in the contact being added * * \retval -1 failure * \retval 0 success * * \warning * This function holds a named write lock on the aor. If you already hold the lock * you should call ast_sip_location_add_contact_nolock instead. */ int ast_sip_location_add_contact(struct ast_sip_aor *aor, const char *uri, struct timeval expiration_time, const char *path_info, const char *user_agent, const char *via_addr, int via_port, const char *call_id, struct ast_sip_endpoint *endpoint); /*! * \brief Add a new contact to an AOR without locking the AOR * \since 13.9.0 * * \param aor Pointer to the AOR * \param uri Full contact URI * \param expiration_time Optional expiration time of the contact * \param path_info Path information * \param user_agent User-Agent header from REGISTER request * \param via_addr * \param via_port * \param call_id * \param endpoint The endpoint that resulted in the contact being added * * \retval -1 failure * \retval 0 success * * \warning * This function should only be called if you already hold a named write lock on the aor. */ int ast_sip_location_add_contact_nolock(struct ast_sip_aor *aor, const char *uri, struct timeval expiration_time, const char *path_info, const char *user_agent, const char *via_addr, int via_port, const char *call_id, struct ast_sip_endpoint *endpoint); /*! * \brief Create a new contact for an AOR without locking the AOR * \since 13.18.0 * * \param aor Pointer to the AOR * \param uri Full contact URI * \param expiration_time Optional expiration time of the contact * \param path_info Path information * \param user_agent User-Agent header from REGISTER request * \param via_addr * \param via_port * \param call_id * \param prune_on_boot Non-zero if the contact cannot survive a restart/boot. * \param endpoint The endpoint that resulted in the contact being added * * \return The created contact or NULL on failure. * * \warning * This function should only be called if you already hold a named write lock on the aor. */ struct ast_sip_contact *ast_sip_location_create_contact(struct ast_sip_aor *aor, const char *uri, struct timeval expiration_time, const char *path_info, const char *user_agent, const char *via_addr, int via_port, const char *call_id, int prune_on_boot, struct ast_sip_endpoint *endpoint); /*! * \brief Update a contact * * \param contact New contact object with details * * \retval -1 failure * \retval 0 success */ int ast_sip_location_update_contact(struct ast_sip_contact *contact); /*! * \brief Delete a contact * * \param contact Contact object to delete * * \retval -1 failure * \retval 0 success */ int ast_sip_location_delete_contact(struct ast_sip_contact *contact); /*! * \brief Prune the prune_on_boot contacts * \since 13.18.0 */ void ast_sip_location_prune_boot_contacts(void); /*! * \brief Callback called when an outbound request with authentication credentials is to be sent in dialog * * This callback will have the created request on it. The callback's purpose is to do any extra * housekeeping that needs to be done as well as to send the request out. * * This callback is only necessary if working with a PJSIP API that sits between the application * and the dialog layer. * * \param dlg The dialog to which the request belongs * \param tdata The created request to be sent out * \param user_data Data supplied with the callback * * \retval 0 Success * \retval -1 Failure */ typedef int (*ast_sip_dialog_outbound_auth_cb)(pjsip_dialog *dlg, pjsip_tx_data *tdata, void *user_data); /*! * \brief Set up outbound authentication on a SIP dialog * * This sets up the infrastructure so that all requests associated with a created dialog * can be re-sent with authentication credentials if the original request is challenged. * * \param dlg The dialog on which requests will be authenticated * \param endpoint The endpoint whom this dialog pertains to * \param cb Callback to call to send requests with authentication * \param user_data Data to be provided to the callback when it is called * * \retval 0 Success * \retval -1 Failure */ int ast_sip_dialog_setup_outbound_authentication(pjsip_dialog *dlg, const struct ast_sip_endpoint *endpoint, ast_sip_dialog_outbound_auth_cb cb, void *user_data); /*! * \brief Retrieves a reference to the artificial auth. * * \retval The artificial auth */ struct ast_sip_auth *ast_sip_get_artificial_auth(void); /*! * \brief Retrieves a reference to the artificial endpoint. * * \retval The artificial endpoint */ struct ast_sip_endpoint *ast_sip_get_artificial_endpoint(void); /*! \defgroup pjsip_threading PJSIP Threading Model * @{ * \page PJSIP PJSIP Threading Model * * There are three major types of threads that SIP will have to deal with: * \li Asterisk threads * \li PJSIP threads * \li SIP threadpool threads (a.k.a. "servants") * * \par Asterisk Threads * * Asterisk threads are those that originate from outside of SIP but within * Asterisk. The most common of these threads are PBX (channel) threads and * the autoservice thread. Most interaction with these threads will be through * channel technology callbacks. Within these threads, it is fine to handle * Asterisk data from outside of SIP, but any handling of SIP data should be * left to servants, \b especially if you wish to call into PJSIP for anything. * Asterisk threads are not registered with PJLIB, so attempting to call into * PJSIP will cause an assertion to be triggered, thus causing the program to * crash. * * \par PJSIP Threads * * PJSIP threads are those that originate from handling of PJSIP events, such * as an incoming SIP request or response, or a transaction timeout. The role * of these threads is to process information as quickly as possible so that * the next item on the SIP socket(s) can be serviced. On incoming messages, * Asterisk automatically will push the request to a servant thread. When your * module callback is called, processing will already be in a servant. However, * for other PJSIP events, such as transaction state changes due to timer * expirations, your module will be called into from a PJSIP thread. If you * are called into from a PJSIP thread, then you should push whatever processing * is needed to a servant as soon as possible. You can discern if you are currently * in a SIP servant thread using the \ref ast_sip_thread_is_servant function. * * \par Servants * * Servants are where the bulk of SIP work should be performed. These threads * exist in order to do the work that Asterisk threads and PJSIP threads hand * off to them. Servant threads register themselves with PJLIB, meaning that * they are capable of calling PJSIP and PJLIB functions if they wish. * * \par Serializer * * Tasks are handed off to servant threads using the API call \ref ast_sip_push_task. * The first parameter of this call is a serializer. If this pointer * is NULL, then the work will be handed off to whatever servant can currently handle * the task. If this pointer is non-NULL, then the task will not be executed until * previous tasks pushed with the same serializer have completed. For more information * on serializers and the benefits they provide, see \ref ast_threadpool_serializer * * \par Scheduler * * Some situations require that a task run periodically or at a future time. Normally * the ast_sched functionality would be used but ast_sched only uses 1 thread for all * tasks and that thread isn't registered with PJLIB and therefore can't do any PJSIP * related work. * * ast_sip_sched uses ast_sched only as a scheduled queue. When a task is ready to run, * it's pushed to a Serializer to be invoked asynchronously by a Servant. This ensures * that the task is executed in a PJLIB registered thread and allows the ast_sched thread * to immediately continue processing the queue. The Serializer used by ast_sip_sched * is one of your choosing or a random one from the res_pjsip pool if you don't choose one. * * \note * * Do not make assumptions about individual threads based on a corresponding serializer. * In other words, just because several tasks use the same serializer when being pushed * to servants, it does not mean that the same thread is necessarily going to execute those * tasks, even though they are all guaranteed to be executed in sequence. */ typedef int (*ast_sip_task)(void *user_data); /*! * \brief Create a new serializer for SIP tasks * \since 13.8.0 * * See \ref ast_threadpool_serializer for more information on serializers. * SIP creates serializers so that tasks operating on similar data will run * in sequence. * * \param name Name of the serializer. (must be unique) * * \retval NULL Failure * \retval non-NULL Newly-created serializer */ struct ast_taskprocessor *ast_sip_create_serializer(const char *name); struct ast_serializer_shutdown_group; /*! * \brief Create a new serializer for SIP tasks * \since 13.8.0 * * See \ref ast_threadpool_serializer for more information on serializers. * SIP creates serializers so that tasks operating on similar data will run * in sequence. * * \param name Name of the serializer. (must be unique) * \param shutdown_group Group shutdown controller. (NULL if no group association) * * \retval NULL Failure * \retval non-NULL Newly-created serializer */ struct ast_taskprocessor *ast_sip_create_serializer_group(const char *name, struct ast_serializer_shutdown_group *shutdown_group); /*! * \brief Determine the distributor serializer for the SIP message. * \since 13.10.0 * * \param rdata The incoming message. * * \retval Calculated distributor serializer on success. * \retval NULL on error. */ struct ast_taskprocessor *ast_sip_get_distributor_serializer(pjsip_rx_data *rdata); /*! * \brief Set a serializer on a SIP dialog so requests and responses are automatically serialized * * Passing a NULL serializer is a way to remove a serializer from a dialog. * * \param dlg The SIP dialog itself * \param serializer The serializer to use */ void ast_sip_dialog_set_serializer(pjsip_dialog *dlg, struct ast_taskprocessor *serializer); /*! * \brief Set an endpoint on a SIP dialog so in-dialog requests do not undergo endpoint lookup. * * \param dlg The SIP dialog itself * \param endpoint The endpoint that this dialog is communicating with */ void ast_sip_dialog_set_endpoint(pjsip_dialog *dlg, struct ast_sip_endpoint *endpoint); /*! * \brief Get the endpoint associated with this dialog * * This function increases the refcount of the endpoint by one. Release * the reference once you are finished with the endpoint. * * \param dlg The SIP dialog from which to retrieve the endpoint * \retval NULL No endpoint associated with this dialog * \retval non-NULL The endpoint. */ struct ast_sip_endpoint *ast_sip_dialog_get_endpoint(pjsip_dialog *dlg); /*! * \brief Pushes a task to SIP servants * * This uses the serializer provided to determine how to push the task. * If the serializer is NULL, then the task will be pushed to the * servants directly. If the serializer is non-NULL, then the task will be * queued behind other tasks associated with the same serializer. * * \param serializer The serializer to which the task belongs. Can be NULL * \param sip_task The task to execute * \param task_data The parameter to pass to the task when it executes * \retval 0 Success * \retval -1 Failure */ int ast_sip_push_task(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data); /*! * \brief Push a task to SIP servants and wait for it to complete. * * Like \ref ast_sip_push_task except that it blocks until the task * completes. If the current thread is a SIP servant thread then the * task executes immediately. Otherwise, the specified serializer * executes the task and the current thread waits for it to complete. * * \note PJPROJECT callbacks tend to have locks already held when * called. * * \warning \b Never hold locks that may be acquired by a SIP servant * thread when calling this function. Doing so may cause a deadlock * if all SIP servant threads are blocked waiting to acquire the lock * while the thread holding the lock is waiting for a free SIP servant * thread. * * \warning \b Use of this function in an ao2 destructor callback is a * bad idea. You don't have control over which thread executes the * destructor. Attempting to shift execution to another thread with * this function is likely to cause deadlock. * * \param serializer The SIP serializer to execute the task if the * current thread is not a SIP servant. NULL if any of the default * serializers can be used. * \param sip_task The task to execute * \param task_data The parameter to pass to the task when it executes * * \note The sip_task() return value may need to be distinguished from * the failure to push the task. * * \return sip_task() return value on success. * \retval -1 Failure to push the task. */ int ast_sip_push_task_wait_servant(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data); /*! * \brief Push a task to SIP servants and wait for it to complete. * \deprecated Replaced with ast_sip_push_task_wait_servant(). */ int ast_sip_push_task_synchronous(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data); /*! * \brief Push a task to the serializer and wait for it to complete. * * Like \ref ast_sip_push_task except that it blocks until the task is * completed by the specified serializer. If the specified serializer * is the current thread then the task executes immediately. * * \note PJPROJECT callbacks tend to have locks already held when * called. * * \warning \b Never hold locks that may be acquired by a SIP servant * thread when calling this function. Doing so may cause a deadlock * if all SIP servant threads are blocked waiting to acquire the lock * while the thread holding the lock is waiting for a free SIP servant * thread for the serializer to execute in. * * \warning \b Never hold locks that may be acquired by the serializer * when calling this function. Doing so will cause a deadlock. * * \warning \b Never use this function in the pjsip monitor thread (It * is a SIP servant thread). This is likely to cause a deadlock. * * \warning \b Use of this function in an ao2 destructor callback is a * bad idea. You don't have control over which thread executes the * destructor. Attempting to shift execution to another thread with * this function is likely to cause deadlock. * * \param serializer The SIP serializer to execute the task. NULL if * any of the default serializers can be used. * \param sip_task The task to execute * \param task_data The parameter to pass to the task when it executes * * \note It is generally better to call * ast_sip_push_task_wait_servant() if you pass NULL for the * serializer parameter. * * \note The sip_task() return value may need to be distinguished from * the failure to push the task. * * \return sip_task() return value on success. * \retval -1 Failure to push the task. */ int ast_sip_push_task_wait_serializer(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data); /*! * \brief Determine if the current thread is a SIP servant thread * * \retval 0 This is not a SIP servant thread * \retval 1 This is a SIP servant thread */ int ast_sip_thread_is_servant(void); /*! * \brief Task flags for the res_pjsip scheduler * * The default is AST_SIP_SCHED_TASK_FIXED * | AST_SIP_SCHED_TASK_DATA_NOT_AO2 * | AST_SIP_SCHED_TASK_DATA_NO_CLEANUP * | AST_SIP_SCHED_TASK_PERIODIC */ enum ast_sip_scheduler_task_flags { /*! * The defaults */ AST_SIP_SCHED_TASK_DEFAULTS = (0 << 0), /*! * Run at a fixed interval. * Stop scheduling if the callback returns <= 0. * Any other value is ignored. */ AST_SIP_SCHED_TASK_FIXED = (0 << 0), /*! * Run at a variable interval. * Stop scheduling if the callback returns <= 0. * Any other return value is used as the new interval. */ AST_SIP_SCHED_TASK_VARIABLE = (1 << 0), /*! * The task data is not an AO2 object. */ AST_SIP_SCHED_TASK_DATA_NOT_AO2 = (0 << 1), /*! * The task data is an AO2 object. * A reference count will be held by the scheduler until * after the task has run for the final time (if ever). */ AST_SIP_SCHED_TASK_DATA_AO2 = (1 << 1), /*! * Don't take any cleanup action on the data */ AST_SIP_SCHED_TASK_DATA_NO_CLEANUP = (0 << 3), /*! * If AST_SIP_SCHED_TASK_DATA_AO2 is set, decrement the reference count * otherwise call ast_free on it. */ AST_SIP_SCHED_TASK_DATA_FREE = ( 1 << 3 ), /*! * \brief The task is scheduled at multiples of interval * \see Interval */ AST_SIP_SCHED_TASK_PERIODIC = (0 << 4), /*! * \brief The next invocation of the task is at last finish + interval * \see Interval */ AST_SIP_SCHED_TASK_DELAY = (1 << 4), /*! * \brief The scheduled task's events are tracked in the debug log. * \details * Schedule events such as scheduling, running, rescheduling, canceling, * and destroying are logged about the task. */ AST_SIP_SCHED_TASK_TRACK = (1 << 5), }; /*! * \brief Scheduler task data structure */ struct ast_sip_sched_task; /*! * \brief Schedule a task to run in the res_pjsip thread pool * \since 13.9.0 * * \param serializer The serializer to use. If NULL, don't use a serializer (see note below) * \param interval The invocation interval in milliseconds (see note below) * \param sip_task The task to invoke * \param name An optional name to associate with the task * \param task_data Optional data to pass to the task * \param flags One of enum ast_sip_scheduler_task_type * * \returns Pointer to \ref ast_sip_sched_task ao2 object which must be dereferenced when done. * * \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, const char *name, void *task_data, enum ast_sip_scheduler_task_flags flags); /*! * \brief Cancels the next invocation of a task * \since 13.9.0 * * \param schtd The task structure pointer * \retval 0 Success * \retval -1 Failure * \note Only cancels future invocations not the currently running invocation. */ int ast_sip_sched_task_cancel(struct ast_sip_sched_task *schtd); /*! * \brief Cancels the next invocation of a task by name * \since 13.9.0 * * \param name The task name * \retval 0 Success * \retval -1 Failure * \note Only cancels future invocations not the currently running invocation. */ int ast_sip_sched_task_cancel_by_name(const char *name); /*! * \brief Gets the last start and end times of the task * \since 13.9.0 * * \param schtd The task structure pointer * \param[out] when_queued Pointer to a timeval structure to contain the time when queued * \param[out] last_start Pointer to a timeval structure to contain the time when last started * \param[out] last_end Pointer to a timeval structure to contain the time when last ended * \retval 0 Success * \retval -1 Failure * \note Any of the pointers can be NULL if you don't need them. */ int ast_sip_sched_task_get_times(struct ast_sip_sched_task *schtd, struct timeval *when_queued, struct timeval *last_start, struct timeval *last_end); /*! * \brief Gets the 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 * \since 13.15.0 * * \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 * \param contact_uri The contact URI of the message * * \retval 0 success * \retval -1 failure */ int ast_sip_create_rdata_with_contact(pjsip_rx_data *rdata, char *packet, const char *src_name, int src_port, char *transport_type, const char *local_name, int local_port, const char *contact_uri); /*! * \brief General purpose method for creating an rdata structure using specific information * * \param 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 Create and copy a pj_str_t into a standard character buffer. * * pj_str_t is not NULL-terminated. Any place that expects a NULL- * terminated string needs to have the pj_str_t copied into a separate * buffer. * * Copies the pj_str_t contents into a newly allocated buffer pointed to * by dest. NULL-terminates the buffer. * * \note Caller is responsible for freeing the allocated memory. * * \param dest [out] The destination buffer * \param src The pj_str_t to copy * \retval Number of characters copied or negative value on error */ int ast_copy_pj_str2(char **dest, const pj_str_t *src); /*! * \brief Get the looked-up endpoint on an out-of dialog request or response * * The function may ONLY be called on out-of-dialog requests or responses. For * in-dialog requests and responses, it is required that the user of the dialog * has the looked-up endpoint stored locally. * * This function should never return NULL if the message is out-of-dialog. It will * always return NULL if the message is in-dialog. * * This function will increase the reference count of the returned endpoint by one. * Release your reference using the ao2_ref function when finished. * * \param rdata Out-of-dialog request or response * \return The looked up endpoint */ struct ast_sip_endpoint *ast_pjsip_rdata_get_endpoint(pjsip_rx_data *rdata); /*! * \brief Add 'user=phone' parameter to URI if enabled and user is a phone number. * * \param endpoint The endpoint to use for configuration * \param pool The memory pool to allocate the parameter from * \param uri The URI to check for user and to add parameter to */ void ast_sip_add_usereqphone(const struct ast_sip_endpoint *endpoint, pj_pool_t *pool, pjsip_uri *uri); /*! * \brief Retrieve any endpoints available to sorcery. * * \retval Endpoints available to sorcery, NULL if no endpoints found. */ struct ao2_container *ast_sip_get_endpoints(void); /*! * \brief Retrieve the default outbound endpoint. * * \retval The default outbound endpoint, NULL if not found. */ struct ast_sip_endpoint *ast_sip_default_outbound_endpoint(void); /*! * \brief Retrieve relevant SIP auth structures from sorcery * * \param auths Vector of sorcery IDs of auth credentials to retrieve * \param[out] out The retrieved auths are stored here */ int ast_sip_retrieve_auths(const struct ast_sip_auth_vector *auths, struct ast_sip_auth **out); /*! * \brief Clean up retrieved auth structures from memory * * Call this function once you have completed operating on auths * retrieved from \ref ast_sip_retrieve_auths * * \param auths An 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 */ void ast_sip_register_endpoint_formatter(struct ast_sip_endpoint_formatter *obj); /*! * \brief Unregister an endpoint formatter. * * \param obj the formatter to unregister */ void ast_sip_unregister_endpoint_formatter(struct ast_sip_endpoint_formatter *obj); /*! * \brief Converts a sorcery object to a string of object properties. * * \param obj the sorcery object to convert * \param 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 Formats the contact and sends over AMI. * * \param obj a pointer an ast_sip_contact_wrapper structure * \param arg a pointer to an ast_sip_ami structure * \param flags ignored * \retval 0 Success, otherwise non-zero on error */ int ast_sip_format_contact_ami(void *obj, void *arg, int flags); /*! * \brief Format auth details for AMI. * * \param auths an auth array * \param ami ami variable container * \retval 0 Success, non-zero on failure */ int ast_sip_format_auths_ami(const struct ast_sip_auth_vector *auths, struct ast_sip_ami *ami); /*! * \brief Retrieve the endpoint snapshot for an endpoint * * \param endpoint The endpoint whose snapshot is to be 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 */ void ast_sip_register_supplement(struct ast_sip_supplement *supplement); /*! * \brief Unregister a an supplement to SIP out of dialog processing * * \param supplement The supplement to unregister */ void ast_sip_unregister_supplement(struct ast_sip_supplement *supplement); /*! * \brief Retrieve the global MWI taskprocessor high water alert trigger level. * * \since 13.12.0 * * \retval the system MWI taskprocessor high water alert trigger level */ unsigned int ast_sip_get_mwi_tps_queue_high(void); /*! * \brief Retrieve the global MWI taskprocessor low water clear alert level. * * \since 13.12.0 * * \retval the system MWI taskprocessor low water clear alert level */ int ast_sip_get_mwi_tps_queue_low(void); /*! * \brief Retrieve the global setting 'disable sending unsolicited mwi on startup'. * \since 13.12.0 * * \retval non zero if disable. */ unsigned int ast_sip_get_mwi_disable_initial_unsolicited(void); /*! * \brief Retrieve the global setting '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 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); /*! * \brief Get the transport name from an endpoint or request uri * \since 13.15.0 * * \param endpoint * \param sip_uri * \param buf Buffer to receive transport name * \param buf_len Buffer length * * \retval 0 Success * \retval -1 Failure * * \note * If endpoint->transport is not NULL, it is returned in buf. * Otherwise if sip_uri has an 'x-ast-txp' parameter AND the sip_uri host is * an ip4 or ip6 address, its value is returned, */ int ast_sip_get_transport_name(const struct ast_sip_endpoint *endpoint, pjsip_sip_uri *sip_uri, char *buf, size_t buf_len); /*! * \brief Sets pjsip_tpselector from an endpoint or uri * \since 13.15.0 * * \param endpoint If endpoint->transport is set, it's used * \param sip_uri If sip_uri contains a x-ast-txp parameter, it's used * \param selector The selector to be populated * * \retval 0 success * \retval -1 failure */ int ast_sip_set_tpselector_from_ep_or_uri(const struct ast_sip_endpoint *endpoint, pjsip_sip_uri *sip_uri, pjsip_tpselector *selector); /*! * \brief Set the transport on a dialog * \since 13.15.0 * * \param endpoint * \param dlg * \param selector (optional) * * \note * This API calls ast_sip_get_transport_name(endpoint, dlg->target) and if the result is * non-NULL, calls pjsip_dlg_set_transport. If 'selector' is non-NULL, it is updated with * the selector used. */ int ast_sip_dlg_set_transport(const struct ast_sip_endpoint *endpoint, pjsip_dialog *dlg, pjsip_tpselector *selector); /*! * \brief Convert the DTMF mode enum value into a string * \since 13.18.0 * * \param dtmf the dtmf mode * \param buf Buffer to receive dtmf mode string * \param buf_len Buffer length * * \retval 0 Success * \retval -1 Failure * */ int ast_sip_dtmf_to_str(const enum ast_sip_dtmf_mode dtmf, char *buf, size_t buf_len); /*! * \brief Convert the DTMF mode name into an enum * \since 13.18.0 * * \param dtmf_mode dtmf mode as a string * * \retval >= 0 The enum value * \retval -1 Failure * */ int ast_sip_str_to_dtmf(const char *dtmf_mode); /*! * \brief Transport shutdown monitor callback. * \since 13.18.0 * * \param data User data to know what to do when transport shuts down. * * \note The callback does not need to care that data is an ao2 object. * * \return Nothing */ typedef void (*ast_transport_monitor_shutdown_cb)(void *data); /*! * \brief Transport shutdown monitor data matcher * \since 13.20.0 * * \param a User data to compare. * \param b User data to compare. * * \retval 1 The data objects match * \retval 0 The data objects don't match */ typedef int (*ast_transport_monitor_data_matcher)(void *a, void *b); enum ast_transport_monitor_reg { /*! \brief Successfully registered the transport monitor */ AST_TRANSPORT_MONITOR_REG_SUCCESS, /*! \brief Replaced the already existing transport monitor with new one. */ AST_TRANSPORT_MONITOR_REG_REPLACED, /*! * \brief Transport not found to monitor. * \note Transport is either already shutdown or is not reliable. */ AST_TRANSPORT_MONITOR_REG_NOT_FOUND, /*! \brief Error while registering transport monitor. */ AST_TRANSPORT_MONITOR_REG_FAILED, }; /*! * \brief Register a reliable transport shutdown monitor callback. * \since 13.20.0 * * \param transport Transport to monitor for shutdown. * \param cb Who to call when transport is shutdown. * \param ao2_data Data to pass with the callback. * * \note The data object passed will have its reference count automatically * incremented by this call and automatically decremented after the callback * runs or when the callback is unregistered. * * There is no checking for duplicate registrations. * * \return enum ast_transport_monitor_reg */ enum ast_transport_monitor_reg ast_sip_transport_monitor_register(pjsip_transport *transport, ast_transport_monitor_shutdown_cb cb, void *ao2_data); /*! * \brief Unregister a reliable transport shutdown monitor * \since 13.20.0 * * \param transport Transport to monitor for shutdown. * \param cb The callback that was used for the original register. * \param data Data to pass to the matcher. May be NULL and does NOT need to be an ao2 object. * If NULL, all monitors with the provided callbck are unregistered. * \param matches Matcher function that returns true if data matches the previously * registered data object. If NULL, a simple pointer comparison is done. * * \note The data object passed into the original register will have its reference count * automatically decremeneted. * * \return Nothing */ void ast_sip_transport_monitor_unregister(pjsip_transport *transport, ast_transport_monitor_shutdown_cb cb, void *data, ast_transport_monitor_data_matcher matches); /*! * \brief Unregister a transport shutdown monitor from all reliable transports * \since 13.20.0 * * \param cb The callback that was used for the original register. * \param data Data to pass to the matcher. May be NULL and does NOT need to be an ao2 object. * If NULL, all monitors with the provided callbck are unregistered. * \param matches Matcher function that returns true if ao2_data matches the previously * registered data object. If NULL, a simple pointer comparison is done. * * \note The data object passed into the original register will have its reference count * automatically decremeneted. * * \return Nothing */ void ast_sip_transport_monitor_unregister_all(ast_transport_monitor_shutdown_cb cb, void *data, ast_transport_monitor_data_matcher matches); /*! Transport state notification registration element. */ struct ast_sip_tpmgr_state_callback { /*! PJPROJECT transport state notification callback */ pjsip_tp_state_callback cb; AST_LIST_ENTRY(ast_sip_tpmgr_state_callback) node; }; /*! * \brief Register a transport state notification callback element. * \since 13.18.0 * * \param element What we are registering. * * \return Nothing */ void ast_sip_transport_state_register(struct ast_sip_tpmgr_state_callback *element); /*! * \brief Unregister a transport state notification callback element. * \since 13.18.0 * * \param element What we are unregistering. * * \return Nothing */ void ast_sip_transport_state_unregister(struct ast_sip_tpmgr_state_callback *element); #endif /* _RES_PJSIP_H */