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Diffstat (limited to 'include/asterisk/res_pjsip.h')
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diff --git a/include/asterisk/res_pjsip.h b/include/asterisk/res_pjsip.h new file mode 100644 index 000000000..23d1a641e --- /dev/null +++ b/include/asterisk/res_pjsip.h @@ -0,0 +1,1502 @@ +/* + * Asterisk -- An open source telephony toolkit. + * + * Copyright (C) 2013, Digium, Inc. + * + * Mark Michelson <mmichelson@digium.com> + * + * See http://www.asterisk.org for more information about + * the Asterisk project. Please do not directly contact + * any of the maintainers of this project for assistance; + * the project provides a web site, mailing lists and IRC + * channels for your use. + * + * This program is free software, distributed under the terms of + * the GNU General Public License Version 2. See the LICENSE file + * at the top of the source tree. + */ + +#ifndef _RES_SIP_H +#define _RES_SIP_H + +#include "asterisk/stringfields.h" +/* Needed for struct ast_sockaddr */ +#include "asterisk/netsock2.h" +/* Needed for linked list macros */ +#include "asterisk/linkedlists.h" +/* Needed for ast_party_id */ +#include "asterisk/channel.h" +/* Needed for ast_sorcery */ +#include "asterisk/sorcery.h" +/* Needed for ast_dnsmgr */ +#include "asterisk/dnsmgr.h" +/* Needed for ast_endpoint */ +#include "asterisk/endpoints.h" +/* Needed for ast_t38_ec_modes */ +#include "asterisk/udptl.h" +/* Needed for pj_sockaddr */ +#include <pjlib.h> +/* Needed for ast_rtp_dtls_cfg struct */ +#include "asterisk/rtp_engine.h" + +/* 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 Structure for SIP transport information + */ +struct ast_sip_transport_state { + /*! \brief Transport itself */ + struct pjsip_transport *transport; + + /*! \brief Transport factory */ + struct pjsip_tpfactory *factory; +}; + +#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 Maximum number of ciphers supported for a TLS transport */ +#define SIP_TLS_MAX_CIPHERS 64 + +/* + * \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); + /*! 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; + /*! 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; + /*! TLS settings */ + pjsip_tls_setting tls; + /*! Configured TLS ciphers */ + pj_ssl_cipher ciphers[SIP_TLS_MAX_CIPHERS]; + /*! Optional local network information, used for NAT purposes */ + struct ast_ha *localnet; + /*! DNS manager for refreshing the external address */ + struct ast_dnsmgr_entry *external_address_refresher; + /*! Optional external address information */ + struct ast_sockaddr external_address; + /*! Transport state information */ + struct ast_sip_transport_state *state; + /*! QOS DSCP TOS bits */ + unsigned int tos; + /*! QOS COS value */ + unsigned int cos; +}; + +/*! + * \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); + ); + /*! Absolute time that this contact is no longer valid after */ + struct timeval expiration_time; + /*! Frequency to send OPTIONS requests to contact. 0 is disabled. */ + unsigned int qualify_frequency; + /*! If true authenticate the qualify if needed */ + int authenticate_qualify; +}; + +#define CONTACT_STATUS "contact_status" + +/*! + * \brief Status type for a contact. + */ +enum ast_sip_contact_status_type { + UNAVAILABLE, + AVAILABLE +}; + +/*! + * \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); + /*! 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; +}; + +/*! + * \brief A transport to be used for messages to a contact + */ +struct ast_sip_contact_transport { + AST_DECLARE_STRING_FIELDS( + /*! Full URI of the contact */ + AST_STRING_FIELD(uri); + ); + pjsip_transport *transport; +}; + +/*! + * \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); + ); + /*! Minimum expiration time */ + unsigned int minimum_expiration; + /*! Maximum expiration time */ + unsigned int maximum_expiration; + /*! Default contact expiration if one is not provided in the contact */ + unsigned int default_expiration; + /*! Frequency to send OPTIONS requests to AOR contacts. 0 is disabled. */ + unsigned int qualify_frequency; + /*! If true authenticate the qualify if needed */ + int authenticate_qualify; + /*! Maximum number of external contacts, 0 to disable */ + unsigned int max_contacts; + /*! Whether to remove any existing contacts not related to an incoming REGISTER when it comes in */ + unsigned int remove_existing; + /*! Any permanent configured contacts */ + struct ao2_container *permanent_contacts; +}; + +/*! + * \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, +}; + +/*! + * \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; +}; + +struct ast_sip_auth_array { + /*! Array of Sorcery IDs of auth sections */ + const char **names; + /*! Number of credentials in the array */ + unsigned int num; +}; + +/*! + * \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), +}; + +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, +}; + +/*! + * \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; +}; + +/*! + * \brief Endpoint subscription configuration + */ +struct ast_sip_endpoint_subscription_configuration { + /*! Indicates if endpoint is allowed to initiate subscriptions */ + unsigned int allow; + /*! The minimum allowed expiration for subscriptions from endpoint */ + unsigned int minexpiry; + /*! Message waiting configuration */ + struct ast_sip_mwi_configuration mwi; +}; + +/*! + * \brief NAT configuration options for endpoints + */ +struct ast_sip_endpoint_nat_configuration { + /*! Whether to force using the source IP address/port for sending responses */ + unsigned int force_rport; + /*! Whether to rewrite the Contact header with the source IP address/port or not */ + unsigned int rewrite_contact; +}; + +/*! + * \brief Party identification options for endpoints + * + * This includes caller ID, connected line, and redirecting-related options + */ +struct ast_sip_endpoint_id_configuration { + struct ast_party_id self; + /*! Do we accept identification information from this endpoint */ + unsigned int trust_inbound; + /*! Do we send private identification information to this endpoint? */ + unsigned int trust_outbound; + /*! Do we send P-Asserted-Identity headers to this endpoint? */ + unsigned int send_pai; + /*! Do we send Remote-Party-ID headers to this endpoint? */ + unsigned int send_rpid; + /*! Do we 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; + /*! \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; +}; + +/*! + * \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 external media address to use in SDP */ + AST_STRING_FIELD(external_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; + /*! Codec preferences */ + struct ast_codec_pref prefs; + /*! Configured codecs */ + struct ast_format_cap *codecs; + /*! DSCP TOS bits for audio streams */ + unsigned int tos_audio; + /*! Priority for audio streams */ + unsigned int cos_audio; + /*! DSCP TOS bits for video streams */ + unsigned int tos_video; + /*! Priority for video streams */ + unsigned int cos_video; +}; + +/*! + * \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); + ); + /*! 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_array inbound_auths; + /*! Outbound authentication credentials */ + struct ast_sip_auth_array 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; + /*! 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; +}; + +/*! + * \brief Initialize an auth array with the configured values. + * + * \param array Array 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_array_init(struct ast_sip_auth_array *array, const char *auth_names); + +/*! + * \brief Free contents of an auth array. + * + * \param array Array whose contents are to be freed + */ +void ast_sip_auth_array_destroy(struct ast_sip_auth_array *array); + +/*! + * \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 An array of IDs of auth sorcery objects + * \param challenge The SIP response with authentication challenge(s) + * \param tsx The transaction in which the challenge was received + * \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_array *auths, struct pjsip_rx_data *challenge, + struct pjsip_transaction *tsx, 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); +}; + +#define SIP_SORCERY_SECURITY_TYPE "security" + +/*! + * \brief SIP security details and configuration. + */ +struct ast_sip_security { + SORCERY_OBJECT(details); + struct ast_acl_list *acl; + struct ast_acl_list *contact_acl; +}; + +/*! + * \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 + * + * 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 + * + * \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 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 Initialize transport support on a sorcery instance + * + * \param sorcery The sorcery instance + * + * \retval -1 failure + * \retval 0 success + */ +int ast_sip_initialize_sorcery_transport(struct ast_sorcery *sorcery); + +/*! + * \brief Initialize qualify support on a sorcery instance + * + * \param sorcery The sorcery instance + * + * \retval -1 failure + * \retval 0 success + */ +int ast_sip_initialize_sorcery_qualify(struct ast_sorcery *sorcery); + +/*! + * \brief Initialize location support on a sorcery instance + * + * \param sorcery The sorcery instance + * + * \retval -1 failure + * \retval 0 success + */ +int ast_sip_initialize_sorcery_location(struct ast_sorcery *sorcery); + +/*! + * \brief Retrieve a named AOR + * + * \param aor_name Name of the AOR + * + * \retval NULL if not found + * \retval non-NULL if found + */ +struct ast_sip_aor *ast_sip_location_retrieve_aor(const char *aor_name); + +/*! + * \brief Retrieve the first bound contact for an AOR + * + * \param aor Pointer to the AOR + * \retval NULL if no contacts available + * \retval non-NULL if contacts available + */ +struct ast_sip_contact *ast_sip_location_retrieve_first_aor_contact(const struct ast_sip_aor *aor); + +/*! + * \brief Retrieve all contacts currently available for an AOR + * + * \param aor Pointer to the AOR + * + * \retval NULL if no contacts available + * \retval non-NULL if contacts available + */ +struct ao2_container *ast_sip_location_retrieve_aor_contacts(const struct ast_sip_aor *aor); + +/*! + * \brief Retrieve the first bound contact from a list of AORs + * + * \param aor_list A comma-separated list of AOR names + * \retval NULL if no contacts available + * \retval non-NULL if contacts available + */ +struct ast_sip_contact *ast_sip_location_retrieve_contact_from_aor_list(const char *aor_list); + +/*! + * \brief Retrieve 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 transport for a contact to use + */ + +void ast_sip_location_add_contact_transport(struct ast_sip_contact_transport *ct); + +/*! + * \brief Delete a transport for a contact that went away + */ +void ast_sip_location_delete_contact_transport(struct ast_sip_contact_transport *ct); + +/*! + * \brief Retrieve a contact_transport, by URI + * + * \param contact_uri URI of the contact + * + * \retval NULL if not found + * \retval non-NULL if found + */ +struct ast_sip_contact_transport *ast_sip_location_retrieve_contact_transport_by_uri(const char *contact_uri); + +/*! + * \brief Retrieve a contact_transport, by transport + * + * \param transport transport the contact uses + * + * \retval NULL if not found + * \retval non-NULL if found + */ +struct ast_sip_contact_transport *ast_sip_location_retrieve_contact_transport_by_transport(pjsip_transport *transport); + +/*! + * \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 + * + * \retval -1 failure + * \retval 0 success + */ +int ast_sip_location_add_contact(struct ast_sip_aor *aor, const char *uri, struct timeval expiration_time); + +/*! + * \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 Initialize domain aliases support on a sorcery instance + * + * \param sorcery The sorcery instance + * + * \retval -1 failure + * \retval 0 success + */ +int ast_sip_initialize_sorcery_domain_alias(struct ast_sorcery *sorcery); + +/*! + * \brief Initialize authentication support on a sorcery instance + * + * \param sorcery The sorcery instance + * + * \retval -1 failure + * \retval 0 success + */ +int ast_sip_initialize_sorcery_auth(struct ast_sorcery *sorcery); + +/*! + * \brief Initialize security support on a sorcery instance + * + * \param sorcery The sorcery instance + * + * \retval -1 failure + * \retval 0 success + */ +int ast_sip_initialize_sorcery_security(struct ast_sorcery *sorcery); + +/*! + * \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 Initialize the distributor module + * + * The distributor module is responsible for taking an incoming + * SIP message and placing it into the threadpool. Once in the threadpool, + * the distributor will perform endpoint lookups and authentication, and + * then distribute the message up the stack to any further modules. + * + * \retval -1 Failure + * \retval 0 Success + */ +int ast_sip_initialize_distributor(void); + +/*! + * \brief Destruct the distributor module. + * + * Unregisters pjsip modules and cleans up any allocated resources. + */ +void ast_sip_destroy_distributor(void); + +/*! + * \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); + +/*! + * \page Threading model for SIP + * + * There are three major types of threads that SIP will have to deal with: + * \li Asterisk threads + * \li PJSIP threads + * \li SIP threadpool threads (a.k.a. "servants") + * + * \par Asterisk Threads + * + * Asterisk threads are those that originate from outside of SIP but within + * Asterisk. The most common of these threads are PBX (channel) threads and + * the autoservice thread. Most interaction with these threads will be through + * channel technology callbacks. Within these threads, it is fine to handle + * Asterisk data from outside of SIP, but any handling of SIP data should be + * left to servants, \b especially if you wish to call into PJSIP for anything. + * Asterisk threads are not registered with PJLIB, so attempting to call into + * PJSIP will cause an assertion to be triggered, thus causing the program to + * crash. + * + * \par PJSIP Threads + * + * PJSIP threads are those that originate from handling of PJSIP events, such + * as an incoming SIP request or response, or a transaction timeout. The role + * of these threads is to process information as quickly as possible so that + * the next item on the SIP socket(s) can be serviced. On incoming messages, + * Asterisk automatically will push the request to a servant thread. When your + * module callback is called, processing will already be in a servant. However, + * for other PSJIP events, such as transaction state changes due to timer + * expirations, your module will be called into from a PJSIP thread. If you + * are called into from a PJSIP thread, then you should push whatever processing + * is needed to a servant as soon as possible. You can discern if you are currently + * in a SIP servant thread using the \ref ast_sip_thread_is_servant function. + * + * \par Servants + * + * Servants are where the bulk of SIP work should be performed. These threads + * exist in order to do the work that Asterisk threads and PJSIP threads hand + * off to them. Servant threads register themselves with PJLIB, meaning that + * they are capable of calling PJSIP and PJLIB functions if they wish. + * + * \par Serializer + * + * Tasks are handed off to servant threads using the API call \ref ast_sip_push_task. + * The first parameter of this call is a serializer. If this pointer + * is NULL, then the work will be handed off to whatever servant can currently handle + * the task. If this pointer is non-NULL, then the task will not be executed until + * previous tasks pushed with the same serializer have completed. For more information + * on serializers and the benefits they provide, see \ref ast_threadpool_serializer + * + * \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. + */ + +/*! + * \brief Create a new serializer for SIP tasks + * + * See \ref ast_threadpool_serializer for more information on serializers. + * SIP creates serializers so that tasks operating on similar data will run + * in sequence. + * + * \retval NULL Failure + * \retval non-NULL Newly-created serializer + */ +struct ast_taskprocessor *ast_sip_create_serializer(void); + +/*! + * \brief Set a serializer on a SIP dialog so requests and responses are automatically serialized + * + * Passing a NULL serializer is a way to remove a serializer from a dialog. + * + * \param dlg The SIP dialog itself + * \param serializer The serializer to use + */ +void ast_sip_dialog_set_serializer(pjsip_dialog *dlg, struct ast_taskprocessor *serializer); + +/*! + * \brief Set an endpoint on a SIP dialog so in-dialog requests do not undergo endpoint lookup. + * + * \param dlg The SIP dialog itself + * \param endpoint The endpoint that this dialog is communicating with + */ +void ast_sip_dialog_set_endpoint(pjsip_dialog *dlg, struct ast_sip_endpoint *endpoint); + +/*! + * \brief Get the endpoint associated with this dialog + * + * This function increases the refcount of the endpoint by one. Release + * the reference once you are finished with the endpoint. + * + * \param dlg The SIP dialog from which to retrieve the endpoint + * \retval NULL No endpoint associated with this dialog + * \retval non-NULL The endpoint. + */ +struct ast_sip_endpoint *ast_sip_dialog_get_endpoint(pjsip_dialog *dlg); + +/*! + * \brief Pushes a task to SIP servants + * + * This uses the serializer provided to determine how to push the task. + * If the serializer is NULL, then the task will be pushed to the + * servants directly. If the serializer is non-NULL, then the task will be + * queued behind other tasks associated with the same serializer. + * + * \param serializer The serializer to which the task belongs. Can be NULL + * \param sip_task The task to execute + * \param task_data The parameter to pass to the task when it executes + * \retval 0 Success + * \retval -1 Failure + */ +int ast_sip_push_task(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data); + +/*! + * \brief Push a task to SIP servants and wait for it to complete + * + * Like \ref ast_sip_push_task except that it blocks until the task completes. + * + * \warning \b Never use this function in a SIP servant thread. This can potentially + * cause a deadlock. If you are in a SIP servant thread, just call your function + * in-line. + * + * \param serializer The SIP serializer to which the task belongs. May be NULL. + * \param sip_task The task to execute + * \param task_data The parameter to pass to the task when it executes + * \retval 0 Success + * \retval -1 Failure + */ +int ast_sip_push_task_synchronous(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data); + +/*! + * \brief Determine if the current thread is a SIP servant thread + * + * \retval 0 This is not a SIP servant thread + * \retval 1 This is a SIP servant thread + */ +int ast_sip_thread_is_servant(void); + +/*! + * \brief 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 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(const struct ast_sip_endpoint *endpoint, const char *aor_name, const char *request_user); + +/*! + * \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 + * this value. + * than one configured for the endpoint. + * \param[out] tdata The newly-created request + * \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, + 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. If specified, the dialog on which the request should be sent + * \param endpoint Optional. If specified, the request is sent out-of-dialog to the endpoint. + * \retval 0 Success + * \retval -1 Failure + */ +int ast_sip_send_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg, struct ast_sip_endpoint *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_array *auths, pjsip_rx_data *challenge, + pjsip_transaction *tsx, 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 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 array as part of a multipart body and + * add each part to the SIP message. + * + * \param tdata The message to add the body to + * \param bodies The parts of the body to add + * \retval 0 Success + * \retval -1 Failure + */ +int ast_sip_add_body_multipart(pjsip_tx_data *tdata, const struct ast_sip_body *bodies[], int num_bodies); + +/*! + * \brief Append body data to a SIP message + * + * This acts mostly the same as ast_sip_add_body, except that rather than replacing + * a body if it currently exists, it appends data to an existing body. + * + * \param tdata The message to append the body to + * \param body The string to append to the end of the current body + * \retval 0 Success + * \retval -1 Failure + */ +int ast_sip_append_body(pjsip_tx_data *tdata, const char *body_text); + +/*! + * \brief Copy a pj_str_t into a standard character buffer. + * + * pj_str_t is not NULL-terminated. Any place that expects a NULL- + * terminated string needs to have the pj_str_t copied into a separate + * buffer. + * + * This method copies the pj_str_t contents into the destination buffer + * and NULL-terminates the buffer. + * + * \param dest The destination buffer + * \param src The pj_str_t to copy + * \param size The size of the destination buffer. + */ +void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size); + +/*! + * \brief Get the looked-up endpoint on an out-of dialog request or response + * + * The function may ONLY be called on out-of-dialog requests or responses. For + * in-dialog requests and responses, it is required that the user of the dialog + * has the looked-up endpoint stored locally. + * + * This function should never return NULL if the message is out-of-dialog. It will + * always return NULL if the message is in-dialog. + * + * This function will increase the reference count of the returned endpoint by one. + * Release your reference using the ao2_ref function when finished. + * + * \param rdata Out-of-dialog request or response + * \return The looked up endpoint + */ +struct ast_sip_endpoint *ast_pjsip_rdata_get_endpoint(pjsip_rx_data *rdata); + +/*! + * \brief 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 relevant SIP auth structures from sorcery + * + * \param auths Array 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_array *auths, struct ast_sip_auth **out); + +/*! + * \brief Clean up retrieved auth structures from memory + * + * Call this function once you have completed operating on auths + * retrieved from \ref ast_sip_retrieve_auths + * + * \param auths An array of auth structures to clean up + * \param num_auths The number of auths in the array + */ +void ast_sip_cleanup_auths(struct ast_sip_auth *auths[], size_t num_auths); + +/*! + * \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); + +void ast_sip_initialize_global_headers(void); +void ast_sip_destroy_global_headers(void); + +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); + +int ast_sip_initialize_sorcery_global(struct ast_sorcery *sorcery); + +#endif /* _RES_SIP_H */ |