diff options
Diffstat (limited to 'include/asterisk')
| -rw-r--r-- | include/asterisk/autoconfig.h.in | 76 | ||||
| -rw-r--r-- | include/asterisk/res_sip.h | 1092 | ||||
| -rw-r--r-- | include/asterisk/res_sip_pubsub.h | 346 | ||||
| -rw-r--r-- | include/asterisk/res_sip_session.h | 468 | ||||
| -rw-r--r-- | include/asterisk/sorcery.h | 7 | ||||
| -rw-r--r-- | include/asterisk/threadpool.h | 14 |
6 files changed, 1966 insertions, 37 deletions
diff --git a/include/asterisk/autoconfig.h.in b/include/asterisk/autoconfig.h.in index f7294b36e..60f2068e5 100644 --- a/include/asterisk/autoconfig.h.in +++ b/include/asterisk/autoconfig.h.in @@ -294,7 +294,7 @@ /* Define if your system has the GLOB_NOMAGIC headers. */ #undef HAVE_GLOB_NOMAGIC -/* Define if your system has the GMIME libraries. */ +/* Define to 1 if you have the GMime library. */ #undef HAVE_GMIME /* Define to indicate the GSM library */ @@ -306,7 +306,7 @@ /* Define to indicate that gsm.h has no prefix for its location */ #undef HAVE_GSM_HEADER -/* Define if your system has the GTK2 libraries. */ +/* Define to 1 if you have the gtk2 library. */ #undef HAVE_GTK2 /* Define to 1 if you have the Hoard Memory Allocator library. */ @@ -324,7 +324,7 @@ /* Define to 1 if you have the Iksemel Jabber library. */ #undef HAVE_IKSEMEL -/* Define if your system has the ILBC libraries. */ +/* Define to 1 if you have the System iLBC library. */ #undef HAVE_ILBC /* Define if your system has the UW IMAP Toolkit c-client library. */ @@ -376,7 +376,7 @@ /* Define to 1 if you have the OpenLDAP library. */ #undef HAVE_LDAP -/* Define if your system has the LIBEDIT libraries. */ +/* Define to 1 if you have the NetBSD Editline library library. */ #undef HAVE_LIBEDIT /* Define to 1 if you have the <libintl.h> header file. */ @@ -551,7 +551,7 @@ /* Define to indicate presence of the pg_encoding_to_char API. */ #undef HAVE_PGSQL_pg_encoding_to_char -/* Define if your system has the PJPROJECT libraries. */ +/* Define to 1 if you have the PJPROJECT library. */ #undef HAVE_PJPROJECT /* Define to 1 if your system defines IP_PKTINFO. */ @@ -854,19 +854,19 @@ /* Define to 1 if you have the `strtoq' function. */ #undef HAVE_STRTOQ -/* Define to 1 if `ifr_ifru.ifru_hwaddr' is a member of `struct ifreq'. */ +/* Define to 1 if `ifr_ifru.ifru_hwaddr' is member of `struct ifreq'. */ #undef HAVE_STRUCT_IFREQ_IFR_IFRU_IFRU_HWADDR -/* Define to 1 if `uid' is a member of `struct sockpeercred'. */ +/* Define to 1 if `uid' is member of `struct sockpeercred'. */ #undef HAVE_STRUCT_SOCKPEERCRED_UID -/* Define to 1 if `st_blksize' is a member of `struct stat'. */ +/* Define to 1 if `st_blksize' is member of `struct stat'. */ #undef HAVE_STRUCT_STAT_ST_BLKSIZE -/* Define to 1 if `cr_uid' is a member of `struct ucred'. */ +/* Define to 1 if `cr_uid' is member of `struct ucred'. */ #undef HAVE_STRUCT_UCRED_CR_UID -/* Define to 1 if `uid' is a member of `struct ucred'. */ +/* Define to 1 if `uid' is member of `struct ucred'. */ #undef HAVE_STRUCT_UCRED_UID /* Define to 1 if you have the mISDN Supplemental Services library. */ @@ -1144,12 +1144,12 @@ /* Define to the one symbol short name of this package. */ #undef PACKAGE_TARNAME -/* Define to the home page for this package. */ -#undef PACKAGE_URL - /* Define to the version of this package. */ #undef PACKAGE_VERSION +/* Define to 1 if the C compiler supports function prototypes. */ +#undef PROTOTYPES + /* Define to necessary symbol if this constant uses a non-standard name on your system. */ #undef PTHREAD_CREATE_JOINABLE @@ -1169,6 +1169,11 @@ /* Define to the type of arg 5 for `select'. */ #undef SELECT_TYPE_ARG5 +/* Define to 1 if the `setvbuf' function takes the buffering type as its + second argument and the buffer pointer as the third, as on System V before + release 3. */ +#undef SETVBUF_REVERSED + /* The size of `char *', as computed by sizeof. */ #undef SIZEOF_CHAR_P @@ -1204,39 +1209,24 @@ /* Define to a type of the same size as fd_set.fds_bits[[0]] */ #undef TYPEOF_FD_SET_FDS_BITS -/* Enable extensions on AIX 3, Interix. */ +/* Define to 1 if on AIX 3. + System headers sometimes define this. + We just want to avoid a redefinition error message. */ #ifndef _ALL_SOURCE # undef _ALL_SOURCE #endif -/* Enable GNU extensions on systems that have them. */ -#ifndef _GNU_SOURCE -# undef _GNU_SOURCE -#endif -/* Enable threading extensions on Solaris. */ -#ifndef _POSIX_PTHREAD_SEMANTICS -# undef _POSIX_PTHREAD_SEMANTICS -#endif -/* Enable extensions on HP NonStop. */ -#ifndef _TANDEM_SOURCE -# undef _TANDEM_SOURCE -#endif -/* Enable general extensions on Solaris. */ -#ifndef __EXTENSIONS__ -# undef __EXTENSIONS__ -#endif - /* Define to 1 if running on Darwin. */ #undef _DARWIN_UNLIMITED_SELECT -/* Enable large inode numbers on Mac OS X 10.5. */ -#ifndef _DARWIN_USE_64_BIT_INODE -# define _DARWIN_USE_64_BIT_INODE 1 -#endif - /* Number of bits in a file offset, on hosts where this is settable. */ #undef _FILE_OFFSET_BITS +/* Enable GNU extensions on systems that have them. */ +#ifndef _GNU_SOURCE +# undef _GNU_SOURCE +#endif + /* Define to 1 to make fseeko visible on some hosts (e.g. glibc 2.2). */ #undef _LARGEFILE_SOURCE @@ -1253,6 +1243,20 @@ /* Define to 1 if you need to in order for `stat' and other things to work. */ #undef _POSIX_SOURCE +/* Enable extensions on Solaris. */ +#ifndef __EXTENSIONS__ +# undef __EXTENSIONS__ +#endif +#ifndef _POSIX_PTHREAD_SEMANTICS +# undef _POSIX_PTHREAD_SEMANTICS +#endif +#ifndef _TANDEM_SOURCE +# undef _TANDEM_SOURCE +#endif + +/* Define like PROTOTYPES; this can be used by system headers. */ +#undef __PROTOTYPES + /* Define to empty if `const' does not conform to ANSI C. */ #undef const diff --git a/include/asterisk/res_sip.h b/include/asterisk/res_sip.h new file mode 100644 index 000000000..7cfc38260 --- /dev/null +++ b/include/asterisk/res_sip.h @@ -0,0 +1,1092 @@ +/* + * 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 pj_sockaddr */ +#include <pjlib.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 Types of supported transports + */ +enum ast_sip_transport_type { + AST_SIP_TRANSPORT_UDP, + AST_SIP_TRANSPORT_TCP, + AST_SIP_TRANSPORT_TLS, + /* XXX Websocket ? */ +}; + +/*! \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_sip_transport_type 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; +}; + +/*! + * \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; +}; + +/*! + * \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; + /*! 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, +}; + +#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; +}; + +/*! + * \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 source location of the SIP message */ + AST_SIP_ENDPOINT_IDENTIFY_BY_LOCATION = (1 << 1), +}; + +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, +}; + +/*! + * \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); + /*! Optional external media address to use in SDP */ + AST_STRING_FIELD(external_media_address); + /*! Configured voicemail boxes for this endpoint. Used for MWI */ + AST_STRING_FIELD(mailboxes); + ); + /*! Identification information for this endpoint */ + struct ast_party_id id; + /*! Domain to which this endpoint belongs */ + struct ast_sip_domain *domain; + /*! Address of record for incoming registrations */ + struct ast_sip_aor *aor; + /*! Codec preferences */ + struct ast_codec_pref prefs; + /*! Configured codecs */ + struct ast_format_cap *codecs; + /*! Names of inbound authentication credentials */ + const char **sip_inbound_auths; + /*! Number of configured auths */ + size_t num_inbound_auths; + /*! Names of outbound authentication credentials */ + const char **sip_outbound_auths; + /*! Number of configured outbound auths */ + size_t num_outbound_auths; + /*! DTMF mode to use with this endpoint */ + enum ast_sip_dtmf_mode dtmf; + /*! Whether IPv6 RTP is enabled or not */ + unsigned int rtp_ipv6; + /*! Whether symmetric RTP is enabled or not */ + unsigned int rtp_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; + /*! 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; + /*! Enabled SIP extensions */ + unsigned int extensions; + /*! Minimum session expiration period, in seconds */ + unsigned int min_se; + /*! Session expiration period, in seconds */ + unsigned int sess_expires; + /*! List of outbound registrations */ + AST_LIST_HEAD_NOLOCK(, ast_sip_registration) registrations; + /*! Frequency to send OPTIONS requests to endpoint. 0 is disabled. */ + unsigned int qualify_frequency; + /*! Method(s) by which the endpoint should be identified. */ + enum ast_sip_endpoint_identifier_type ident_method; + /*! Boolean indicating if direct_media is permissible */ + unsigned int direct_media; + /*! When using direct media, which method should be used */ + enum ast_sip_session_refresh_method direct_media_method; + /*! Take steps to mitigate glare for direct media */ + enum ast_sip_direct_media_glare_mitigation direct_media_glare_mitigation; + /*! Do not attempt direct media session refreshes if a media NAT is detected */ + unsigned int disable_direct_media_on_nat; + /*! Do we trust the endpoint with our outbound identity? */ + unsigned int trust_id_outbound; + /*! Do we trust identity information that originates externally (e.g. P-Asserted-Identity header)? */ + unsigned int trust_id_inbound; + /*! 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; + /*! Should unsolicited MWI be aggregated into a single NOTIFY? */ + unsigned int aggregate_mwi; +}; + +/*! + * \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 num_auths The number of IDs in the array + * \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 char **auths, size_t num_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); +}; + +/*! + * \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 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 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 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); + +/*! + * \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 + * 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 char **auths, size_t num_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, 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 relevant SIP auth structures from sorcery + * + * \param auth_names The sorcery IDs of auths to retrieve + * \param num_auths The number of auths to retrieve + * \param[out] out The retrieved auths are stored here + */ +int ast_sip_retrieve_auths(const char *auth_names[], size_t num_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); + +#endif /* _RES_SIP_H */ diff --git a/include/asterisk/res_sip_pubsub.h b/include/asterisk/res_sip_pubsub.h new file mode 100644 index 000000000..33614b285 --- /dev/null +++ b/include/asterisk/res_sip_pubsub.h @@ -0,0 +1,346 @@ +/* + * 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_PUBSUB_H +#define _RES_SIP_PUBSUB_H + +#include "asterisk/linkedlists.h" + +/* Forward declarations */ +struct pjsip_rx_data; +struct pjsip_tx_data; +struct pjsip_evsub; +struct ast_sip_endpoint; +struct ast_datastore; +struct ast_datastore_info; + +/*! + * \brief Opaque structure representing an RFC 3265 SIP subscription + */ +struct ast_sip_subscription; + +/*! + * \brief Role for the subscription that is being created + */ +enum ast_sip_subscription_role { + /* Sending SUBSCRIBEs, receiving NOTIFYs */ + AST_SIP_SUBSCRIBER, + /* Sending NOTIFYs, receiving SUBSCRIBEs */ + AST_SIP_NOTIFIER, +}; + +/*! + * \brief Data for responses to SUBSCRIBEs and NOTIFIEs + * + * Some of PJSIP's evsub callbacks expect us to provide them + * with data so that they can craft a response rather than have + * us create our own response. + * + * Filling in the structure is optional, since the framework + * will automatically respond with a 200 OK response if we do + * not provide it with any additional data. + */ +struct ast_sip_subscription_response_data { + /*! Status code of the response */ + int status_code; + /*! Optional status text */ + const char *status_text; + /*! Optional additional headers to add to the response */ + struct ast_variable *headers; + /*! Optional body to add to the response */ + struct ast_sip_body *body; +}; + +#define AST_SIP_MAX_ACCEPT 32 + +struct ast_sip_subscription_handler { + /*! The name of the event this handler deals with */ + const char *event_name; + /*! The types of body this handler accepts */ + const char *accept[AST_SIP_MAX_ACCEPT]; + + /*! + * \brief Called when a subscription is to be destroyed + * + * This is a subscriber and notifier callback. + * + * The handler is not expected to send any sort of requests or responses + * during this callback. The handler MUST, however, begin the destruction + * process for the subscription during this callback. + */ + void (*subscription_shutdown)(struct ast_sip_subscription *subscription); + + /*! + * \brief Called when a SUBSCRIBE arrives in order to create a new subscription + * + * This is a notifier callback. + * + * If the notifier wishes to accept the subscription, then it can create + * a new ast_sip_subscription to do so. + * + * If the notifier chooses to create a new subscription, then it must accept + * the incoming subscription using pjsip_evsub_accept() and it must also + * send an initial NOTIFY with the current subscription state. + * + * \param endpoint The endpoint from which we received the SUBSCRIBE + * \param rdata The SUBSCRIBE request + * \retval NULL The SUBSCRIBE has not been accepted + * \retval non-NULL The newly-created subscription + */ + struct ast_sip_subscription *(*new_subscribe)(struct ast_sip_endpoint *endpoint, + pjsip_rx_data *rdata); + + /*! + * \brief Called when an endpoint renews a subscription. + * + * This is a notifier callback. + * + * Because of the way that the PJSIP evsub framework works, it will automatically + * send a response to the SUBSCRIBE. However, the subscription handler must send + * a NOTIFY with the current subscription state when this callback is called. + * + * The response_data that is passed into this callback is used to craft what should + * be in the response to the incoming SUBSCRIBE. It is initialized with a 200 status + * code and all other parameters are empty. + * + * \param sub The subscription that is being renewed + * \param rdata The SUBSCRIBE request in question + * \param[out] response_data Data pertaining to the SIP response that should be + * sent to the SUBSCRIBE + */ + void (*resubscribe)(struct ast_sip_subscription *sub, + pjsip_rx_data *rdata, struct ast_sip_subscription_response_data *response_data); + + /*! + * \brief Called when a subscription times out. + * + * This is a notifier callback + * + * This indicates that the subscription has timed out. The subscription handler is + * expected to send a NOTIFY that terminates the subscription. + * + * \param sub The subscription that has timed out + */ + void (*subscription_timeout)(struct ast_sip_subscription *sub); + + /*! + * \brief Called when a subscription is terminated via a SUBSCRIBE or NOTIFY request + * + * This is a notifier and subscriber callback. + * + * The PJSIP subscription framework will automatically send the response to the + * request. If a notifier receives this callback, then the subscription handler + * is expected to send a final NOTIFY to terminate the subscription. + * + * \param sub The subscription being terminated + * \param rdata The request that terminated the subscription + */ + void (*subscription_terminated)(struct ast_sip_subscription *sub, pjsip_rx_data *rdata); + + /*! + * \brief Called when a subscription handler's outbound NOTIFY receives a response + * + * This is a notifier callback. + * + * \param sub The subscription + * \param rdata The NOTIFY response + */ + void (*notify_response)(struct ast_sip_subscription *sub, pjsip_rx_data *rdata); + + /*! + * \brief Called when a subscription handler receives an inbound NOTIFY + * + * This is a subscriber callback. + * + * Because of the way that the PJSIP evsub framework works, it will automatically + * send a response to the NOTIFY. By default this will be a 200 OK response, but + * this callback can change details of the response by returning response data + * to use. + * + * The response_data that is passed into this callback is used to craft what should + * be in the response to the incoming SUBSCRIBE. It is initialized with a 200 status + * code and all other parameters are empty. + * + * \param sub The subscription + * \param rdata The NOTIFY request + * \param[out] response_data Data pertaining to the SIP response that should be + * sent to the SUBSCRIBE + */ + void (*notify_request)(struct ast_sip_subscription *sub, + pjsip_rx_data *rdata, struct ast_sip_subscription_response_data *response_data); + + /*! + * \brief Called when it is time for a subscriber to resubscribe + * + * This is a subscriber callback. + * + * The subscriber can reresh the subscription using the pjsip_evsub_initiate() + * function. + * + * \param sub The subscription to refresh + * \retval 0 Success + * \retval non-zero Failure + */ + int (*refresh_subscription)(struct ast_sip_subscription *sub); + AST_LIST_ENTRY(ast_sip_subscription_handler) next; +}; + +/*! + * \brief Create a new ast_sip_subscription structure + * + * In most cases the pubsub core will create a general purpose subscription + * within PJSIP. However, PJSIP provides enhanced support for the following + * event packages: + * + * presence + * message-summary + * + * If either of these events are handled by the subscription handler, then + * the special-purpose event subscriptions will be created within PJSIP, + * and it will be expected that your subscription handler make use of the + * special PJSIP APIs. + * + * \param handler The subsription handler for this subscription + * \param role Whether we are acting as subscriber or notifier for this subscription + * \param endpoint The endpoint involved in this subscription + * \param rdata If acting as a notifier, the SUBSCRIBE request that triggered subscription creation + */ +struct ast_sip_subscription *ast_sip_create_subscription(const struct ast_sip_subscription_handler *handler, + enum ast_sip_subscription_role role, struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata); + + +/*! + * \brief Get the endpoint that is associated with this subscription + * + * This function will increase the reference count of the endpoint. Be sure to + * release the reference to it when you are finished with the endpoint. + * + * \retval NULL Could not get endpoint + * \retval non-NULL The endpoint + */ +struct ast_sip_endpoint *ast_sip_subscription_get_endpoint(struct ast_sip_subscription *sub); + +/*! + * \brief Get the serializer for the subscription + * + * Tasks that originate outside of a SIP servant thread should get the serializer + * and push the task to the serializer. + * + * \param sub The subscription + * \retval NULL Failure + * \retval non-NULL The subscription's serializer + */ +struct ast_taskprocessor *ast_sip_subscription_get_serializer(struct ast_sip_subscription *sub); + +/*! + * \brief Get the underlying PJSIP evsub structure + * + * This is useful when wishing to call PJSIP's API calls in order to + * create SUBSCRIBEs, NOTIFIES, etc. as well as get subscription state + * + * This function, as well as all methods called on the pjsip_evsub should + * be done in a SIP servant thread. + * + * \param sub The subscription + * \retval NULL Failure + * \retval non-NULL The underlying pjsip_evsub + */ +pjsip_evsub *ast_sip_subscription_get_evsub(struct ast_sip_subscription *sub); + +/*! + * \brief Send a request created via a PJSIP evsub method + * + * Callers of this function should take care to do so within a SIP servant + * thread. + * + * \param sub The subscription on which to send the request + * \param tdata The request to send + * \retval 0 Success + * \retval non-zero Failure + */ +int ast_sip_subscription_send_request(struct ast_sip_subscription *sub, pjsip_tx_data *tdata); + +/*! + * \brief Alternative for ast_datastore_alloc() + * + * There are two major differences between this and ast_datastore_alloc() + * 1) This allocates a refcounted object + * 2) This will fill in a uid if one is not provided + * + * DO NOT call ast_datastore_free() on a datastore allocated in this + * way since that function will attempt to free the datastore rather + * than play nicely with its refcount. + * + * \param info Callbacks for datastore + * \param uid Identifier for datastore + * \retval NULL Failed to allocate datastore + * \retval non-NULL Newly allocated datastore + */ +struct ast_datastore *ast_sip_subscription_alloc_datastore(const struct ast_datastore_info *info, const char *uid); + +/*! + * \brief Add a datastore to a SIP subscription + * + * Note that SIP uses reference counted datastores. The datastore passed into this function + * must have been allocated using ao2_alloc() or there will be serious problems. + * + * \param subscription The ssubscription to add the datastore to + * \param datastore The datastore to be added to the subscription + * \retval 0 Success + * \retval -1 Failure + */ +int ast_sip_subscription_add_datastore(struct ast_sip_subscription *subscription, struct ast_datastore *datastore); + +/*! + * \brief Retrieve a subscription datastore + * + * The datastore retrieved will have its reference count incremented. When the caller is done + * with the datastore, the reference counted needs to be decremented using ao2_ref(). + * + * \param subscription The subscription from which to retrieve the datastore + * \param name The name of the datastore to retrieve + * \retval NULL Failed to find the specified datastore + * \retval non-NULL The specified datastore + */ +struct ast_datastore *ast_sip_subscription_get_datastore(struct ast_sip_subscription *subscription, const char *name); + +/*! + * \brief Remove a subscription datastore from the subscription + * + * This operation may cause the datastore's free() callback to be called if the reference + * count reaches zero. + * + * \param subscription The subscription to remove the datastore from + * \param name The name of the datastore to remove + */ +void ast_sip_subscription_remove_datastore(struct ast_sip_subscription *subscription, const char *name); + +/*! + * \brief Register a subscription handler + * + * \retval 0 Handler was registered successfully + * \retval non-zero Handler was not registered successfully + */ +int ast_sip_register_subscription_handler(struct ast_sip_subscription_handler *handler); + +/*! + * \brief Unregister a subscription handler + */ +void ast_sip_unregister_subscription_handler(struct ast_sip_subscription_handler *handler); + +#endif /* RES_SIP_PUBSUB_H */ diff --git a/include/asterisk/res_sip_session.h b/include/asterisk/res_sip_session.h new file mode 100644 index 000000000..cbed52621 --- /dev/null +++ b/include/asterisk/res_sip_session.h @@ -0,0 +1,468 @@ +/* + * 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_SESSION_H +#define _RES_SIP_SESSION_H + +/* Needed for pj_timer_entry definition */ +#include "pjlib.h" +#include "asterisk/linkedlists.h" +/* Needed for AST_MAX_EXTENSION constant */ +#include "asterisk/channel.h" +/* Needed for ast_sockaddr struct */ +#include "asterisk/netsock.h" + +/* Forward declarations */ +struct ast_sip_endpoint; +struct ast_sip_transport; +struct pjsip_inv_session; +struct ast_channel; +struct ast_datastore; +struct ast_datastore_info; +struct ao2_container; +struct pjsip_tx_data; +struct pjsip_rx_data; +struct ast_party_id; +struct pjmedia_sdp_media; +struct pjmedia_sdp_session; +struct ast_rtp_instance; + +struct ast_sip_session_sdp_handler; + +/*! + * \brief A structure containing SIP session media information + */ +struct ast_sip_session_media { + /*! \brief RTP instance itself */ + struct ast_rtp_instance *rtp; + /*! \brief Direct media address */ + struct ast_sockaddr direct_media_addr; + /*! \brief SDP handler that setup the RTP */ + struct ast_sip_session_sdp_handler *handler; + /*! \brief Stream is on hold */ + unsigned int held:1; + /*! \brief Stream type this session media handles */ + char stream_type[1]; +}; + +/*! + * \brief Opaque structure representing a request that could not be sent + * due to an outstanding INVITE transaction + */ +struct ast_sip_session_delayed_request; + +/*! + * \brief A structure describing a SIP session + * + * For the sake of brevity, a "SIP session" in Asterisk is referring to + * a dialog initiated by an INVITE. While "session" is typically interpreted + * to refer to the negotiated media within a SIP dialog, we have opted + * to use the term "SIP session" to refer to the INVITE dialog itself. + */ +struct ast_sip_session { + /* Dialplan extension where incoming call is destined */ + char exten[AST_MAX_EXTENSION]; + /* The endpoint with which Asterisk is communicating */ + struct ast_sip_endpoint *endpoint; + /* The PJSIP details of the session, which includes the dialog */ + struct pjsip_inv_session *inv_session; + /* The Asterisk channel associated with the session */ + struct ast_channel *channel; + /* Registered session supplements */ + AST_LIST_HEAD(, ast_sip_session_supplement) supplements; + /* Datastores added to the session by supplements to the session */ + struct ao2_container *datastores; + /* Media streams */ + struct ao2_container *media; + /* Serializer for tasks relating to this SIP session */ + struct ast_taskprocessor *serializer; + /* Requests that could not be sent due to current inv_session state */ + AST_LIST_HEAD_NOLOCK(, ast_sip_session_delayed_request) delayed_requests; + /* When we need to reschedule a reinvite, we use this structure to do it */ + pj_timer_entry rescheduled_reinvite; + /* Format capabilities pertaining to direct media */ + struct ast_format_cap *direct_media_cap; + /* Identity of endpoint this session deals with */ + struct ast_party_id id; + /* Requested capabilities */ + struct ast_format_cap *req_caps; +}; + +typedef int (*ast_sip_session_request_creation_cb)(struct ast_sip_session *session, pjsip_tx_data *tdata); +typedef int (*ast_sip_session_response_cb)(struct ast_sip_session *session, pjsip_rx_data *rdata); + +enum ast_sip_session_supplement_priority { + /*! Top priority. Supplements with this priority are those that need to run before any others */ + AST_SIP_SESSION_SUPPLEMENT_PRIORITY_FIRST = 0, + /*! Channel creation priority. + * chan_gulp 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_SESSION_SUPPLEMENT_PRIORITY_CHANNEL = 1000000, + /*! Lowest priority. Supplements with this priority should be run after all other supplements */ + AST_SIP_SESSION_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 requests and responses + */ +struct ast_sip_session_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_session_supplement_priority priority; + /*! + * \brief Notification that the session has begun + * This method will always be called from a SIP servant thread. + */ + void (*session_begin)(struct ast_sip_session *session); + /*! + * \brief Notification that the session has ended + * + * This method may or may not be called from a SIP servant thread. Do + * not make assumptions about being able to call PJSIP methods from within + * this method. + */ + void (*session_end)(struct ast_sip_session *session); + /*! + * \brief Notification that the session is being destroyed + */ + void (*session_destroy)(struct ast_sip_session *session); + /*! + * \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. + */ + int (*incoming_request)(struct ast_sip_session *session, 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. + */ + void (*incoming_response)(struct ast_sip_session *session, 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_session *session, 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_session *session, struct pjsip_tx_data *tdata); + /*! Next item in the list */ + AST_LIST_ENTRY(ast_sip_session_supplement) next; +}; + +/*! + * \brief A handler for SDPs in SIP sessions + * + * An SDP handler is registered by a module that is interested in being the + * responsible party for specific types of SDP streams. + */ +struct ast_sip_session_sdp_handler { + /*! An identifier for this handler */ + const char *id; + /*! + * \brief Set session details based on a stream in an incoming SDP offer or answer + * \param session The session for which the media is being negotiated + * \param session_media The media to be setup for this session + * \param sdp The entire SDP. Useful for getting "global" information, such as connections or attributes + * \param stream The stream on which to operate + * \retval 0 The stream was not handled by this handler. If there are other registered handlers for this stream type, they will be called. + * \retval <0 There was an error encountered. No further operation will take place and the current negotiation will be abandoned. + * \retval >0 The stream was handled by this handler. No further handler of this stream type will be called. + */ + int (*negotiate_incoming_sdp_stream)(struct ast_sip_session *session, struct ast_sip_session_media *session_media, const struct pjmedia_sdp_session *sdp, const struct pjmedia_sdp_media *stream); + /*! + * \brief Create an SDP media stream and add it to the outgoing SDP offer or answer + * \param session The session for which media is being added + * \param session_media The media to be setup for this session + * \param stream The stream on which to operate + * \retval 0 The stream was not handled by this handler. If there are other registered handlers for this stream type, they will be called. + * \retval <0 There was an error encountered. No further operation will take place and the current negotiation will be abandoned. + * \retval >0 The stream was handled by this handler. No further handler of this stream type will be called. + */ + int (*handle_incoming_sdp_stream)(struct ast_sip_session *session, struct ast_sip_session_media *session_media, const struct pjmedia_sdp_session *sdp, struct pjmedia_sdp_media *stream); + /*! + * \brief Create an SDP media stream and add it to the outgoing SDP offer or answer + * \param session The session for which media is being added + * \param session_media The media to be setup for this session + * \param sdp The entire SDP as currently built + * \retval 0 This handler has no stream to add. If there are other registered handlers for this stream type, they will be called. + * \retval <0 There was an error encountered. No further operation will take place and the current SDP negotiation will be abandoned. + * \retval >0 The handler has a stream to be added to the SDP. No further handler of this stream type will be called. + */ + int (*create_outgoing_sdp_stream)(struct ast_sip_session *session, struct ast_sip_session_media *session_media, struct pjmedia_sdp_session *sdp); + /*! + * \brief Update media stream with external address if applicable + * \param tdata The outgoing message itself + * \param stream The stream on which to operate + * \param transport The transport the SDP is going out on + */ + void (*change_outgoing_sdp_stream_media_address)(struct pjsip_tx_data *tdata, struct pjmedia_sdp_media *stream, struct ast_sip_transport *transport); + /*! + * \brief Apply a negotiated SDP media stream + * \param session The session for which media is being applied + * \param session_media The media to be setup for this session + * \param local The entire local negotiated SDP + * \param local_stream The local stream which to apply + * \param remote The entire remote negotiated SDP + * \param remote_stream The remote stream which to apply + * \retval 0 The stream was not applied by this handler. If there are other registered handlers for this stream type, they will be called. + * \retval <0 There was an error encountered. No further operation will take place and the current application will be abandoned. + * \retval >0 The stream was handled by this handler. No further handler of this stream type will be called. + */ + int (*apply_negotiated_sdp_stream)(struct ast_sip_session *session, struct ast_sip_session_media *session_media, const struct pjmedia_sdp_session *local, const struct pjmedia_sdp_media *local_stream, + const struct pjmedia_sdp_session *remote, const struct pjmedia_sdp_media *remote_stream); + /*! + * \brief Destroy a session_media created by this handler + * \param session The session for which media is being destroyed + * \param session_media The media to destroy + */ + void (*stream_destroy)(struct ast_sip_session_media *session_media); + /*! Next item in the list. */ + AST_LIST_ENTRY(ast_sip_session_sdp_handler) next; +}; + +/*! + * \brief Allocate a new SIP session + * + * This will take care of allocating the datastores container on the session as well + * as placing all registered supplements onto the session. + * + * The endpoint that is passed in will have its reference count increased by one since + * the session will be keeping a reference to the endpoint. The session will relinquish + * this reference when the session is destroyed. + * + * \param endpoint The endpoint that this session communicates with + * \param inv_session The PJSIP INVITE session data + */ +struct ast_sip_session *ast_sip_session_alloc(struct ast_sip_endpoint *endpoint, pjsip_inv_session *inv); + +/*! + * \brief Create a new outgoing SIP session + * + * The endpoint that is passed in will have its reference count increased by one since + * the session will be keeping a reference to the endpoint. The session will relinquish + * this reference when the session is destroyed. + * + * \param endpoint The endpoint that this session uses for settings + * \param location Optional name of the location to call, be it named location or explicit URI + * \param request_user Optional request user to place in the request URI if permitted + * \param req_caps The requested capabilities + */ +struct ast_sip_session *ast_sip_session_create_outgoing(struct ast_sip_endpoint *endpoint, const char *location, const char *request_user, struct ast_format_cap *req_caps); + +/*! + * \brief Register an SDP handler + * + * An SDP handler is responsible for parsing incoming SDP streams and ensuring that + * Asterisk can cope with the contents. Similarly, the SDP handler will be + * responsible for constructing outgoing SDP streams. + * + * Multiple handlers for the same stream type may be registered. They will be + * visited in the order they were registered. Handlers will be visited for each + * stream type until one claims to have handled the stream. + * + * \param handler The SDP handler to register + * \param stream_type The type of media stream for which to call the handler + * \retval 0 Success + * \retval -1 Failure + */ +int ast_sip_session_register_sdp_handler(struct ast_sip_session_sdp_handler *handler, const char *stream_type); + +/*! + * \brief Unregister an SDP handler + * + * \param handler The SDP handler to unregister + * \param stream_type Stream type for which the SDP handler was registered + */ +void ast_sip_session_unregister_sdp_handler(struct ast_sip_session_sdp_handler *handler, const char *stream_type); + +/*! + * \brief Register a supplement to SIP session processing + * + * This allows for someone to insert themselves in the processing of SIP + * requests and responses. This, for example could allow for a module to + * set channel data based on headers in an incoming message. Similarly, + * a module could reject an incoming request if desired. + * + * \param supplement The supplement to register + * \retval 0 Success + * \retval -1 Failure + */ +int ast_sip_session_register_supplement(struct ast_sip_session_supplement *supplement); + +/*! + * \brief Unregister a an supplement to SIP session processing + * + * \param supplement The supplement to unregister + */ +void ast_sip_session_unregister_supplement(struct ast_sip_session_supplement *supplement); + +/*! + * \brief Alternative for ast_datastore_alloc() + * + * There are two major differences between this and ast_datastore_alloc() + * 1) This allocates a refcounted object + * 2) This will fill in a uid if one is not provided + * + * DO NOT call ast_datastore_free() on a datastore allocated in this + * way since that function will attempt to free the datastore rather + * than play nicely with its refcount. + * + * \param info Callbacks for datastore + * \param uid Identifier for datastore + * \retval NULL Failed to allocate datastore + * \retval non-NULL Newly allocated datastore + */ +struct ast_datastore *ast_sip_session_alloc_datastore(const struct ast_datastore_info *info, const char *uid); + +/*! + * \brief Add a datastore to a SIP session + * + * Note that SIP uses reference counted datastores. The datastore passed into this function + * must have been allocated using ao2_alloc() or there will be serious problems. + * + * \param session The session to add the datastore to + * \param datastore The datastore to be added to the session + * \retval 0 Success + * \retval -1 Failure + */ +int ast_sip_session_add_datastore(struct ast_sip_session *session, struct ast_datastore *datastore); + +/*! + * \brief Retrieve a session datastore + * + * The datastore retrieved will have its reference count incremented. When the caller is done + * with the datastore, the reference counted needs to be decremented using ao2_ref(). + * + * \param session The session from which to retrieve the datastore + * \param name The name of the datastore to retrieve + * \retval NULL Failed to find the specified datastore + * \retval non-NULL The specified datastore + */ +struct ast_datastore *ast_sip_session_get_datastore(struct ast_sip_session *session, const char *name); + +/*! + * \brief Remove a session datastore from the session + * + * This operation may cause the datastore's free() callback to be called if the reference + * count reaches zero. + * + * \param session The session to remove the datastore from + * \param name The name of the datastore to remove + */ +void ast_sip_session_remove_datastore(struct ast_sip_session *session, const char *name); + +/*! + * \brief Retrieve identifying information from an incoming request + * + * This will retrieve identifying information and place it in the + * id parameter. The caller of the function can then apply this to + * caller ID, connected line, or whatever else may be proper. + * + * \param rdata The incoming request or response + * \param[out] id The collected identity information + * \retval 0 Successfully found identifying information + * \retval -1 Identifying information could not be found + */ +int ast_sip_session_get_identity(struct pjsip_rx_data *rdata, struct ast_party_id *id); + +/*! + * \brief Send a reinvite or UPDATE on a session + * + * This method will inspect the session in order to construct an appropriate + * session refresh request. As with any outgoing request in res_sip_session, + * this will call into registered supplements in case they wish to add anything. + * + * Note: The on_request_creation callback may or may not be called in the same + * thread where this function is called. Request creation may need to be delayed + * due to the current INVITE transaction state. + * + * \param session The session on which the reinvite will be sent + * \param on_request_creation Callback called when request is created + * \param on_response Callback called when response for request is received + * \param method The method that should be used when constructing the session refresh + * \param generate_new_sdp Boolean to indicate if a new SDP should be created + * \retval 0 Successfully sent refresh + * \retval -1 Failure to send refresh + */ +int ast_sip_session_refresh(struct ast_sip_session *session, + ast_sip_session_request_creation_cb on_request_creation, + ast_sip_session_response_cb on_response, + enum ast_sip_session_refresh_method method, + int generate_new_sdp); + +/*! + * \brief Send a SIP response + * + * This will send the SIP response specified in tdata and + * call into any registered supplements' outgoing_response callback. + * + * \param session The session on which to send the response. + * \param tdata The response to send + */ +void ast_sip_session_send_response(struct ast_sip_session *session, pjsip_tx_data *tdata); + +/*! + * \brief Send a SIP request + * + * This will send the SIP request specified in tdata and + * call into any registered supplements' outgoing_request callback. + * + * \param session The session to which to send the request + * \param tdata The request to send + */ +void ast_sip_session_send_request(struct ast_sip_session *session, pjsip_tx_data *tdata); + +/*! + * \brief Send a SIP request and get called back when a response is received + * + * This will send the request out exactly the same as ast_sip_send_request() does. + * The difference is that when a response arrives, the specified callback will be + * called into + * + * \param session The session on which to send the request + * \param tdata The request to send + * \param on_response Callback to be called when a response is received + */ +void ast_sip_session_send_request_with_cb(struct ast_sip_session *session, pjsip_tx_data *tdata, + ast_sip_session_response_cb on_response); + +#endif /* _RES_SIP_SESSION_H */ diff --git a/include/asterisk/sorcery.h b/include/asterisk/sorcery.h index e390b43cf..434f5595a 100644 --- a/include/asterisk/sorcery.h +++ b/include/asterisk/sorcery.h @@ -157,10 +157,15 @@ typedef struct ast_variable *(*sorcery_transform_handler)(struct ast_variable *s /*! * \brief A callback function for when an object set is successfully applied to an object * + * \note On a failure return, the state of the object is left undefined. It is a bad + * idea to try to use this object. + * * \param sorcery Sorcery structure in use * \param obj The object itself + * \retval 0 Success + * \retval non-zero Failure */ -typedef void (*sorcery_apply_handler)(const struct ast_sorcery *sorcery, void *obj); +typedef int (*sorcery_apply_handler)(const struct ast_sorcery *sorcery, void *obj); /*! * \brief A callback function for copying the contents of one object to another diff --git a/include/asterisk/threadpool.h b/include/asterisk/threadpool.h index 89076265e..e1e7727f5 100644 --- a/include/asterisk/threadpool.h +++ b/include/asterisk/threadpool.h @@ -108,6 +108,20 @@ struct ast_threadpool_options { * maximum size. */ int max_size; + /*! + * \brief Function to call when a thread starts + * + * This is useful if there is something common that all threads + * in a threadpool need to do when they start. + */ + void (*thread_start)(void); + /*! + * \brief Function to call when a thread ends + * + * This is useful if there is common cleanup to execute when + * a thread completes + */ + void (*thread_end)(void); }; /*! |