diff options
Diffstat (limited to 'include/asterisk')
| -rw-r--r-- | include/asterisk/res_sip.h | 178 | ||||
| -rw-r--r-- | include/asterisk/res_sip_exten_state.h | 94 | ||||
| -rw-r--r-- | include/asterisk/res_sip_pubsub.h | 17 | ||||
| -rw-r--r-- | include/asterisk/res_sip_session.h | 42 | ||||
| -rw-r--r-- | include/asterisk/sdp_srtp.h | 125 |
5 files changed, 452 insertions, 4 deletions
diff --git a/include/asterisk/res_sip.h b/include/asterisk/res_sip.h index b48ed9f82..7c486aa68 100644 --- a/include/asterisk/res_sip.h +++ b/include/asterisk/res_sip.h @@ -139,6 +139,47 @@ struct ast_sip_contact { ); /*! Absolute time that this contact is no longer valid after */ struct timeval expiration_time; + /*!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; }; /*! @@ -157,6 +198,10 @@ struct ast_sip_aor { 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 */ @@ -245,6 +290,17 @@ enum ast_sip_direct_media_glare_mitigation { 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 An entity with which Asterisk communicates */ @@ -306,14 +362,14 @@ struct ast_sip_endpoint { 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; + /*! When performing connected line update, which method should be used */ + enum ast_sip_session_refresh_method connected_line_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 */ @@ -326,8 +382,26 @@ struct ast_sip_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; /*! Should unsolicited MWI be aggregated into a single NOTIFY? */ unsigned int aggregate_mwi; + /*! Do we use media encryption? what type? */ + enum ast_sip_session_media_encryption media_encryption; + /*! Do we use AVPF exclusively for this endpoint? */ + unsigned int use_avpf; + /*! Is one-touch recording permitted? */ + unsigned int one_touch_recording; + /*! Boolean indicating if ringing should be sent as inband progress */ + unsigned int inband_progress; + /*! 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; /*! Pointer to the persistent Asterisk endpoint */ struct ast_endpoint *persistent; /*! The number of channels at which busy device state is returned */ @@ -553,6 +627,16 @@ struct ast_sorcery *ast_sip_get_sorcery(void); 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 @@ -611,6 +695,37 @@ struct ast_sip_contact *ast_sip_location_retrieve_contact_from_aor_list(const ch 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 @@ -1045,7 +1160,7 @@ int ast_sip_append_body(pjsip_tx_data *tdata, const char *body_text); * \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); +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 @@ -1085,4 +1200,61 @@ int ast_sip_retrieve_auths(const char *auth_names[], size_t num_auths, struct as */ 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); + #endif /* _RES_SIP_H */ diff --git a/include/asterisk/res_sip_exten_state.h b/include/asterisk/res_sip_exten_state.h new file mode 100644 index 000000000..62662f930 --- /dev/null +++ b/include/asterisk/res_sip_exten_state.h @@ -0,0 +1,94 @@ +/* + * Asterisk -- An open source telephony toolkit. + * + * Copyright (C) 2013, Digium, Inc. + * + * Kevin Harwell <kharwell@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_EXTEN_STATE_H +#define _RES_SIP_EXTEN_STATE_H + +#include "asterisk/stringfields.h" +#include "asterisk/linkedlists.h" + +#include "asterisk/pbx.h" +#include "asterisk/presencestate.h" + + +/*! + * \brief Contains information pertaining to extension/device state changes. + */ +struct ast_sip_exten_state_data { + /*! The extension of the current state change */ + const char *exten; + /*! The extension state of the change */ + enum ast_extension_states exten_state; + /*! The presence state of the change */ + enum ast_presence_state presence_state; + /*! Current device state information */ + struct ao2_container *device_state_info; +}; + +/*! + * \brief Extension state provider. + */ +struct ast_sip_exten_state_provider { + /*! The name of the event this provider registers for */ + const char *event_name; + /*! Type of the body, ex: "application" */ + const char *type; + /*! Subtype of the body, ex: "pidf+xml" */ + const char *subtype; + /*! Type/Subtype together - ex: application/pidf+xml */ + const char *body_type; + /*! Subscription handler to be used and associated with provider */ + struct ast_sip_subscription_handler *handler; + + /*! + * \brief Create the body text of a NOTIFY request. + * + * Implementors use this to create body information within the given + * ast_str. That information is then added to the NOTIFY request. + * + * \param data Current extension state changes + * \param local URI of the dialog's local party, e.g. 'from' + * \param remote URI of the dialog's remote party, e.g. 'to' + * \param body_text Out parameter used to populate the NOTIFY msg body + * \retval 0 Successfully created the body's text + * \retval -1 Failed to create the body's text + */ + int (*create_body)(struct ast_sip_exten_state_data *data, const char *local, + const char *remote, struct ast_str **body_text); + + /*! Next item in the list */ + AST_LIST_ENTRY(ast_sip_exten_state_provider) next; +}; + +/*! + * \brief Registers an extension state provider. + * + * \param obj An extension state provider + * \retval 0 Successfully registered the extension state provider + * \retval -1 Failed to register the extension state provider + */ +int ast_sip_register_exten_state_provider(struct ast_sip_exten_state_provider *obj); + +/*! + * \brief Unregisters an extension state provider. + * + * \param obj An extension state provider + */ +void ast_sip_unregister_exten_state_provider(struct ast_sip_exten_state_provider *obj); + +#endif diff --git a/include/asterisk/res_sip_pubsub.h b/include/asterisk/res_sip_pubsub.h index 33614b285..be443299c 100644 --- a/include/asterisk/res_sip_pubsub.h +++ b/include/asterisk/res_sip_pubsub.h @@ -261,7 +261,22 @@ struct ast_taskprocessor *ast_sip_subscription_get_serializer(struct ast_sip_sub * \retval non-NULL The underlying pjsip_evsub */ pjsip_evsub *ast_sip_subscription_get_evsub(struct ast_sip_subscription *sub); - + +/*! + * \brief Get the underlying PJSIP dialog structure + * + * Call this function when information needs to be retrieved from the + * underlying pjsip dialog. + * + * 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_dialog + */ +pjsip_dialog *ast_sip_subscription_get_dlg(struct ast_sip_subscription *sub); + /*! * \brief Send a request created via a PJSIP evsub method * diff --git a/include/asterisk/res_sip_session.h b/include/asterisk/res_sip_session.h index cbed52621..e4b05f7c3 100644 --- a/include/asterisk/res_sip_session.h +++ b/include/asterisk/res_sip_session.h @@ -26,6 +26,8 @@ #include "asterisk/channel.h" /* Needed for ast_sockaddr struct */ #include "asterisk/netsock.h" +/* Neeed for ast_sdp_srtp struct */ +#include "asterisk/sdp_srtp.h" /* Forward declarations */ struct ast_sip_endpoint; @@ -41,6 +43,7 @@ struct ast_party_id; struct pjmedia_sdp_media; struct pjmedia_sdp_session; struct ast_rtp_instance; +struct ast_dsp; struct ast_sip_session_sdp_handler; @@ -54,6 +57,8 @@ struct ast_sip_session_media { struct ast_sockaddr direct_media_addr; /*! \brief SDP handler that setup the RTP */ struct ast_sip_session_sdp_handler *handler; + /*! \brief Holds SRTP information */ + struct ast_sdp_srtp *srtp; /*! \brief Stream is on hold */ unsigned int held:1; /*! \brief Stream type this session media handles */ @@ -97,10 +102,18 @@ struct ast_sip_session { pj_timer_entry rescheduled_reinvite; /* Format capabilities pertaining to direct media */ struct ast_format_cap *direct_media_cap; + /* When we need to forcefully end the session */ + pj_timer_entry scheduled_termination; /* Identity of endpoint this session deals with */ struct ast_party_id id; /* Requested capabilities */ struct ast_format_cap *req_caps; + /* Codecs overriden by dialplan on an outgoing request */ + struct ast_codec_pref override_prefs; + /* Optional DSP, used only for inband DTMF detection if configured */ + struct ast_dsp *dsp; + /* Whether the termination of the session should be deferred */ + unsigned int defer_terminate:1; }; typedef int (*ast_sip_session_request_creation_cb)(struct ast_sip_session *session, pjsip_tx_data *tdata); @@ -289,6 +302,13 @@ struct ast_sip_session *ast_sip_session_alloc(struct ast_sip_endpoint *endpoint, 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 Defer local termination of a session until remote side terminates, or an amount of time passes + * + * \param session The session to defer termination on + */ +void ast_sip_session_defer_termination(struct ast_sip_session *session); + +/*! * \brief Register an SDP handler * * An SDP handler is responsible for parsing incoming SDP streams and ensuring that @@ -452,6 +472,14 @@ void ast_sip_session_send_response(struct ast_sip_session *session, pjsip_tx_dat void ast_sip_session_send_request(struct ast_sip_session *session, pjsip_tx_data *tdata); /*! + * \brief Creates an INVITE request. + * + * \param session Starting session for the INVITE + * \param tdata The created request. + */ +int ast_sip_session_create_invite(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. @@ -465,4 +493,18 @@ void ast_sip_session_send_request(struct ast_sip_session *session, pjsip_tx_data void ast_sip_session_send_request_with_cb(struct ast_sip_session *session, pjsip_tx_data *tdata, ast_sip_session_response_cb on_response); +/*! + * \brief Retrieves a session from a dialog + * + * \param dlg The dialog to retrieve the session from + * + * \retval non-NULL if session exists + * \retval NULL if no session + * + * \note The reference count of the session is increased when returned + * + * \note This function *must* be called with the dialog locked + */ +struct ast_sip_session *ast_sip_dialog_get_session(pjsip_dialog *dlg); + #endif /* _RES_SIP_SESSION_H */ diff --git a/include/asterisk/sdp_srtp.h b/include/asterisk/sdp_srtp.h new file mode 100644 index 000000000..9b92e0e3f --- /dev/null +++ b/include/asterisk/sdp_srtp.h @@ -0,0 +1,125 @@ +/* + * Asterisk -- An open source telephony toolkit. + * + * Copyright (C) 2006 - 2007, Mikael Magnusson + * + * Mikael Magnusson <mikma@users.sourceforge.net> + * + * 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. + */ + +/*! \file sdp_srtp.h + * + * \brief SRTP and SDP Security descriptions + * + * Specified in RFC 4568 + * Specified in RFC 3711 + * + * \author Mikael Magnusson <mikma@users.sourceforge.net> + */ + +#ifndef _SDP_SRTP_H +#define _SDP_SRTP_H + +#include <asterisk/rtp_engine.h> + +struct ast_sdp_crypto; + +/*! \brief structure for secure RTP audio */ +struct ast_sdp_srtp { + unsigned int flags; + struct ast_sdp_crypto *crypto; +}; + +/* SRTP flags */ +#define AST_SRTP_CRYPTO_OFFER_OK (1 << 1) +#define AST_SRTP_CRYPTO_TAG_32 (1 << 2) +#define AST_SRTP_CRYPTO_TAG_80 (1 << 3) + +/*! + * \brief allocate a ast_sdp_srtp structure + * \retval a new malloc'd ast_sdp_srtp structure on success + * \retval NULL on failure +*/ +struct ast_sdp_srtp *ast_sdp_srtp_alloc(void); + +/*! + * \brief free a ast_sdp_srtp structure + * \param srtp a ast_sdp_srtp structure +*/ +void ast_sdp_srtp_destroy(struct ast_sdp_srtp *srtp); + +/*! \brief Initialize an return an ast_sdp_crypto struct + * + * \details + * This function allocates a new ast_sdp_crypto struct and initializes its values + * + * \retval NULL on failure + * \retval a pointer to a new ast_sdp_crypto structure + */ +struct ast_sdp_crypto *ast_sdp_crypto_alloc(void); + +/*! \brief Destroy a previously allocated ast_sdp_crypto struct */ +void ast_sdp_crypto_destroy(struct ast_sdp_crypto *crypto); + +/*! \brief Parse the a=crypto line from SDP and set appropriate values on the + * ast_sdp_crypto struct. + * + * The attribute line should already have "a=crypto:" removed. + * + * \param p A valid ast_sdp_crypto struct + * \param attr the a:crypto line from SDP + * \param rtp The rtp instance associated with the SDP being parsed + * \param srtp SRTP structure + * + * \retval 0 success + * \retval nonzero failure + */ +int ast_sdp_crypto_process(struct ast_rtp_instance *rtp, struct ast_sdp_srtp *srtp, const char *attr); + +/*! \brief Generate an SRTP a=crypto offer + * + * \details + * The offer is stored on the ast_sdp_crypto struct in a_crypto + * + * \param p A valid ast_sdp_crypto struct + * \param taglen Length + * + * \retval 0 success + * \retval nonzero failure + */ +int ast_sdp_crypto_build_offer(struct ast_sdp_crypto *p, int taglen); + + +/*! \brief Get the crypto attribute line for the srtp structure + * + * The attribute line does not contain the initial "a=crypto:" and does + * not terminate with "\r\n". + * + * \param srtp The ast_sdp_srtp structure for which to get an attribute line + * \param dtls_enabled Whether this connection is encrypted with datagram TLS + * \param default_taglen_32 Whether to default to a tag length of 32 instead of 80 + * + * \retval An attribute line containing cryptographic information + * \retval NULL if the srtp structure does not require an attribute line containing crypto information + */ +const char *ast_sdp_srtp_get_attrib(struct ast_sdp_srtp *srtp, int dtls_enabled, int default_taglen_32); + +/*! \brief Get the RTP profile in use by a media session + * + * \param sdes_active Whether the media session is using SDES-SRTP + * \param instance The RTP instance associated with this media session + * \param using_avpf Whether the media session is using early feedback (AVPF) + * + * \retval A non-allocated string describing the profile in use (does not need to be freed) + */ +char *ast_sdp_get_rtp_profile(unsigned int sdes_active, struct ast_rtp_instance *instance, unsigned int using_avpf); +#endif /* _SDP_CRYPTO_H */ |