2 * Asterisk -- An open source telephony toolkit.
4 * Copyright (C) 2013, Digium, Inc.
6 * Mark Michelson <mmichelson@digium.com>
8 * See http://www.asterisk.org for more information about
9 * the Asterisk project. Please do not directly contact
10 * any of the maintainers of this project for assistance;
11 * the project provides a web site, mailing lists and IRC
12 * channels for your use.
14 * This program is free software, distributed under the terms of
15 * the GNU General Public License Version 2. See the LICENSE file
16 * at the top of the source tree.
22 #include "asterisk/stringfields.h"
23 /* Needed for struct ast_sockaddr */
24 #include "asterisk/netsock2.h"
25 /* Needed for linked list macros */
26 #include "asterisk/linkedlists.h"
27 /* Needed for ast_party_id */
28 #include "asterisk/channel.h"
29 /* Needed for ast_sorcery */
30 #include "asterisk/sorcery.h"
31 /* Needed for ast_dnsmgr */
32 #include "asterisk/dnsmgr.h"
33 /* Needed for ast_endpoint */
34 #include "asterisk/endpoints.h"
35 /* Needed for ast_t38_ec_modes */
36 #include "asterisk/udptl.h"
37 /* Needed for pj_sockaddr */
39 /* Needed for ast_rtp_dtls_cfg struct */
40 #include "asterisk/rtp_engine.h"
42 /* Forward declarations of PJSIP stuff */
47 struct pjsip_transport;
48 struct pjsip_tpfactory;
49 struct pjsip_tls_setting;
50 struct pjsip_tpselector;
53 * \brief Structure for SIP transport information
55 struct ast_sip_transport_state {
56 /*! \brief Transport itself */
57 struct pjsip_transport *transport;
59 /*! \brief Transport factory */
60 struct pjsip_tpfactory *factory;
63 #define SIP_SORCERY_DOMAIN_ALIAS_TYPE "domain_alias"
66 * Details about a SIP domain alias
68 struct ast_sip_domain_alias {
69 /*! Sorcery object details */
70 SORCERY_OBJECT(details);
71 AST_DECLARE_STRING_FIELDS(
72 /*! Domain to be aliased to */
73 AST_STRING_FIELD(domain);
77 /*! \brief Maximum number of ciphers supported for a TLS transport */
78 #define SIP_TLS_MAX_CIPHERS 64
81 * \brief Transport to bind to
83 struct ast_sip_transport {
84 /*! Sorcery object details */
85 SORCERY_OBJECT(details);
86 AST_DECLARE_STRING_FIELDS(
87 /*! Certificate of authority list file */
88 AST_STRING_FIELD(ca_list_file);
89 /*! Public certificate file */
90 AST_STRING_FIELD(cert_file);
91 /*! Optional private key of the certificate file */
92 AST_STRING_FIELD(privkey_file);
93 /*! Password to open the private key */
94 AST_STRING_FIELD(password);
95 /*! External signaling address */
96 AST_STRING_FIELD(external_signaling_address);
97 /*! External media address */
98 AST_STRING_FIELD(external_media_address);
99 /*! Optional domain to use for messages if provided could not be found */
100 AST_STRING_FIELD(domain);
102 /*! Type of transport */
103 enum ast_transport type;
104 /*! Address and port to bind to */
106 /*! Number of simultaneous asynchronous operations */
107 unsigned int async_operations;
108 /*! Optional external port for signaling */
109 unsigned int external_signaling_port;
111 pjsip_tls_setting tls;
112 /*! Configured TLS ciphers */
113 pj_ssl_cipher ciphers[SIP_TLS_MAX_CIPHERS];
114 /*! Optional local network information, used for NAT purposes */
115 struct ast_ha *localnet;
116 /*! DNS manager for refreshing the external address */
117 struct ast_dnsmgr_entry *external_address_refresher;
118 /*! Optional external address information */
119 struct ast_sockaddr external_address;
120 /*! Transport state information */
121 struct ast_sip_transport_state *state;
122 /*! QOS DSCP TOS bits */
129 * \brief Structure for SIP nat hook information
131 struct ast_sip_nat_hook {
132 /*! Sorcery object details */
133 SORCERY_OBJECT(details);
134 /*! Callback for when a message is going outside of our local network */
135 void (*outgoing_external_message)(struct pjsip_tx_data *tdata, struct ast_sip_transport *transport);
139 * \brief Contact associated with an address of record
141 struct ast_sip_contact {
142 /*! Sorcery object details, the id is the aor name plus a random string */
143 SORCERY_OBJECT(details);
144 AST_DECLARE_STRING_FIELDS(
145 /*! Full URI of the contact */
146 AST_STRING_FIELD(uri);
148 /*! Absolute time that this contact is no longer valid after */
149 struct timeval expiration_time;
150 /*! Frequency to send OPTIONS requests to contact. 0 is disabled. */
151 unsigned int qualify_frequency;
152 /*! If true authenticate the qualify if needed */
153 int authenticate_qualify;
156 #define CONTACT_STATUS "contact_status"
159 * \brief Status type for a contact.
161 enum ast_sip_contact_status_type {
167 * \brief A contact's status.
169 * \detail Maintains a contact's current status and round trip time
172 struct ast_sip_contact_status {
173 SORCERY_OBJECT(details);
174 /*! Current status for a contact (default - unavailable) */
175 enum ast_sip_contact_status_type status;
176 /*! The round trip start time set before sending a qualify request */
177 struct timeval rtt_start;
178 /*! The round trip time in microseconds */
183 * \brief A SIP address of record
186 /*! Sorcery object details, the id is the AOR name */
187 SORCERY_OBJECT(details);
188 AST_DECLARE_STRING_FIELDS(
189 /*! Voicemail boxes for this AOR */
190 AST_STRING_FIELD(mailboxes);
192 /*! Minimum expiration time */
193 unsigned int minimum_expiration;
194 /*! Maximum expiration time */
195 unsigned int maximum_expiration;
196 /*! Default contact expiration if one is not provided in the contact */
197 unsigned int default_expiration;
198 /*! Frequency to send OPTIONS requests to AOR contacts. 0 is disabled. */
199 unsigned int qualify_frequency;
200 /*! If true authenticate the qualify if needed */
201 int authenticate_qualify;
202 /*! Maximum number of external contacts, 0 to disable */
203 unsigned int max_contacts;
204 /*! Whether to remove any existing contacts not related to an incoming REGISTER when it comes in */
205 unsigned int remove_existing;
206 /*! Any permanent configured contacts */
207 struct ao2_container *permanent_contacts;
211 * \brief DTMF modes for SIP endpoints
213 enum ast_sip_dtmf_mode {
214 /*! No DTMF to be used */
216 /* XXX Should this be 2833 instead? */
217 /*! Use RFC 4733 events for DTMF */
218 AST_SIP_DTMF_RFC_4733,
219 /*! Use DTMF in the audio stream */
221 /*! Use SIP INFO DTMF (blech) */
226 * \brief Methods of storing SIP digest authentication credentials.
228 * Note that both methods result in MD5 digest authentication being
229 * used. The two methods simply alter how Asterisk determines the
230 * credentials for a SIP authentication
232 enum ast_sip_auth_type {
233 /*! Credentials stored as a username and password combination */
234 AST_SIP_AUTH_TYPE_USER_PASS,
235 /*! Credentials stored as an MD5 sum */
236 AST_SIP_AUTH_TYPE_MD5,
237 /*! Credentials not stored this is a fake auth */
238 AST_SIP_AUTH_TYPE_ARTIFICIAL
241 #define SIP_SORCERY_AUTH_TYPE "auth"
243 struct ast_sip_auth {
244 /* Sorcery ID of the auth is its name */
245 SORCERY_OBJECT(details);
246 AST_DECLARE_STRING_FIELDS(
247 /* Identification for these credentials */
248 AST_STRING_FIELD(realm);
249 /* Authentication username */
250 AST_STRING_FIELD(auth_user);
251 /* Authentication password */
252 AST_STRING_FIELD(auth_pass);
253 /* Authentication credentials in MD5 format (hash of user:realm:pass) */
254 AST_STRING_FIELD(md5_creds);
256 /* The time period (in seconds) that a nonce may be reused */
257 unsigned int nonce_lifetime;
258 /* Used to determine what to use when authenticating */
259 enum ast_sip_auth_type type;
262 struct ast_sip_auth_array {
263 /*! Array of Sorcery IDs of auth sections */
265 /*! Number of credentials in the array */
270 * \brief Different methods by which incoming requests can be matched to endpoints
272 enum ast_sip_endpoint_identifier_type {
273 /*! Identify based on user name in From header */
274 AST_SIP_ENDPOINT_IDENTIFY_BY_USERNAME = (1 << 0),
277 enum ast_sip_session_refresh_method {
278 /*! Use reinvite to negotiate direct media */
279 AST_SIP_SESSION_REFRESH_METHOD_INVITE,
280 /*! Use UPDATE to negotiate direct media */
281 AST_SIP_SESSION_REFRESH_METHOD_UPDATE,
284 enum ast_sip_direct_media_glare_mitigation {
285 /*! Take no special action to mitigate reinvite glare */
286 AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_NONE,
287 /*! Do not send an initial direct media session refresh on outgoing call legs
288 * Subsequent session refreshes will be sent no matter the session direction
290 AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_OUTGOING,
291 /*! Do not send an initial direct media session refresh on incoming call legs
292 * Subsequent session refreshes will be sent no matter the session direction
294 AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_INCOMING,
297 enum ast_sip_session_media_encryption {
298 /*! Invalid media encryption configuration */
299 AST_SIP_MEDIA_TRANSPORT_INVALID = 0,
300 /*! Do not allow any encryption of session media */
301 AST_SIP_MEDIA_ENCRYPT_NONE,
302 /*! Offer SDES-encrypted session media */
303 AST_SIP_MEDIA_ENCRYPT_SDES,
304 /*! Offer encrypted session media with datagram TLS key exchange */
305 AST_SIP_MEDIA_ENCRYPT_DTLS,
308 enum ast_sip_session_redirect {
309 /*! User portion of the target URI should be used as the target in the dialplan */
310 AST_SIP_REDIRECT_USER = 0,
311 /*! Target URI should be used as the target in the dialplan */
312 AST_SIP_REDIRECT_URI_CORE,
313 /*! Target URI should be used as the target within chan_pjsip itself */
314 AST_SIP_REDIRECT_URI_PJSIP,
318 * \brief Session timers options
320 struct ast_sip_timer_options {
321 /*! Minimum session expiration period, in seconds */
323 /*! Session expiration period, in seconds */
324 unsigned int sess_expires;
328 * \brief Endpoint configuration for SIP extensions.
330 * SIP extensions, in this case refers to features
331 * indicated in Supported or Required headers.
333 struct ast_sip_endpoint_extensions {
334 /*! Enabled SIP extensions */
337 struct ast_sip_timer_options timer;
341 * \brief Endpoint configuration for unsolicited MWI
343 struct ast_sip_mwi_configuration {
344 AST_DECLARE_STRING_FIELDS(
345 /*! Configured voicemail boxes for this endpoint. Used for MWI */
346 AST_STRING_FIELD(mailboxes);
347 /*! Username to use when sending MWI NOTIFYs to this endpoint */
348 AST_STRING_FIELD(fromuser);
350 /* Should mailbox states be combined into a single notification? */
351 unsigned int aggregate;
355 * \brief Endpoint subscription configuration
357 struct ast_sip_endpoint_subscription_configuration {
358 /*! Indicates if endpoint is allowed to initiate subscriptions */
360 /*! The minimum allowed expiration for subscriptions from endpoint */
361 unsigned int minexpiry;
362 /*! Message waiting configuration */
363 struct ast_sip_mwi_configuration mwi;
367 * \brief NAT configuration options for endpoints
369 struct ast_sip_endpoint_nat_configuration {
370 /*! Whether to force using the source IP address/port for sending responses */
371 unsigned int force_rport;
372 /*! Whether to rewrite the Contact header with the source IP address/port or not */
373 unsigned int rewrite_contact;
377 * \brief Party identification options for endpoints
379 * This includes caller ID, connected line, and redirecting-related options
381 struct ast_sip_endpoint_id_configuration {
382 struct ast_party_id self;
383 /*! Do we accept identification information from this endpoint */
384 unsigned int trust_inbound;
385 /*! Do we send private identification information to this endpoint? */
386 unsigned int trust_outbound;
387 /*! Do we send P-Asserted-Identity headers to this endpoint? */
388 unsigned int send_pai;
389 /*! Do we send Remote-Party-ID headers to this endpoint? */
390 unsigned int send_rpid;
391 /*! Do we add Diversion headers to applicable outgoing requests/responses? */
392 unsigned int send_diversion;
393 /*! When performing connected line update, which method should be used */
394 enum ast_sip_session_refresh_method refresh_method;
398 * \brief Call pickup configuration options for endpoints
400 struct ast_sip_endpoint_pickup_configuration {
402 ast_group_t callgroup;
404 ast_group_t pickupgroup;
405 /*! Named call group */
406 struct ast_namedgroups *named_callgroups;
407 /*! Named pickup group */
408 struct ast_namedgroups *named_pickupgroups;
412 * \brief Configuration for one-touch INFO recording
414 struct ast_sip_info_recording_configuration {
415 AST_DECLARE_STRING_FIELDS(
416 /*! Feature to enact when one-touch recording INFO with Record: On is received */
417 AST_STRING_FIELD(onfeature);
418 /*! Feature to enact when one-touch recording INFO with Record: Off is received */
419 AST_STRING_FIELD(offfeature);
421 /*! Is one-touch recording permitted? */
422 unsigned int enabled;
426 * \brief Endpoint configuration options for INFO packages
428 struct ast_sip_endpoint_info_configuration {
429 /*! Configuration for one-touch recording */
430 struct ast_sip_info_recording_configuration recording;
434 * \brief RTP configuration for SIP endpoints
436 struct ast_sip_media_rtp_configuration {
437 AST_DECLARE_STRING_FIELDS(
438 /*! Configured RTP engine for this endpoint. */
439 AST_STRING_FIELD(engine);
441 /*! Whether IPv6 RTP is enabled or not */
443 /*! Whether symmetric RTP is enabled or not */
444 unsigned int symmetric;
445 /*! Whether ICE support is enabled or not */
446 unsigned int ice_support;
447 /*! Whether to use the "ptime" attribute received from the endpoint or not */
448 unsigned int use_ptime;
449 /*! Do we use AVPF exclusively for this endpoint? */
450 unsigned int use_avpf;
451 /*! \brief DTLS-SRTP configuration information */
452 struct ast_rtp_dtls_cfg dtls_cfg;
453 /*! Should SRTP use a 32 byte tag instead of an 80 byte tag? */
454 unsigned int srtp_tag_32;
455 /*! Do we use media encryption? what type? */
456 enum ast_sip_session_media_encryption encryption;
460 * \brief Direct media options for SIP endpoints
462 struct ast_sip_direct_media_configuration {
463 /*! Boolean indicating if direct_media is permissible */
464 unsigned int enabled;
465 /*! When using direct media, which method should be used */
466 enum ast_sip_session_refresh_method method;
467 /*! Take steps to mitigate glare for direct media */
468 enum ast_sip_direct_media_glare_mitigation glare_mitigation;
469 /*! Do not attempt direct media session refreshes if a media NAT is detected */
470 unsigned int disable_on_nat;
473 struct ast_sip_t38_configuration {
474 /*! Whether T.38 UDPTL support is enabled or not */
475 unsigned int enabled;
476 /*! Error correction setting for T.38 UDPTL */
477 enum ast_t38_ec_modes error_correction;
478 /*! Explicit T.38 max datagram value, may be 0 to indicate the remote side can be trusted */
479 unsigned int maxdatagram;
480 /*! Whether NAT Support is enabled for T.38 UDPTL sessions or not */
482 /*! Whether to use IPv6 for UDPTL or not */
487 * \brief Media configuration for SIP endpoints
489 struct ast_sip_endpoint_media_configuration {
490 AST_DECLARE_STRING_FIELDS(
491 /*! Optional media address to use in SDP */
492 AST_STRING_FIELD(address);
493 /*! SDP origin username */
494 AST_STRING_FIELD(sdpowner);
495 /*! SDP session name */
496 AST_STRING_FIELD(sdpsession);
498 /*! RTP media configuration */
499 struct ast_sip_media_rtp_configuration rtp;
500 /*! Direct media options */
501 struct ast_sip_direct_media_configuration direct_media;
502 /*! T.38 (FoIP) options */
503 struct ast_sip_t38_configuration t38;
504 /*! Codec preferences */
505 struct ast_codec_pref prefs;
506 /*! Configured codecs */
507 struct ast_format_cap *codecs;
508 /*! DSCP TOS bits for audio streams */
509 unsigned int tos_audio;
510 /*! Priority for audio streams */
511 unsigned int cos_audio;
512 /*! DSCP TOS bits for video streams */
513 unsigned int tos_video;
514 /*! Priority for video streams */
515 unsigned int cos_video;
519 * \brief An entity with which Asterisk communicates
521 struct ast_sip_endpoint {
522 SORCERY_OBJECT(details);
523 AST_DECLARE_STRING_FIELDS(
524 /*! Context to send incoming calls to */
525 AST_STRING_FIELD(context);
526 /*! Name of an explicit transport to use */
527 AST_STRING_FIELD(transport);
528 /*! Outbound proxy to use */
529 AST_STRING_FIELD(outbound_proxy);
530 /*! Explicit AORs to dial if none are specified */
531 AST_STRING_FIELD(aors);
532 /*! Musiconhold class to suggest that the other side use when placing on hold */
533 AST_STRING_FIELD(mohsuggest);
534 /*! Configured tone zone for this endpoint. */
535 AST_STRING_FIELD(zone);
536 /*! Configured language for this endpoint. */
537 AST_STRING_FIELD(language);
538 /*! Default username to place in From header */
539 AST_STRING_FIELD(fromuser);
540 /*! Domain to place in From header */
541 AST_STRING_FIELD(fromdomain);
543 /*! Configuration for extensions */
544 struct ast_sip_endpoint_extensions extensions;
545 /*! Configuration relating to media */
546 struct ast_sip_endpoint_media_configuration media;
547 /*! SUBSCRIBE/NOTIFY configuration options */
548 struct ast_sip_endpoint_subscription_configuration subscription;
549 /*! NAT configuration */
550 struct ast_sip_endpoint_nat_configuration nat;
551 /*! Party identification options */
552 struct ast_sip_endpoint_id_configuration id;
553 /*! Configuration options for INFO packages */
554 struct ast_sip_endpoint_info_configuration info;
555 /*! Call pickup configuration */
556 struct ast_sip_endpoint_pickup_configuration pickup;
557 /*! Inbound authentication credentials */
558 struct ast_sip_auth_array inbound_auths;
559 /*! Outbound authentication credentials */
560 struct ast_sip_auth_array outbound_auths;
561 /*! DTMF mode to use with this endpoint */
562 enum ast_sip_dtmf_mode dtmf;
563 /*! Method(s) by which the endpoint should be identified. */
564 enum ast_sip_endpoint_identifier_type ident_method;
565 /*! Boolean indicating if ringing should be sent as inband progress */
566 unsigned int inband_progress;
567 /*! Pointer to the persistent Asterisk endpoint */
568 struct ast_endpoint *persistent;
569 /*! The number of channels at which busy device state is returned */
570 unsigned int devicestate_busy_at;
571 /*! Whether fax detection is enabled or not (CNG tone detection) */
572 unsigned int faxdetect;
573 /*! Determines if transfers (using REFER) are allowed by this endpoint */
574 unsigned int allowtransfer;
575 /*! Method used when handling redirects */
576 enum ast_sip_session_redirect redirect_method;
580 * \brief Initialize an auth array with the configured values.
582 * \param array Array to initialize
583 * \param auth_names Comma-separated list of names to set in the array
585 * \retval non-zero Failure
587 int ast_sip_auth_array_init(struct ast_sip_auth_array *array, const char *auth_names);
590 * \brief Free contents of an auth array.
592 * \param array Array whose contents are to be freed
594 void ast_sip_auth_array_destroy(struct ast_sip_auth_array *array);
597 * \brief Possible returns from ast_sip_check_authentication
599 enum ast_sip_check_auth_result {
600 /*! Authentication needs to be challenged */
601 AST_SIP_AUTHENTICATION_CHALLENGE,
602 /*! Authentication succeeded */
603 AST_SIP_AUTHENTICATION_SUCCESS,
604 /*! Authentication failed */
605 AST_SIP_AUTHENTICATION_FAILED,
606 /*! Authentication encountered some internal error */
607 AST_SIP_AUTHENTICATION_ERROR,
611 * \brief An interchangeable way of handling digest authentication for SIP.
613 * An authenticator is responsible for filling in the callbacks provided below. Each is called from a publicly available
614 * function in res_sip. The authenticator can use configuration or other local policy to determine whether authentication
615 * should take place and what credentials should be used when challenging and authenticating a request.
617 struct ast_sip_authenticator {
619 * \brief Check if a request requires authentication
620 * See ast_sip_requires_authentication for more details
622 int (*requires_authentication)(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata);
624 * \brief Check that an incoming request passes authentication.
626 * The tdata parameter is useful for adding information such as digest challenges.
628 * \param endpoint The endpoint sending the incoming request
629 * \param rdata The incoming request
630 * \param tdata Tentative outgoing request.
632 enum ast_sip_check_auth_result (*check_authentication)(struct ast_sip_endpoint *endpoint,
633 pjsip_rx_data *rdata, pjsip_tx_data *tdata);
637 * \brief an interchangeable way of responding to authentication challenges
639 * An outbound authenticator takes incoming challenges and formulates a new SIP request with
642 struct ast_sip_outbound_authenticator {
644 * \brief Create a new request with authentication credentials
646 * \param auths An array of IDs of auth sorcery objects
647 * \param challenge The SIP response with authentication challenge(s)
648 * \param tsx The transaction in which the challenge was received
649 * \param new_request The new SIP request with challenge response(s)
650 * \retval 0 Successfully created new request
651 * \retval -1 Failed to create a new request
653 int (*create_request_with_auth)(const struct ast_sip_auth_array *auths, struct pjsip_rx_data *challenge,
654 struct pjsip_transaction *tsx, struct pjsip_tx_data **new_request);
658 * \brief An entity responsible for identifying the source of a SIP message
660 struct ast_sip_endpoint_identifier {
662 * \brief Callback used to identify the source of a message.
663 * See ast_sip_identify_endpoint for more details
665 struct ast_sip_endpoint *(*identify_endpoint)(pjsip_rx_data *rdata);
669 * \brief Register a SIP service in Asterisk.
671 * This is more-or-less a wrapper around pjsip_endpt_register_module().
672 * Registering a service makes it so that PJSIP will call into the
673 * service at appropriate times. For more information about PJSIP module
674 * callbacks, see the PJSIP documentation. Asterisk modules that call
675 * this function will likely do so at module load time.
677 * \param module The module that is to be registered with PJSIP
681 int ast_sip_register_service(pjsip_module *module);
684 * This is the opposite of ast_sip_register_service(). Unregistering a
685 * service means that PJSIP will no longer call into the module any more.
686 * This will likely occur when an Asterisk module is unloaded.
688 * \param module The PJSIP module to unregister
690 void ast_sip_unregister_service(pjsip_module *module);
693 * \brief Register a SIP authenticator
695 * An authenticator has three main purposes:
696 * 1) Determining if authentication should be performed on an incoming request
697 * 2) Gathering credentials necessary for issuing an authentication challenge
698 * 3) Authenticating a request that has credentials
700 * Asterisk provides a default authenticator, but it may be replaced by a
701 * custom one if desired.
703 * \param auth The authenticator to register
707 int ast_sip_register_authenticator(struct ast_sip_authenticator *auth);
710 * \brief Unregister a SIP authenticator
712 * When there is no authenticator registered, requests cannot be challenged
715 * \param auth The authenticator to unregister
717 void ast_sip_unregister_authenticator(struct ast_sip_authenticator *auth);
720 * \brief Register an outbound SIP authenticator
722 * An outbound authenticator is responsible for creating responses to
723 * authentication challenges by remote endpoints.
725 * \param auth The authenticator to register
729 int ast_sip_register_outbound_authenticator(struct ast_sip_outbound_authenticator *outbound_auth);
732 * \brief Unregister an outbound SIP authenticator
734 * When there is no outbound authenticator registered, authentication challenges
735 * will be handled as any other final response would be.
737 * \param auth The authenticator to unregister
739 void ast_sip_unregister_outbound_authenticator(struct ast_sip_outbound_authenticator *auth);
742 * \brief Register a SIP endpoint identifier
744 * An endpoint identifier's purpose is to determine which endpoint a given SIP
745 * message has come from.
747 * Multiple endpoint identifiers may be registered so that if an endpoint
748 * cannot be identified by one identifier, it may be identified by another.
750 * Asterisk provides two endpoint identifiers. One identifies endpoints based
751 * on the user part of the From header URI. The other identifies endpoints based
752 * on the source IP address.
754 * If the order in which endpoint identifiers is run is important to you, then
755 * be sure to load individual endpoint identifier modules in the order you wish
756 * for them to be run in modules.conf
758 * \param identifier The SIP endpoint identifier to register
762 int ast_sip_register_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier);
765 * \brief Unregister a SIP endpoint identifier
767 * This stops an endpoint identifier from being used.
769 * \param identifier The SIP endoint identifier to unregister
771 void ast_sip_unregister_endpoint_identifier(struct ast_sip_endpoint_identifier *identifier);
774 * \brief Allocate a new SIP endpoint
776 * This will return an endpoint with its refcount increased by one. This reference
777 * can be released using ao2_ref().
779 * \param name The name of the endpoint.
780 * \retval NULL Endpoint allocation failed
781 * \retval non-NULL The newly allocated endpoint
783 void *ast_sip_endpoint_alloc(const char *name);
786 * \brief Get a pointer to the PJSIP endpoint.
788 * This is useful when modules have specific information they need
789 * to register with the PJSIP core.
790 * \retval NULL endpoint has not been created yet.
791 * \retval non-NULL PJSIP endpoint.
793 pjsip_endpoint *ast_sip_get_pjsip_endpoint(void);
796 * \brief Get a pointer to the SIP sorcery structure.
798 * \retval NULL sorcery has not been initialized
799 * \retval non-NULL sorcery structure
801 struct ast_sorcery *ast_sip_get_sorcery(void);
804 * \brief Initialize transport support on a sorcery instance
806 * \param sorcery The sorcery instance
811 int ast_sip_initialize_sorcery_transport(struct ast_sorcery *sorcery);
814 * \brief Initialize qualify support on a sorcery instance
816 * \param sorcery The sorcery instance
821 int ast_sip_initialize_sorcery_qualify(struct ast_sorcery *sorcery);
824 * \brief Initialize location support on a sorcery instance
826 * \param sorcery The sorcery instance
831 int ast_sip_initialize_sorcery_location(struct ast_sorcery *sorcery);
834 * \brief Retrieve a named AOR
836 * \param aor_name Name of the AOR
838 * \retval NULL if not found
839 * \retval non-NULL if found
841 struct ast_sip_aor *ast_sip_location_retrieve_aor(const char *aor_name);
844 * \brief Retrieve the first bound contact for an AOR
846 * \param aor Pointer to the AOR
847 * \retval NULL if no contacts available
848 * \retval non-NULL if contacts available
850 struct ast_sip_contact *ast_sip_location_retrieve_first_aor_contact(const struct ast_sip_aor *aor);
853 * \brief Retrieve all contacts currently available for an AOR
855 * \param aor Pointer to the AOR
857 * \retval NULL if no contacts available
858 * \retval non-NULL if contacts available
860 struct ao2_container *ast_sip_location_retrieve_aor_contacts(const struct ast_sip_aor *aor);
863 * \brief Retrieve the first bound contact from a list of AORs
865 * \param aor_list A comma-separated list of AOR names
866 * \retval NULL if no contacts available
867 * \retval non-NULL if contacts available
869 struct ast_sip_contact *ast_sip_location_retrieve_contact_from_aor_list(const char *aor_list);
872 * \brief Retrieve a named contact
874 * \param contact_name Name of the contact
876 * \retval NULL if not found
877 * \retval non-NULL if found
879 struct ast_sip_contact *ast_sip_location_retrieve_contact(const char *contact_name);
882 * \brief Add a new contact to an AOR
884 * \param aor Pointer to the AOR
885 * \param uri Full contact URI
886 * \param expiration_time Optional expiration time of the contact
891 int ast_sip_location_add_contact(struct ast_sip_aor *aor, const char *uri, struct timeval expiration_time);
894 * \brief Update a contact
896 * \param contact New contact object with details
901 int ast_sip_location_update_contact(struct ast_sip_contact *contact);
904 * \brief Delete a contact
906 * \param contact Contact object to delete
911 int ast_sip_location_delete_contact(struct ast_sip_contact *contact);
914 * \brief Initialize domain aliases support on a sorcery instance
916 * \param sorcery The sorcery instance
921 int ast_sip_initialize_sorcery_domain_alias(struct ast_sorcery *sorcery);
924 * \brief Initialize authentication support on a sorcery instance
926 * \param sorcery The sorcery instance
931 int ast_sip_initialize_sorcery_auth(struct ast_sorcery *sorcery);
934 * \brief Callback called when an outbound request with authentication credentials is to be sent in dialog
936 * This callback will have the created request on it. The callback's purpose is to do any extra
937 * housekeeping that needs to be done as well as to send the request out.
939 * This callback is only necessary if working with a PJSIP API that sits between the application
940 * and the dialog layer.
942 * \param dlg The dialog to which the request belongs
943 * \param tdata The created request to be sent out
944 * \param user_data Data supplied with the callback
949 typedef int (*ast_sip_dialog_outbound_auth_cb)(pjsip_dialog *dlg, pjsip_tx_data *tdata, void *user_data);
952 * \brief Set up outbound authentication on a SIP dialog
954 * This sets up the infrastructure so that all requests associated with a created dialog
955 * can be re-sent with authentication credentials if the original request is challenged.
957 * \param dlg The dialog on which requests will be authenticated
958 * \param endpoint The endpoint whom this dialog pertains to
959 * \param cb Callback to call to send requests with authentication
960 * \param user_data Data to be provided to the callback when it is called
965 int ast_sip_dialog_setup_outbound_authentication(pjsip_dialog *dlg, const struct ast_sip_endpoint *endpoint,
966 ast_sip_dialog_outbound_auth_cb cb, void *user_data);
969 * \brief Initialize the distributor module
971 * The distributor module is responsible for taking an incoming
972 * SIP message and placing it into the threadpool. Once in the threadpool,
973 * the distributor will perform endpoint lookups and authentication, and
974 * then distribute the message up the stack to any further modules.
979 int ast_sip_initialize_distributor(void);
982 * \brief Destruct the distributor module.
984 * Unregisters pjsip modules and cleans up any allocated resources.
986 void ast_sip_destroy_distributor(void);
989 * \brief Retrieves a reference to the artificial auth.
991 * \retval The artificial auth
993 struct ast_sip_auth *ast_sip_get_artificial_auth(void);
996 * \brief Retrieves a reference to the artificial endpoint.
998 * \retval The artificial endpoint
1000 struct ast_sip_endpoint *ast_sip_get_artificial_endpoint(void);
1003 * \page Threading model for SIP
1005 * There are three major types of threads that SIP will have to deal with:
1006 * \li Asterisk threads
1008 * \li SIP threadpool threads (a.k.a. "servants")
1010 * \par Asterisk Threads
1012 * Asterisk threads are those that originate from outside of SIP but within
1013 * Asterisk. The most common of these threads are PBX (channel) threads and
1014 * the autoservice thread. Most interaction with these threads will be through
1015 * channel technology callbacks. Within these threads, it is fine to handle
1016 * Asterisk data from outside of SIP, but any handling of SIP data should be
1017 * left to servants, \b especially if you wish to call into PJSIP for anything.
1018 * Asterisk threads are not registered with PJLIB, so attempting to call into
1019 * PJSIP will cause an assertion to be triggered, thus causing the program to
1022 * \par PJSIP Threads
1024 * PJSIP threads are those that originate from handling of PJSIP events, such
1025 * as an incoming SIP request or response, or a transaction timeout. The role
1026 * of these threads is to process information as quickly as possible so that
1027 * the next item on the SIP socket(s) can be serviced. On incoming messages,
1028 * Asterisk automatically will push the request to a servant thread. When your
1029 * module callback is called, processing will already be in a servant. However,
1030 * for other PSJIP events, such as transaction state changes due to timer
1031 * expirations, your module will be called into from a PJSIP thread. If you
1032 * are called into from a PJSIP thread, then you should push whatever processing
1033 * is needed to a servant as soon as possible. You can discern if you are currently
1034 * in a SIP servant thread using the \ref ast_sip_thread_is_servant function.
1038 * Servants are where the bulk of SIP work should be performed. These threads
1039 * exist in order to do the work that Asterisk threads and PJSIP threads hand
1040 * off to them. Servant threads register themselves with PJLIB, meaning that
1041 * they are capable of calling PJSIP and PJLIB functions if they wish.
1045 * Tasks are handed off to servant threads using the API call \ref ast_sip_push_task.
1046 * The first parameter of this call is a serializer. If this pointer
1047 * is NULL, then the work will be handed off to whatever servant can currently handle
1048 * the task. If this pointer is non-NULL, then the task will not be executed until
1049 * previous tasks pushed with the same serializer have completed. For more information
1050 * on serializers and the benefits they provide, see \ref ast_threadpool_serializer
1054 * Do not make assumptions about individual threads based on a corresponding serializer.
1055 * In other words, just because several tasks use the same serializer when being pushed
1056 * to servants, it does not mean that the same thread is necessarily going to execute those
1057 * tasks, even though they are all guaranteed to be executed in sequence.
1061 * \brief Create a new serializer for SIP tasks
1063 * See \ref ast_threadpool_serializer for more information on serializers.
1064 * SIP creates serializers so that tasks operating on similar data will run
1067 * \retval NULL Failure
1068 * \retval non-NULL Newly-created serializer
1070 struct ast_taskprocessor *ast_sip_create_serializer(void);
1073 * \brief Set a serializer on a SIP dialog so requests and responses are automatically serialized
1075 * Passing a NULL serializer is a way to remove a serializer from a dialog.
1077 * \param dlg The SIP dialog itself
1078 * \param serializer The serializer to use
1080 void ast_sip_dialog_set_serializer(pjsip_dialog *dlg, struct ast_taskprocessor *serializer);
1083 * \brief Set an endpoint on a SIP dialog so in-dialog requests do not undergo endpoint lookup.
1085 * \param dlg The SIP dialog itself
1086 * \param endpoint The endpoint that this dialog is communicating with
1088 void ast_sip_dialog_set_endpoint(pjsip_dialog *dlg, struct ast_sip_endpoint *endpoint);
1091 * \brief Get the endpoint associated with this dialog
1093 * This function increases the refcount of the endpoint by one. Release
1094 * the reference once you are finished with the endpoint.
1096 * \param dlg The SIP dialog from which to retrieve the endpoint
1097 * \retval NULL No endpoint associated with this dialog
1098 * \retval non-NULL The endpoint.
1100 struct ast_sip_endpoint *ast_sip_dialog_get_endpoint(pjsip_dialog *dlg);
1103 * \brief Pushes a task to SIP servants
1105 * This uses the serializer provided to determine how to push the task.
1106 * If the serializer is NULL, then the task will be pushed to the
1107 * servants directly. If the serializer is non-NULL, then the task will be
1108 * queued behind other tasks associated with the same serializer.
1110 * \param serializer The serializer to which the task belongs. Can be NULL
1111 * \param sip_task The task to execute
1112 * \param task_data The parameter to pass to the task when it executes
1114 * \retval -1 Failure
1116 int ast_sip_push_task(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data);
1119 * \brief Push a task to SIP servants and wait for it to complete
1121 * Like \ref ast_sip_push_task except that it blocks until the task completes.
1123 * \warning \b Never use this function in a SIP servant thread. This can potentially
1124 * cause a deadlock. If you are in a SIP servant thread, just call your function
1127 * \param serializer The SIP serializer to which the task belongs. May be NULL.
1128 * \param sip_task The task to execute
1129 * \param task_data The parameter to pass to the task when it executes
1131 * \retval -1 Failure
1133 int ast_sip_push_task_synchronous(struct ast_taskprocessor *serializer, int (*sip_task)(void *), void *task_data);
1136 * \brief Determine if the current thread is a SIP servant thread
1138 * \retval 0 This is not a SIP servant thread
1139 * \retval 1 This is a SIP servant thread
1141 int ast_sip_thread_is_servant(void);
1144 * \brief SIP body description
1146 * This contains a type and subtype that will be added as
1147 * the "Content-Type" for the message as well as the body
1150 struct ast_sip_body {
1151 /*! Type of the body, such as "application" */
1153 /*! Subtype of the body, such as "sdp" */
1154 const char *subtype;
1155 /*! The text to go in the body */
1156 const char *body_text;
1160 * \brief General purpose method for creating a UAC dialog with an endpoint
1162 * \param endpoint A pointer to the endpoint
1163 * \param aor_name Optional name of the AOR to target, may even be an explicit SIP URI
1164 * \param request_user Optional user to place into the target URI
1166 * \retval non-NULL success
1167 * \retval NULL failure
1169 pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint, const char *aor_name, const char *request_user);
1172 * \brief General purpose method for creating a UAS dialog with an endpoint
1174 * \param endpoint A pointer to the endpoint
1175 * \param rdata The request that is starting the dialog
1177 pjsip_dialog *ast_sip_create_dialog_uas(const struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata);
1180 * \brief General purpose method for creating a SIP request
1182 * Its typical use would be to create one-off requests such as an out of dialog
1185 * The request can either be in- or out-of-dialog. If in-dialog, the
1186 * dlg parameter MUST be present. If out-of-dialog the endpoint parameter
1187 * MUST be present. If both are present, then we will assume that the message
1188 * is to be sent in-dialog.
1190 * The uri parameter can be specified if the request should be sent to an explicit
1191 * URI rather than one configured on the endpoint.
1193 * \param method The method of the SIP request to send
1194 * \param dlg Optional. If specified, the dialog on which to request the message.
1195 * \param endpoint Optional. If specified, the request will be created out-of-dialog
1197 * \param uri Optional. If specified, the request will be sent to this URI rather
1199 * than one configured for the endpoint.
1200 * \param[out] tdata The newly-created request
1202 * \retval -1 Failure
1204 int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
1205 struct ast_sip_endpoint *endpoint, const char *uri,
1206 pjsip_tx_data **tdata);
1209 * \brief General purpose method for sending a SIP request
1211 * This is a companion function for \ref ast_sip_create_request. The request
1212 * created there can be passed to this function, though any request may be
1215 * This will automatically set up handling outbound authentication challenges if
1218 * \param tdata The request to send
1219 * \param dlg Optional. If specified, the dialog on which the request should be sent
1220 * \param endpoint Optional. If specified, the request is sent out-of-dialog to the endpoint.
1222 * \retval -1 Failure
1224 int ast_sip_send_request(pjsip_tx_data *tdata, struct pjsip_dialog *dlg, struct ast_sip_endpoint *endpoint);
1227 * \brief Determine if an incoming request requires authentication
1229 * This calls into the registered authenticator's requires_authentication callback
1230 * in order to determine if the request requires authentication.
1232 * If there is no registered authenticator, then authentication will be assumed
1233 * not to be required.
1235 * \param endpoint The endpoint from which the request originates
1236 * \param rdata The incoming SIP request
1237 * \retval non-zero The request requires authentication
1238 * \retval 0 The request does not require authentication
1240 int ast_sip_requires_authentication(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata);
1243 * \brief Method to determine authentication status of an incoming request
1245 * This will call into a registered authenticator. The registered authenticator will
1246 * do what is necessary to determine whether the incoming request passes authentication.
1247 * A tentative response is passed into this function so that if, say, a digest authentication
1248 * challenge should be sent in the ensuing response, it can be added to the response.
1250 * \param endpoint The endpoint from the request was sent
1251 * \param rdata The request to potentially authenticate
1252 * \param tdata Tentative response to the request
1253 * \return The result of checking authentication.
1255 enum ast_sip_check_auth_result ast_sip_check_authentication(struct ast_sip_endpoint *endpoint,
1256 pjsip_rx_data *rdata, pjsip_tx_data *tdata);
1259 * \brief Create a response to an authentication challenge
1261 * This will call into an outbound authenticator's create_request_with_auth callback
1262 * to create a new request with authentication credentials. See the create_request_with_auth
1263 * callback in the \ref ast_sip_outbound_authenticator structure for details about
1264 * the parameters and return values.
1266 int ast_sip_create_request_with_auth(const struct ast_sip_auth_array *auths, pjsip_rx_data *challenge,
1267 pjsip_transaction *tsx, pjsip_tx_data **new_request);
1270 * \brief Determine the endpoint that has sent a SIP message
1272 * This will call into each of the registered endpoint identifiers'
1273 * identify_endpoint() callbacks until one returns a non-NULL endpoint.
1274 * This will return an ao2 object. Its reference count will need to be
1275 * decremented when completed using the endpoint.
1277 * \param rdata The inbound SIP message to use when identifying the endpoint.
1278 * \retval NULL No matching endpoint
1279 * \retval non-NULL The matching endpoint
1281 struct ast_sip_endpoint *ast_sip_identify_endpoint(pjsip_rx_data *rdata);
1284 * \brief Add a header to an outbound SIP message
1286 * \param tdata The message to add the header to
1287 * \param name The header name
1288 * \param value The header value
1290 * \retval -1 Failure
1292 int ast_sip_add_header(pjsip_tx_data *tdata, const char *name, const char *value);
1295 * \brief Add a body to an outbound SIP message
1297 * If this is called multiple times, the latest body will replace the current
1300 * \param tdata The message to add the body to
1301 * \param body The message body to add
1303 * \retval -1 Failure
1305 int ast_sip_add_body(pjsip_tx_data *tdata, const struct ast_sip_body *body);
1308 * \brief Add a multipart body to an outbound SIP message
1310 * This will treat each part of the input array as part of a multipart body and
1311 * add each part to the SIP message.
1313 * \param tdata The message to add the body to
1314 * \param bodies The parts of the body to add
1316 * \retval -1 Failure
1318 int ast_sip_add_body_multipart(pjsip_tx_data *tdata, const struct ast_sip_body *bodies[], int num_bodies);
1321 * \brief Append body data to a SIP message
1323 * This acts mostly the same as ast_sip_add_body, except that rather than replacing
1324 * a body if it currently exists, it appends data to an existing body.
1326 * \param tdata The message to append the body to
1327 * \param body The string to append to the end of the current body
1329 * \retval -1 Failure
1331 int ast_sip_append_body(pjsip_tx_data *tdata, const char *body_text);
1334 * \brief Copy a pj_str_t into a standard character buffer.
1336 * pj_str_t is not NULL-terminated. Any place that expects a NULL-
1337 * terminated string needs to have the pj_str_t copied into a separate
1340 * This method copies the pj_str_t contents into the destination buffer
1341 * and NULL-terminates the buffer.
1343 * \param dest The destination buffer
1344 * \param src The pj_str_t to copy
1345 * \param size The size of the destination buffer.
1347 void ast_copy_pj_str(char *dest, const pj_str_t *src, size_t size);
1350 * \brief Get the looked-up endpoint on an out-of dialog request or response
1352 * The function may ONLY be called on out-of-dialog requests or responses. For
1353 * in-dialog requests and responses, it is required that the user of the dialog
1354 * has the looked-up endpoint stored locally.
1356 * This function should never return NULL if the message is out-of-dialog. It will
1357 * always return NULL if the message is in-dialog.
1359 * This function will increase the reference count of the returned endpoint by one.
1360 * Release your reference using the ao2_ref function when finished.
1362 * \param rdata Out-of-dialog request or response
1363 * \return The looked up endpoint
1365 struct ast_sip_endpoint *ast_pjsip_rdata_get_endpoint(pjsip_rx_data *rdata);
1368 * \brief Retrieve any endpoints available to sorcery.
1370 * \retval Endpoints available to sorcery, NULL if no endpoints found.
1372 struct ao2_container *ast_sip_get_endpoints(void);
1375 * \brief Retrieve relevant SIP auth structures from sorcery
1377 * \param auths Array of sorcery IDs of auth credentials to retrieve
1378 * \param[out] out The retrieved auths are stored here
1380 int ast_sip_retrieve_auths(const struct ast_sip_auth_array *auths, struct ast_sip_auth **out);
1383 * \brief Clean up retrieved auth structures from memory
1385 * Call this function once you have completed operating on auths
1386 * retrieved from \ref ast_sip_retrieve_auths
1388 * \param auths An array of auth structures to clean up
1389 * \param num_auths The number of auths in the array
1391 void ast_sip_cleanup_auths(struct ast_sip_auth *auths[], size_t num_auths);
1394 * \brief Checks if the given content type matches type/subtype.
1396 * Compares the pjsip_media_type with the passed type and subtype and
1397 * returns the result of that comparison. The media type parameters are
1400 * \param content_type The pjsip_media_type structure to compare
1401 * \param type The media type to compare
1402 * \param subtype The media subtype to compare
1403 * \retval 0 No match
1406 int ast_sip_is_content_type(pjsip_media_type *content_type, char *type, char *subtype);
1409 * \brief Send a security event notification for when an invalid endpoint is requested
1411 * \param name Name of the endpoint requested
1412 * \param rdata Received message
1414 void ast_sip_report_invalid_endpoint(const char *name, pjsip_rx_data *rdata);
1417 * \brief Send a security event notification for when an ACL check fails
1419 * \param endpoint Pointer to the endpoint in use
1420 * \param rdata Received message
1421 * \param name Name of the ACL
1423 void ast_sip_report_failed_acl(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata, const char *name);
1426 * \brief Send a security event notification for when a challenge response has failed
1428 * \param endpoint Pointer to the endpoint in use
1429 * \param rdata Received message
1431 void ast_sip_report_auth_failed_challenge_response(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata);
1434 * \brief Send a security event notification for when authentication succeeds
1436 * \param endpoint Pointer to the endpoint in use
1437 * \param rdata Received message
1439 void ast_sip_report_auth_success(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata);
1442 * \brief Send a security event notification for when an authentication challenge is sent
1444 * \param endpoint Pointer to the endpoint in use
1445 * \param rdata Received message
1446 * \param tdata Sent message
1448 void ast_sip_report_auth_challenge_sent(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata, pjsip_tx_data *tdata);
1451 * \brief Send a security event notification for when a request is not supported
1453 * \param endpoint Pointer to the endpoint in use
1454 * \param rdata Received message
1455 * \param req_type the type of request
1457 void ast_sip_report_req_no_support(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata,
1458 const char* req_type);
1461 * \brief Send a security event notification for when a memory limit is hit.
1463 * \param endpoint Pointer to the endpoint in use
1464 * \param rdata Received message
1466 void ast_sip_report_mem_limit(struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata);
1468 void ast_sip_initialize_global_headers(void);
1469 void ast_sip_destroy_global_headers(void);
1471 int ast_sip_add_global_request_header(const char *name, const char *value, int replace);
1472 int ast_sip_add_global_response_header(const char *name, const char *value, int replace);
1474 int ast_sip_initialize_sorcery_global(struct ast_sorcery *sorcery);
1477 * \brief Retrieves the value associated with the given key.
1479 * \param ht the hash table/dictionary to search
1480 * \param key the key to find
1482 * \retval the value associated with the key, NULL otherwise.
1484 void *ast_sip_dict_get(void *ht, const char *key);
1487 * \brief Using the dictionary stored in mod_data array at a given id,
1488 * retrieve the value associated with the given key.
1490 * \param mod_data a module data array
1491 * \param id the mod_data array index
1492 * \param key the key to find
1494 * \retval the value associated with the key, NULL otherwise.
1496 #define ast_sip_mod_data_get(mod_data, id, key) \
1497 ast_sip_dict_get(mod_data[id], key)
1500 * \brief Set the value for the given key.
1502 * Note - if the hash table does not exist one is created first, the key/value
1503 * pair is set, and the hash table returned.
1505 * \param pool the pool to allocate memory in
1506 * \param ht the hash table/dictionary in which to store the key/value pair
1507 * \param key the key to associate a value with
1508 * \param val the value to associate with a key
1510 * \retval the given, or newly created, hash table.
1512 void *ast_sip_dict_set(pj_pool_t* pool, void *ht,
1513 const char *key, void *val);
1516 * \brief Utilizing a mod_data array for a given id, set the value
1517 * associated with the given key.
1519 * For a given structure's mod_data array set the element indexed by id to
1520 * be a dictionary containing the key/val pair.
1522 * \param pool a memory allocation pool
1523 * \param mod_data a module data array
1524 * \param id the mod_data array index
1525 * \param key the key to find
1526 * \param val the value to associate with a key
1528 #define ast_sip_mod_data_set(pool, mod_data, id, key, val) \
1529 mod_data[id] = ast_sip_dict_set(pool, mod_data[id], key, val)
1532 * \brief Function pointer for contact callbacks.
1534 typedef int (*on_contact_t)(const struct ast_sip_aor *aor,
1535 const struct ast_sip_contact *contact,
1536 int last, void *arg);
1539 * \brief For every contact on an AOR call the given 'on_contact' handler.
1541 * \param aor the aor containing a list of contacts to iterate
1542 * \param on_contact callback on each contact on an AOR
1543 * \param arg user data passed to handler
1544 * \retval 0 Success, non-zero on failure
1546 int ast_sip_for_each_contact(const struct ast_sip_aor *aor,
1547 on_contact_t on_contact, void *arg);
1550 * \brief Handler used to convert a contact to a string.
1552 * \param aor the aor containing a list of contacts to iterate
1553 * \param contact the contact to convert
1554 * \param last is this the last contact
1555 * \param arg user data passed to handler
1556 * \retval 0 Success, non-zero on failure
1558 int ast_sip_contact_to_str(const struct ast_sip_aor *aor,
1559 const struct ast_sip_contact *contact,
1560 int last, void *arg);
1563 * \brief For every aor in the comma separated aors string call the
1564 * given 'on_aor' handler.
1566 * \param aors a comma separated list of aors
1567 * \param on_aor callback for each aor
1568 * \param arg user data passed to handler
1569 * \retval 0 Success, non-zero on failure
1571 int ast_sip_for_each_aor(const char *aors, ao2_callback_fn on_aor, void *arg);
1574 * \brief For every auth in the array call the given 'on_auth' handler.
1576 * \param array an array of auths
1577 * \param on_auth callback for each auth
1578 * \param arg user data passed to handler
1579 * \retval 0 Success, non-zero on failure
1581 int ast_sip_for_each_auth(const struct ast_sip_auth_array *array,
1582 ao2_callback_fn on_auth, void *arg);
1585 * \brief Converts the given auth type to a string
1587 * \param type the auth type to convert
1588 * \retval a string representative of the auth type
1590 const char *ast_sip_auth_type_to_str(enum ast_sip_auth_type type);
1593 * \brief Converts an auths array to a string of comma separated values
1595 * \param auths an auth array
1596 * \param buf the string buffer to write the object data
1597 * \retval 0 Success, non-zero on failure
1599 int ast_sip_auths_to_str(const struct ast_sip_auth_array *auths, char **buf);
1602 * \brief AMI variable container
1604 struct ast_sip_ami {
1605 /*! Manager session */
1606 struct mansession *s;
1607 /*! Manager message */
1608 const struct message *m;
1609 /*! user specified argument data */
1614 * \brief Creates a string to store AMI event data in.
1616 * \param event the event to set
1617 * \param ami AMI session and message container
1618 * \retval an initialized ast_str or NULL on error.
1620 struct ast_str *ast_sip_create_ami_event(const char *event,
1621 struct ast_sip_ami *ami);
1624 * \brief An entity responsible formatting endpoint information.
1626 struct ast_sip_endpoint_formatter {
1628 * \brief Callback used to format endpoint information over AMI.
1630 int (*format_ami)(const struct ast_sip_endpoint *endpoint,
1631 struct ast_sip_ami *ami);
1632 AST_RWLIST_ENTRY(ast_sip_endpoint_formatter) next;
1636 * \brief Register an endpoint formatter.
1638 * \param obj the formatter to register
1640 * \retval -1 Failure
1642 int ast_sip_register_endpoint_formatter(struct ast_sip_endpoint_formatter *obj);
1645 * \brief Unregister an endpoint formatter.
1647 * \param obj the formatter to unregister
1649 void ast_sip_unregister_endpoint_formatter(struct ast_sip_endpoint_formatter *obj);
1652 * \brief Converts a sorcery object to a string of object properties.
1654 * \param obj the sorcery object to convert
1655 * \param str the string buffer to write the object data
1656 * \retval 0 Success, non-zero on failure
1658 int ast_sip_sorcery_object_to_ami(const void *obj, struct ast_str **buf);
1661 * \brief Formats the endpoint and sends over AMI.
1663 * \param endpoint the endpoint to format and send
1664 * \param endpoint ami AMI variable container
1665 * \param count the number of formatters operated on
1666 * \retval 0 Success, otherwise non-zero on error
1668 int ast_sip_format_endpoint_ami(struct ast_sip_endpoint *endpoint,
1669 struct ast_sip_ami *ami, int *count);
1672 * \brief Format auth details for AMI.
1674 * \param auths an auth array
1675 * \param ami ami variable container
1676 * \retval 0 Success, non-zero on failure
1678 int ast_sip_format_auths_ami(const struct ast_sip_auth_array *auths,
1679 struct ast_sip_ami *ami);
1681 #endif /* _RES_PJSIP_H */