Merge the pimp_my_sip branch into trunk.

The pimp_my_sip branch is being merged at this point because
it offers basic functionality, and from an API standpoint, things
are complete.

SIP work is *not* feature-complete; however, with the completion
of the SUBSCRIBE/NOTIFY API, all APIs (except a PUBLISH API) have
been created, and thus it is possible for developers to attempt
to create new SIP work.

API documentation can be found in the doxygen in the code, but
usability documentation is still lacking.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386540 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1 files changed, 1092 insertions, 0 deletions

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 */