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core/ari/pjsip: Add refer mechanism
This change adds support for refers that are not session based. It includes a refer implementation for the PJSIP technology which results in out-of-dialog REFERs being sent to a PJSIP endpoint. These can be triggered using the new ARI endpoint `/endpoints/refer`. Resolves: #71 UserNote: There is a new ARI endpoint `/endpoints/refer` for referring an endpoint to some URI or endpoint.
This commit is contained in:
Maximilian Fridrich
committed by
asterisk-org-access-app[bot]
asterisk-org-access-app[bot]
parent
d16046e41f
commit
51a7b18038
@@ -339,6 +339,20 @@ struct ast_sip_nat_hook {
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void (*outgoing_external_message)(struct pjsip_tx_data *tdata, struct ast_sip_transport *transport);
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};
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/*! \brief Structure which contains information about a transport */
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struct ast_sip_request_transport_details {
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/*! \brief Type of transport */
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enum ast_transport type;
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/*! \brief Potential pointer to the transport itself, if UDP */
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pjsip_transport *transport;
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/*! \brief Potential pointer to the transport factory itself, if TCP/TLS */
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pjsip_tpfactory *factory;
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/*! \brief Local address for transport */
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pj_str_t local_address;
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/*! \brief Local port for transport */
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int local_port;
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};
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/*!
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* \brief The kind of security negotiation
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*/
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@@ -3506,18 +3520,65 @@ struct ast_threadpool *ast_sip_threadpool(void);
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* \brief Retrieve transport state
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* \since 13.7.1
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*
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* @param transport_id
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* @returns transport_state
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* \param transport_id
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* \retval transport_state
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*
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* \note ao2_cleanup(...) or ao2_ref(..., -1) must be called on the returned object
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*/
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struct ast_sip_transport_state *ast_sip_get_transport_state(const char *transport_id);
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/*!
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* \brief Return the SIP URI of the Contact header
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*
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* \param tdata
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* \retval Pointer to SIP URI of Contact
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* \retval NULL if Contact header not found or not a SIP(S) URI
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*
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* \note Do not free the returned object.
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*/
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pjsip_sip_uri *ast_sip_get_contact_sip_uri(pjsip_tx_data *tdata);
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/*!
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* \brief Returns the transport state currently in use based on request transport details
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*
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* \param details
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* \retval transport_state
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*
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* \note ao2_cleanup(...) or ao2_ref(..., -1) must be called on the returned object
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*/
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struct ast_sip_transport_state *ast_sip_find_transport_state_in_use(struct ast_sip_request_transport_details *details);
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/*!
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* \brief Sets request transport details based on tdata
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*
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* \param details pre-allocated request transport details to set
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* \param tdata
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* \param use_ipv6 if non-zero, ipv6 transports will be considered
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* \retval 0 success
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* \retval -1 failure
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*/
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int ast_sip_set_request_transport_details(struct ast_sip_request_transport_details *details, pjsip_tx_data *tdata, int use_ipv6);
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/*!
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* \brief Replace domain and port of SIP URI to point to (external) signaling address of this Asterisk instance
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*
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* \param uri
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* \param tdata
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*
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* \retval 0 success
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* \retval -1 failure
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*
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* \note Uses domain and port in Contact header if it exists, otherwise the local URI of the dialog is used if the
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* message is sent within the context of a dialog. Further, NAT settings are considered - i.e. if the target
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* is not in the localnet, the external_signaling_address and port are used.
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*/
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int ast_sip_rewrite_uri_to_local(pjsip_sip_uri *uri, pjsip_tx_data *tdata);
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/*!
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* \brief Retrieves all transport states
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* \since 13.7.1
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*
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* @returns ao2_container
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* \retval ao2_container
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*
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* \note ao2_cleanup(...) or ao2_ref(..., -1) must be called on the returned object
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*/
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@@ -3647,6 +3708,105 @@ int ast_sip_set_id_from_invite(struct pjsip_rx_data *rdata, struct ast_party_id
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void ast_sip_modify_id_header(pj_pool_t *pool, pjsip_fromto_hdr *id_hdr,
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const struct ast_party_id *id);
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/*!
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* \brief Retrieves an endpoint and URI from the "to" string.
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*
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* This URI is used as the Request URI.
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*
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* Expects the given 'to' to be in one of the following formats:
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* Why we allow so many is a mystery.
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*
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* Basic:
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*
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* endpoint : We'll get URI from the default aor/contact
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* endpoint/aor : We'll get the URI from the specific aor/contact
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* endpoint@domain : We toss the domain part and just use the endpoint
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*
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* These all use the endpoint and specified URI:
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* \verbatim
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endpoint/<sip[s]:host>
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endpoint/<sip[s]:user@host>
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endpoint/"Bob" <sip[s]:host>
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endpoint/"Bob" <sip[s]:user@host>
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endpoint/sip[s]:host
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endpoint/sip[s]:user@host
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endpoint/host
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endpoint/user@host
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\endverbatim
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*
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* These all use the default endpoint and specified URI:
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* \verbatim
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<sip[s]:host>
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<sip[s]:user@host>
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"Bob" <sip[s]:host>
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"Bob" <sip[s]:user@host>
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sip[s]:host
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sip[s]:user@host
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\endverbatim
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*
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* These use the default endpoint and specified host:
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* \verbatim
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host
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user@host
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\endverbatim
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*
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* This form is similar to a dialstring:
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* \verbatim
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PJSIP/user@endpoint
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\endverbatim
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*
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* In this case, the user will be added to the endpoint contact's URI.
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* If the contact URI already has a user, it will be replaced.
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*
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* The ones that have the sip[s] scheme are the easiest to parse.
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* The rest all have some issue.
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*
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* endpoint vs host : We have to test for endpoint first
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* endpoint/aor vs endpoint/host : We have to test for aor first
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* What if there's an aor with the same
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* name as the host?
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* endpoint@domain vs user@host : We have to test for endpoint first.
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* What if there's an endpoint with the
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* same name as the user?
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*
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* \param to 'To' field with possible endpoint
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* \param get_default_outbound If nonzero, try to retrieve the default
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* outbound endpoint if no endpoint was found.
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* Otherwise, return NULL if no endpoint was found.
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* \param uri Pointer to a char* which will be set to the URI.
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* Always must be ast_free'd by the caller - even if the return value is NULL!
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*
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* \note The logic below could probably be condensed but then it wouldn't be
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* as clear.
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*/
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struct ast_sip_endpoint *ast_sip_get_endpoint(const char *to, int get_default_outbound, char **uri);
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/*!
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* \brief Replace the To URI in the tdata with the supplied one
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*
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* \param tdata the outbound message data structure
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* \param to URI to replace the To URI with. Must be a valid SIP URI.
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*
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* \retval 0: success, -1: failure
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*/
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int ast_sip_update_to_uri(pjsip_tx_data *tdata, const char *to);
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/*!
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* \brief Overwrite fields in the outbound 'From' header
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*
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* The outbound 'From' header is created/added in ast_sip_create_request with
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* default data. If available that data may be info specified in the 'from_user'
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* and 'from_domain' options found on the endpoint. That information will be
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* overwritten with data in the given 'from' parameter.
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*
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* \param tdata the outbound message data structure
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* \param from info to copy into the header.
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* Can be either a SIP URI, or in the format user[@domain]
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*
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* \retval 0: success, -1: failure
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*/
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int ast_sip_update_from(pjsip_tx_data *tdata, char *from);
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/*!
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* \brief Retrieve the unidentified request security event thresholds
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* \since 13.8.0
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