195 Commits (c1ab8ca74cec45730107cca3ed47fc61460365e4)

Author	SHA1	Message	Date
Richard Begg	6b7697ed48	res_pjsip_session: Enable RFC3578 overlap dialing support. ... Support for RFC3578 overlap dialling (i.e. 484 Response to partially matched destinations) as currently provided by chan_sip is missing from res_pjsip. This patch adds a new endpoint attribute (allow_overlap) [defaults to yes] which when set to yes enables 484 responses to partial destination matches rather than the current 404. ASTERISK-26864 Change-Id: Iea444da3ee7c7d4f1fde1d01d138a3d7b0fe40f6	8 years ago
George Joseph	5013d8f5d3	res_pjsip: Symmetric transports ... A new transport parameter 'symmetric_transport' has been added. When a request from a dynamic contact comes in on a transport with this option set to 'yes', the transport name will be saved and used for subsequent outgoing requests like OPTIONS, NOTIFY and INVITE. It's saved as a contact uri parameter named 'x-ast-txp' and will display with the contact uri in CLI, AMI, and ARI output. On the outgoing request, if a transport wasn't explicitly set on the endpoint AND the request URI is not a hostname, the saved transport will be used and the 'x-ast-txp' parameter stripped from the outgoing packet. * config_transport was modified to accept and store the new parameter. * config_transport/transport_apply was updated to store the transport name in the pjsip_transport->info field using the pjsip_transport->pool on UDP transports. * A 'multihomed_on_rx_message' function was added to pjsip_message_ip_updater that, for incoming requests, retrieves the transport name from pjsip_transport->info and retrieves the transport. If transport->symmetric_transport is set, an 'x-ast-txp' uri parameter containing the transport name is added to the incoming Contact header. * An 'ast_sip_get_transport_name' function was added to res_pjsip. It takes an ast_sip_endpoint and a pjsip_sip_uri and returns a transport name if endpoint->transport is set or if there's an 'x-ast-txp' parameter on the uri and the uri host is an ipv4 or ipv6 address. Otherwise it returns NULL. * An 'ast_sip_dlg_set_transport' function was added to res_pjsip which takes an ast_sip_endpoint, a pjsip_dialog, and an optional pjsip_tpselector. It calls ast_sip_get_transport_name() and if a non-NULL is returned, sets the selector and sets the transport on the dialog. If a selector was passed in, it's updated. * res_pjsip/ast_sip_create_dialog_uac and ast_sip_create_dialog_uas were modified to call ast_sip_dlg_set_transport() instead of their original logic. * res_pjsip/create_out_of_dialog_request was modified to call ast_sip_get_transport_name() and pjsip_tx_data_set_transport() instead of its original logic. * Existing transport logic was removed from endpt_send_request since that can only be called after a create_out_of_dialog_request. * res_pjsip/ast_sip_create_rdata was converted to a wrapper around a new 'ast_sip_create_rdata_with_contact' function which allows a contact_uri to be specified in addition to the existing parameters. (See below) * res_pjsip_pubsub/internal_pjsip_evsub_send_request was eliminated since all it did was transport selection and that is now done in ast_sip_create_dialog_uac and ast_sip_create_dialog_uas. * 'contact_uri' was added to subscription_persistence. This was necessary because although the parsed rdata contact header has the x-ast-txp parameter added (if appropriate), subscription_persistence_update stores the raw packet which doesn't have it. subscription_persistence_recreate was then updated to call ast_sip_create_rdata_with_contact with the persisted contact_uri so the recreated subscription has the correct transport info to send the NOTIFYs. * res_pjsip_session/internal_pjsip_inv_send_msg was eliminated since all it did was transport selection and that is now done in ast_sip_create_dialog_uac. * pjsip_message_ip_updater/multihomed_on_tx_message was updated to remove all traces of the x-ast-txp parameter from the outgoing headers. NOTE: This change does NOT modify the behavior of permanent contacts specified on an aor. To do so would require that the permanent contact's contact uri be updated with the x-ast-txp parameter and the aor sorcery object updated. If we need to persue this, we need to think about cloning permanent contacts into the same store as the dynamic ones on an aor load so they can be updated without disturbing the originally configured value. You CAN add the x-ast-txp parameter to a permanent contact's uri but it would be much simpler to just set endpoint->transport. Change-Id: I4ee1f51473da32ca54b877cd158523efcef9655f	8 years ago
Mark Michelson	10fa49e327	Add rtcp-mux support ... This commit adds support for RFC 5761: Multiplexing RTP Data and Control Packets on a Single Port. Specifically, it enables the feature when using chan_pjsip. A new option, "rtcp_mux" has been added to endpoint configuration in pjsip.conf. If set, then Asterisk will attempt to use rtcp-mux with whatever it communicates with. Asterisk follows the rules set forth in RFC 5761 with regards to falling back to standard RTCP behavior if the far end does not indicate support for rtcp-mux. The lion's share of the changes in this commit are in res_rtp_asterisk.c. This is because it was pretty much hard wired to have an RTP and an RTCP transport. The strategy used here is that when rtcp-mux is enabled, the current RTCP transport and its trappings (such as DTLS SSL session) are freed, and the RTCP session instead just mooches off the RTP session. This leads to a lot of specialized if statements throughout. ASTERISK-26732 #close Reported by Dan Jenkins Change-Id: If46a93ba1282418d2803e3fd7869374da8b77ab5	8 years ago
Joshua Colp	4e3b0cedba	res_pjsip_transport_websocket: Add support for IPv6. ... This change adds a PJSIP patch (which has been contributed upstream) to allow the registration of IPv6 transport types. Using this the res_pjsip_transport_websocket module now registers an IPv6 Websocket transport and uses it for the corresponding traffic. ASTERISK-26685 Change-Id: Id1f9126f995b31dc38db8fdb58afd289b4ad1647	8 years ago
Joshua Colp	2046743938	config: Improve documentation and behavior of outbound_proxy option. ... This change updates the documentation for the outbound_proxy option to ensure it is consistently stated that a full SIP URI must be provided for the option. The res_pjsip_outbound_registration module has also been changed so that the provided outbound_proxy value is checked to ensure it is a URI and if not an error is output stating so. ASTERISK-26782 Change-Id: I6c239a32274846fd44e65b44ad9bf6373479b593	8 years ago
George Joseph	a537dae6d0	pjproject_bundled: Update for pjproject 2.6 ... * Removed all 2.5.5 functional patches. * Updated usages of pj_release_pool to be "safe". * Updated configure options to disable webrtc. * Updated config_site.h to disable webrtc in pjmedia. * Added Richard Mudgett's recent resolver patches. Change-Id: Ib400cc4dfca68b3d07ce14d314e829bfddc252c7	8 years ago
Richard Mudgett	0b660c9989	res_pjsip: Update authentication realm documentation. ... Using the same auth section for inbound and outbound authentication is not recommended. There is a difference in meaning for an empty realm setting between inbound and outbound authentication uses. An empty inbound auth realm represents the global section's default_realm value when the authentication object is used to challenge an incoming request. An empty outgoing auth realm is treated as a don't care wildcard when the authentication object is used to respond to an incoming authentication challenge. ASTERISK-26799 Change-Id: Id3952f7cfa1b6683b9954f2c5d2352d2f11059ce	8 years ago
Richard Mudgett	07abb39d6a	res_pjsip.c: Fix inconsistency between warning and action. ... The original return value corresponded to AST_SIP_AUTHENTICATION_CHALLENGE but we have no authenticator registered to create the challenge. Change-Id: I62368180d774b497411b80fbaabd0c80841f8512	8 years ago
Mark Michelson	46147a8f30	Revert "Update qualifies when AOR configuration changes." ... This reverts commit 6492e91392. The change in question was intended to prevent the need to reload in order to update qualifies on contacts when an AOR changes. However, this ended up causing a deadlock instead. Change-Id: I1a835c90a5bb65b6dc3a1e94cddc12a4afc3d71e	8 years ago
Richard Mudgett	b47cf1a7d6	res_pjsip: Fix some off nominal tdata leaks. ... Change-Id: I243a4be5e7fbfe604923764969c4ee04eee89b9d	8 years ago
Joshua Colp	e63635b578	Merge "Update qualifies when AOR configuration changes."	8 years ago
Mark Michelson	bbed75c3ba	Update qualifies when AOR configuration changes. ... Prior to this change, qualifies would only update in the following cases: * A reload of res_pjsip.so was issued. * A dynamic contact was re-registered after its AOR's qualify_frequency had been changed This does not work well if you are using realtime for your AORs. You can update your database to have a new qualify_frequency, but the permanent contacts on that AOR will not have their qualifies updated. And the dynamic contacts on that AOR will not have their qualifies updated until the next registration, which could be a long time. This change seeks to fix this problem by making it so that whenever AOR configuration is applied, the contacts pertaining to that AOR have their qualifies updated. Additions from this patch: * AOR sorcery objects now have an apply handler that calls into a newly added function in the OPTIONS code. This causes all contacts associated with that AOR to re-schedule qualifies. * When it is time to qualify a contact, the OPTIONS code checks to see if the AOR can still be retrieved. If not, then qualification is canceled on the contact. Alterations from this patch: * The registrar code no longer updates contact's qualify_frequence and qualify_timeout. There is no point to this since those values already get updated when the AOR changes. * Reloading res_pjsip.so no longer calls the OPTIONS initialization function. Reloading res_pjsip.so results in re-loading AORs, which results in re-scheduling qualifies. Change-Id: I2e7c3316da28f389c45954f24c4e9389abac1121	8 years ago
Joshua Colp	aeea634bc0	res_pjsip: Handle invocation of callback on outgoing request when error occurs. ... There are some error cases in PJSIP when sending a request that will result in the callback for the request being invoked. The code did not handle this case and assumed on every error case that the callback was not invoked. The code has been changed to check whether the callback has been invoked and if so to absorb the error and treat it as a success. ASTERISK-26679 ASTERISK-26699 Change-Id: I563982ba204da5aa1428989a11c06dd9087fea91	8 years ago
Joshua Elson	386e3a01b3	res_pjsip: Fix known compact header issues ... ASTERISK-26684 #close Change-Id: Ifd7e401c45015119dd5e8421dbfe3afa6381744a	8 years ago
Richard Mudgett	45a5e2abc6	res_pjsip: Add/update ERROR msg if invalid URI. ... ASTERISK-24499 Change-Id: Ie305153e47e922233b2ff24715e0e326e5fa3a6c	8 years ago
George Joseph	79b09b5f18	res_pjsip_registrar: AMI Add RegistrationInboundContactStatuses command ... The PJSIPShowRegistrationsInbound AMI command was just dumping out all AORs which was pretty useless and resource heavy since it had to get all endpoints, then all aors for each endpoint, then all contacts for each aor. PJSIPShowRegistrationInboundContactStatuses sends ContactStatusDetail events which meets the intended purpose of the other command and has significantly less overhead. Also, some additional fields that were added to Contact since the original creation of the ContactStatusDetail event have been added to the end of the event. For compatibility purposes, PJSIPShowRegistrationsInbound is left intact. ASTERISK-26644 #close Change-Id: I326f12c9ecb52bf37ba03f0748749de4da01490a	9 years ago
Matt Jordan	a3f48be0da	res/res_pjsip: Fix documentation whitespace issues ... Tabs > Spaces. Change-Id: If1e43a71822615a898e958e0f8b2e882606f0bd0	9 years ago
Joshua Colp	d1739bcf07	Merge "res_pjsip.c: Rework endpt_send_request() req_wrapper code."	9 years ago
Richard Mudgett	338f35edcc	res_pjsip.c: Rework endpt_send_request() req_wrapper code. ... * Don't hold the req_wrapper lock too long in endpt_send_request(). We could block the PJSIP monitor thread if the timeout timer expires. sip_get_tpselector_from_endpoint() does a sorcery access that could take awhile accessing a database. pjsip_endpt_send_request() might take awhile if selecting a transport. * Shorten the time that the req_wrapper lock is held in the callback functions. * Simplify endpt_send_request() req_wrapper->timeout code. * Removed some redundant req_wrapper->timeout_timer->id assignments. Change-Id: I3195e3a8e0207bb8e7f49060ad2742cf21a6e4c9	9 years ago
Richard Mudgett	bb196323f9	res_pjsip: Fix tdata leaks in off nominal paths. ... Change-Id: Ie83e06e88c2d60157775263b07e40b61718ac97b	9 years ago
Joshua Colp	aed6c219a3	pjsip: Fix a few media bugs with reinvites and asymmetric payloads. ... When channel format changes occurred as a result of an RTP re-negotiation the bridge was not informed this had happened. As a result the bridge technology was not re-evaluated and the channel may have been in a bridge technology that was incompatible with its formats. The bridge is now unbridged and the technology re-evaluated when this occurs. The chan_pjsip module also allowed asymmetric codecs for sending and receiving. This did not work with all devices and caused one way audio problems. The default has been changed to NOT do this but to match the sending codec to the receiving codec. For users who want asymmetric codecs an option has been added, asymmetric_rtp_codec, which will return chan_pjsip to the previous behavior. The codecs returned by the chan_pjsip module when queried by the bridge_native_rtp module were also not reflective of the actual negotiated codecs. The nativeformats are now returned as they reflect the actual negotiated codecs. ASTERISK-26423 #close Change-Id: I6ec88c6e3912f52c334f1a26983ccb8f267020dc	9 years ago
Joshua Colp	403c4f5833	pjsip: Support dual stack automatically. ... This change adds support for dual stack automatically. No configuration is required and the IP address and version in the SIP messages and SDP will be automatically changed based on the transport over which the message is being sent. RTP usage has also been changed to listen on both IPv4 and IPv6 simultaneously to allow media to flow, and to allow ICE support on both simultaneously. This also allows failover between IPv6 and IPv4 to work as expected. ASTERISK-26309 #close Change-Id: I235a421d8f9a326606d861b449fa6fe3a030572d	9 years ago
zuul	cbd6f7001e	Merge "res_pjsip: Add ignore_uri_user_options option."	9 years ago
Joshua Colp	e3487b9360	res_pjsip: Don't assume a request will have any addresses. ... When performing DNS resolution the failover code present in res_pjsip currently assumes that a request will always have at least one viable address. In practice this is not true. A domain may be used that has no records. The code now checks that at least one address exists on the request which prevents looping. ASTERISK-26364 #close Change-Id: Ic0761b0264864acd85915c94d878a81624940f4c	9 years ago
Richard Mudgett	ba362822f3	res_pjsip: Add ignore_uri_user_options option. ... This implements the chan_sip legacy_useroption_parsing option but with a better name. * Made the caller-id number and redirecting number strings obtained from incoming SIP URI user fields always truncated at the first semicolon. People don't care about anything after the semicolon showing up on their displays even though the RFC allows the semicolon. ASTERISK-26316 #close Reported by: Kevin Harwell Change-Id: Ib42b0e940dd34d84c7b14bc2e90d1ba392624f62	9 years ago
Aaron An	2a50c29101	res/res_pjsip: Add preferred_codec_only config to pjsip endpoint. ... This patch add config to pjsip by endpoint. ;preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec ; rather than advertising all joint codec capabilities. This ; limits the other side's codec choice to exactly what we prefer. ASTERISK-26317 #close Reported by: AaronAn Tested by: AaronAn Change-Id: Iad04dc55055403bbf5ec050997aee2dadc4f0762	9 years ago
George Joseph	534063fd67	res_pjsip: Add contact_user to endpoint ... contact_user, when specified on an endpoint, will override the user portion of the Contact header on outgoing requests. Change-Id: Icd4ebfda2f2e44d3ac749d0b4066630e988407d4	9 years ago
Alexei Gradinari	403b63571c	res_pjsip_mwi: fix unsolicited mwi blocks PJSIP stack ... The PJSIP taskprocessors could be overflowed on startup if there are many (thousands) realtime endpoints configured with unsolicited mwi. The PJSIP stack could be totally unresponsive for a few minutes after boot completed. This patch creates a separate PJSIP serializers pool for mwi and makes unsolicited mwi use serializers from this pool. This patch also adds 2 new global options to tune taskprocessor alert levels: 'mwi_tps_queue_high' and 'mwi_tps_queue_low'. This patch also adds new global option 'mwi_disable_initial_unsolicited' to disable sending unsolicited mwi to all endpoints on startup. If disabled then unsolicited mwi will start processing on next endpoint's contact update. ASTERISK-26230 #close Change-Id: I4c8ecb82c249eb887930980a800c9f87f28f861a	9 years ago
Alexander Traud	3ff964c6b6	res_pjsip: SIP/SDP origin (o=) contained square brackets on IP6 transports. ... ASTERISK-26256 #close Change-Id: I3fd68df561f81fdb8c6c497d465b50c12422f058	9 years ago
Richard Mudgett	4286a369a1	res_pjsip: Whitespace and comment cleanup. ... Change-Id: I11139a4a95df34e223ba622aa6227e33ab8f6c38	9 years ago
Joshua Colp	0933f0cf96	Merge "res_pjsip: Add fax_detect_timeout endpoint option."	9 years ago
Richard Mudgett	e739888d99	res_pjsip: Add fax_detect_timeout endpoint option. ... The new endpoint option allows the PJSIP channel driver's fax_detect endpoint option to timeout on a call after the specified number of seconds into a call. The new feature is disabled if the timeout is set to zero. The option is disabled by default. ASTERISK-26214 Reported by: Richard Mudgett Change-Id: Id5a87375fb2c4f9dc1d4b44c78ec8735ba65453d	9 years ago
Corey Farrell	cf1188a1be	Unit tests: Use AST_TEST_DEFINE in conditional code only. ... If AST_TEST_DEFINE is not conditional to TEST_FRAMEWORK it produces dead code. This places all existing unit tests into a conditional block if they weren't already. ASTERISK-26211 #close Change-Id: I8ef83ee11cbc991b07b7a37ecb41433e8c734686	9 years ago
Alexei Gradinari	1c949eea6c	res_pjsip: Added "subscribe_context" to endpoint ... If specified, incoming SUBSCRIBE requests will be searched for the matching extension in the indicated context. If no "subscribe_context" is specified, then the "context" setting is used. ASTERISK-25471 #close Change-Id: I3fb7a15f5bc154079bd348c08b7ad1cdd2d5e514	9 years ago
Joshua Colp	01a8d9844b	Merge "res_pjsip.c: Register PJMEDIA error code decoder."	9 years ago
Richard Mudgett	e6e12c752c	res_pjsip.c: Register PJMEDIA error code decoder. ... Registering the PJMEDIA error codes allows errors found when parsing an incoming SDP to be easier to figure out. "Missing SDP rtpmap for dynamic payload type (PJMEDIA_SDP_EMISSINGRTPMAP)" is much easier to understand than "Unknown error 220030". ASTERISK-25772 Change-Id: I44b2dcea656fedd7593171be9e845880a2c70ca0	9 years ago
Alexei Gradinari	6fa3ed0679	res_pjsip: improve realtime performance #2 ... The patch removes updating all Endpoints' status on startup. Instead, only non-qualified aors with static contact and non-qualified non-expired contacts are retrieved from the realtime to update the endpoint status to ONLINE. The endpoint name was added to the contact object to simply find the endpoint that created this contact. The status of endpoints with qualified aors will be updated by 'qualify' functions. ASTERISK-26061 #close Change-Id: Id324c1776fa55d3741e0c5457ecac0304cb1a0df	9 years ago
Richard Mudgett	7c59f2126f	res_pjsip.c: Add check that timer actually got scheduled. ... Change-Id: Iabaa2e5dccf0762c258101ea0eb1487cf6959ad1	9 years ago
Alexei Gradinari	31f17abe44	res_pjsip: add "via_addr", "via_port", "call_id" to contact ... As res_pjsip_nat rewrites contact's address, only the last Via header can contain the source address of registered endpoint. Also Call-Id header may contain the source address of registered endpoint. Added "via_addr", "via_port", "call_id" to contact. Added new fields ViaAddress, CallID to AMI event ContactStatus. ASTERISK-26011 Change-Id: I36bcc0bf422b3e0623680152d80486aeafe4c576	9 years ago
Joshua Colp	85d0272e76	res_pjsip: Only check transaction on transaction state events. ... The send request callback function currently assumes that it will only ever be called on transaction state changes. This is not always true. If our own timer callback occurs we will call the callback with a timer event instead of a transaction state change event. In this case the transaction on the event is invalid and accessing it will result in a crash. ASTERISK-26049 #close Change-Id: I623211c8533eb73056b0250b4580b49ad4174dfc	9 years ago
Alexei Gradinari	69a85a519f	res_pjsip: Endpoint IP Access Controls ... With the old SIP module we can use IP access controls per peer. PJSIP module missing this feature. This patch added next configuration Endpoint options: "acl" - list of IP ACL section names in acl.conf "deny" - List of IP addresses to deny access from "permit" - List of IP addresses to permit access from "contact_acl" - List of Contact ACL section names in acl.conf "contact_deny" - List of Contact header addresses to deny "contact_permit" - List of Contact header addresses to permit This patch also better logging failed request: add custom message instead of "No matching endpoint found" add SIP method to logging ASTERISK-25900 Change-Id: I456dea3909d929d413864fb347d28578415ebf02	9 years ago
Alexei Gradinari	cc4c5f5693	res_pjsip: improve realtime performance ... This patch modified pjsip_options to retrieve only permament contacts for aor if the qualify_frequency is > 0 and persisted contacts if the qualify_frequency is > 0. This patch also fixed a bug in res_sorcery_astdb. res_sorcery_astdb doesn't save object data retrived from astdb. ASTERISK-25826 Change-Id: I1831fa46c4578eae5a3e574ee3362fddf08a1f05	9 years ago
Alexei Gradinari	2b1edee772	pjsip: Added "reg_server" to contacts. ... If the Asterisk system name is set in asterisk.conf, it will be stored into the "reg_server" field in the ps_contacts table to facilitate multi-server setups. ASTERISK-25931 Change-Id: Ia8f6bd2267809c78753b52bcf21835b9b59f4cb8	9 years ago
George Joseph	4ebf9a938d	res_pjsip: Add ability to identify by Authorization username ... A feature of chan_sip that service providers relied upon was the ability to identify by the Authorization username. This is most often used when customers have a PBX that needs to register rather than identify by IP address. From my own experiance, this is pretty common with small businesses who otherwise don't need a static IP. In this scenario, a register from the customer's PBX may succeed because From will usually contain the PBXs account id but an INVITE will contain the caller id. With nothing recognizable in From, the service provider's Asterisk can never match to an endpoint and the INVITE just stays unauthorized. The fixes: A new value "auth_username" has been added to endpoint/identify_by that will use the username and digest fields in the Authorization header instead of username and domain in the the From header to match an endpoint, or the To header to match an aor. This code as added to res_pjsip_endpoint_identifier_user rather than creating a new module. Although identify_by was always a comma-separated list, there was only 1 choice so order wasn't preserved. So to keep the order, a vector was added to the end of ast_sip_endpoint. This is only used by res_pjsip_registrar to find the aor. The res_pjsip_endpoint_identifier_* modules are called in globals/endpoint_identifier_order. Along the way, the logic in res_pjsip_registrar was corrected to match most-specific to least-specific as res_pjsip_endpoint_identifier_user does. The order is: username@domain username@domain_alias username Auth by username does present 1 problem however, the first INVITE won't have an Authorization header so the distributor, not finding a match on anything, sends a securty_alert. It still sends a 401 with a challenge so the next INVITE will have the Authorization header and presumably succeed. As a result though, that first security alert is actually a false alarm. To address this, a new feature has been added to pjsip_distributor that keeps track of unidentified requests and only sends the security alert if a configurable number of unidentified requests come from the same IP in a configurable amout of time. Those configuration options have been added to the global config object. This feature is only used when auth_username is enabled. Finally, default_realm was added to the globals object to replace the hard coded "asterisk" used when an endpoint is not yet identified. The testsuite tests all pass but new tests are forthcoming for this new feature. ASTERISK-25835 #close Reported-by: Ross Beer Change-Id: I30ba62d208e6f63439600916fcd1c08a365ed69d	9 years ago
Joshua Colp	d1b9b96456	Merge "res_pjsip: disable multi domain to improve realtime performace"	9 years ago
Alexei Gradinari	860b135c88	res_pjsip: disable multi domain to improve realtime performace ... This patch added new global pjsip option 'disable_multi_domain'. Disabling Multi Domain can improve Realtime performance by reducing number of database requests. ASTERISK-25930 #close Change-Id: I2e7160f3aae68475d52742107949a799aa2c7dc7	9 years ago
George Joseph	e83499df56	res_pjsip: Add serialized scheduler (res_pjsip/pjsip_scheduler.c) ... There are several places that do scheduled tasks or periodic housecleaning, each with its own implementation: * res_pjsip_keepalive has a thread that sends keepalives. * pjsip_distributor has a thread that cleans up expired unidentified requests. * res_pjsip_registrar_expire has a thread that cleans up expired contacts. * res_pjsip_pubsub uses ast_sched directly and then calls ast_sip_push_task. * res_pjsip_sdp_rtp also uses ast_sched to send keepalives. There are also places where we should be doing scheduled work but aren't. A good example are the places we have sorcery observers to start registration or qualify. These don't work when changes are made to a backend database without a pjsip reload. We need to check periodically. As a first step to solving these issues, a new ast_sip_sched facility has been created. ast_sip_sched wraps ast_sched but only uses ast_sched as a scheduled queue. When a task is ready to run, ast_sip_task_pusk is called for it. This ensures that the task is executed in a PJLIB registered thread and doesn't hold up the ast_sched thread so it can immediately continue processing the queue. The serializer used by ast_sip_sched is one of your choosing or a random one from the res_pjsip pool if you don't choose one. Another feature is the ability to automatically clean up the task_data when the task expires (if ever). If it's an ao2 object, it will be dereferenced, if it's a malloc'd object it will be freed. This is selectable when the task is scheduled. Even if you choose to not auto dereference an ao2 task data object, the scheduler itself maintains a reference to it while the task is under it's control. This prevents the data from disappearing out from under the task. There are two scheduling models. AST_SIP_SCHED_TASK_PERIODIC specifies that the invocations of the task occur at the specific interval. That is, every "interval" milliseconds, regardless of how long the task takes. If the task takes longer than the interval, it will be scheduled at the next available multiple of interval. For exmaple: If the task has an interval of 60 secs and the task takes 70 secs (it better not), the next invocation will happen at 120 seconds. AST_SIP_SCHED_TASK_DELAY specifies that the next invocation of the task should start "interval" milliseconds after the current invocation has finished. Also, the same ast_sched facility for fixed or variable intervals exists. The task's return code in conjunction with the AST_SIP_SCHED_TASK_FIXED or AST_SIP_SCHED_TASK_VARIABLE flags controls the next invocation start time. One res_pjsip.h housekeeping change was made. The pjsip header files were added to the top. There have been a few cases lately where I've needed res_pjsip.h just for ast_sip calls and had compiles fail spectacularly because I didn't add the pjsip header files to my source even though I never referenced any pjsip calls. Finally, a few new convenience APIs were added to astobj2 to make things a little easier in the scheduler. ao2_ref_and_lock() calls ao2_ref() and ao2_lock() in one go. ao2_unlock_and_unref() does the reverse. A few macros were also copied from res_phoneprov because I got tired of having to duplicate the same hash, sort and compare functions over and over again. The AO2_STRING_FIELD_(HASH|SORT|CMP)_FN macros will insert functions suitable for aor_container_alloc into your source. This facility can be used immediately for the situations where we already have a thread that wakes up periodically or do some scheduled work. For the registration and qualify issues, additional sorcery and schema changes would need to be made so that we can easily detect changed objects on a periodic basis without having to pull the entire database back to check. I'm thinking of a last-updated timestamp on the rows but more on this later. Change-Id: I7af6ad2b2d896ea68e478aa1ae201d6dd016ba1c	9 years ago
Alexei Gradinari	49813bc9e5	res_pjsip: Add headers to AMI Event ContactStatusDetail ... * Added Useragent and RegExpire headers to AMI Event ContactStatusDetail with associated documentation. ASTERISK-25903 #close Change-Id: If3d121e943e588d016ba51d4eb9c6a421a562239	9 years ago
George Joseph	e2524fcee3	res_pjsip_mwi: Add voicemail extension and mwi_subscribe_replaces_unsolicited ... res_pjsip_mwi was missing the chan_sip "vmexten" functionality which adds the Message-Account header to the MWI NOTIFY. Also, specifying mailboxes on endpoints for unsolicited mwi and on aors for subscriptions required that the admin know in advance which the client wanted. If you specified mailboxes on the endpoint, subscriptions were rejected even if you also specified mailboxes on the aor. Voicemail extension: * Added a global default_voicemail_extension which defaults to "". * Added voicemail_extension to both endpoint and aor. * Added ast_sip_subscription_get_dialog for support. * Added ast_sip_subscription_get_sip_uri for support. When an unsolicited NOTIFY is constructed, the From header is parsed, the voicemail extension from the endpoint is substituted for the user, and the result placed in the Message-Account field in the body. When a subscribed NOTIFY is constructed, the subscription dialog local uri is parsed, the voicemail_extension from the aor (looked up from the subscription resource name) is substituted for the user, and the result placed in the Message-Account field in the body. If no voicemail extension was defined, the Message-Account field is not added to the NOTIFY body. mwi_subscribe_replaces_unsolicited: * Added mwi_subscribe_replaces_unsolicited to endpoint. The previous behavior was to reject a subscribe if a previous internal subscription for unsolicited MWI was found for the mailbox. That remains the default. However, if there are mailboxes also set on the aor and the client subscribes and mwi_subscribe_replaces_unsolicited is set, the existing internal subscription is removed and replaced with the external subscription. This allows an admin to configure mailboxes on both the endpoint and aor and allows the client to select which to use. ASTERISK-25865 #close Reported-by: Ross Beer Change-Id: Ic15a9415091760539c7134a5ba3dc4a6a1217cea	9 years ago
zuul	23d2a561d5	Merge "res_pjsip/pjsip_options: Fix From generation on outgoing OPTIONS"	9 years ago