asterisk

Commit Graph

Author	SHA1	Message	Date
George Joseph	4a23f39e3a	res_pjsip: Add 'suppress_q850_reason_headers' option to endpoint A new option 'suppress_q850_reason_headers' has been added to the endpoint object. Some devices can't accept multiple Reason headers and get confused when both 'SIP' and 'Q.850' Reason headers are received. This option allows the 'Q.850' Reason header to be suppressed. The default value is 'no'. ASTERISK-27949 Reported-by: Ross Beer Change-Id: I54cf37a827d77de2079256bb3de7e90fa5e1deb1	8 years ago
Richard Mudgett	7b63f32649	AMI SendText action: Fix to use correct thread to send the text. The AMI action was directly sending the text to the channel driver. However, this makes two threads attempt to handle media and runs afowl of CHECK_BLOCKING. * Queue a read action to make the channel's media handling thread actually send the text message. This changes the AMI actions success/fail response to just mean the text was queued to be sent not that the text actually got sent. The channel driver may not even support sending text messages. ASTERISK-27943 Change-Id: I9dce343d8fa634ba5a416a1326d8a6340f98c379	8 years ago
Jenkins2	ce6a661386	Merge "res_pjsip_session: Add ability to accept multiple sdp answers" into 15	8 years ago
George Joseph	541d343c20	res_pjsip_session: Add ability to accept multiple sdp answers pjproject by default currently will follow media forked during an INVITE on outbound calls if the To tag is different on a subsequent response as that on an earlier response. We handle this correctly. There have been reported cases where the To tag is the same but we still need to follow the media. The pjproject patch in this commit adds the capability to sip_inv and also adds the capability to control it at runtime. The original "different tag" behavior was always controllable at runtime but we never did anything with it and left it to default to TRUE. So, along with the pjproject patch, this commit adds options to both the system and endpoint objects to control the two behaviors, and a small logic change to session_inv_on_media_update in res_pjsip_session to control the behavior at the endpoint level. The default behavior for "different tags" remains the same at TRUE and the default for "same tag" is FALSE. Change-Id: I64d071942b79adb2f0a4e13137389b19404fe3d6 ASTERISK-27936 Reported-by: Ross Beer	8 years ago
Alexander Traud	54a3340e3a	uuid: Enable UUID in Solaris 11. ASTERISK-27933 Reported by: bautsche Change-Id: I9b8362824efbfb2a16981e46e85f7c8322908c49	8 years ago
George Joseph	08b2d32e65	Merge "channel.c: Make CHECK_BLOCKING() save thread LWP id for messages." into 15	8 years ago
George Joseph	108f20a88a	Merge changes from topic 'ASTERISK-27625' into 15 * changes: channel.c: Fix usage of CHECK_BLOCKING() autoservice: Don't start channel autoservice if the thread is a user interface.	8 years ago
Jenkins2	31263f0ff6	Merge "ARI POST DTMF: Make not compete with channel's media thread." into 15	8 years ago
Jenkins2	d41b2a23e1	Merge "AMI PlayDTMF Action: Make not compete with channel's media thread." into 15	8 years ago
Kevin Harwell	6c90e0fb69	Merge "Fix some doxygen and curly placement." into 15	8 years ago
Richard Mudgett	4d598931ba	channel.c: Make CHECK_BLOCKING() save thread LWP id for messages. * Removed an unnecessary call to ast_channel_blocker_set() in __ast_read(). ASTERISK-27625 Change-Id: I342168b999984666fb869cd519fe779583a73834	8 years ago
Richard Mudgett	f82d0a9380	channel.c: Fix usage of CHECK_BLOCKING() The CHECK_BLOCKING() macro is used to indicate if a channel's handling thread is about to do a blocking operation (poll, read, or write) of media. A few operations such as ast_queue_frame(), soft hangup, and masquerades use the indication to wake up the blocked thread to reevaluate what is going on. ASTERISK-27625 Change-Id: I4dfc33e01e60627d962efa29d0a4244cf151a84d	8 years ago
Richard Mudgett	da2f32bd85	autoservice: Don't start channel autoservice if the thread is a user interface. Executing dialplan functions from either AMI or ARI by getting a variable could place the channel into autoservice. However, these user interface threads do not handle the channel's media so we wind up with two threads attempting to handle the media. There can be one and only one thread handling a channel's media at a time. Otherwise, we don't know which thread is going to handle the media frames. ASTERISK-27625 Change-Id: If2dc94ce15ddabf923ed1e2a65ea0ef56e013e49	8 years ago
Richard Mudgett	437fe5ac76	ARI POST DTMF: Make not compete with channel's media thread. There can be one and only one thread handling a channel's media at a time. Otherwise, we don't know which thread is going to handle the media frames. ASTERISK-27625 Change-Id: I4d6a2fe7386ea447ee199003bf8ad681cb30454e	8 years ago
Richard Mudgett	cf56ec739d	AMI PlayDTMF Action: Make not compete with channel's media thread. There can be one and only one thread handling a channel's media at a time. Otherwise, we don't know which thread is going to handle the media frames. ASTERISK-27625 Change-Id: Ia341f1a6f4d54f2022261abec9021fe5b2eb4905	8 years ago
Richard Mudgett	ce2aad08ee	Fix some doxygen and curly placement. Change-Id: I9a784a7c804120a8fa826c2a4cb9957e4b0b2fc8	8 years ago
Richard Mudgett	f0ebfa45b5	tcptls.h: Remove redundant SSL_CTX typedef. It is invalid to typedef something more than once. Though not all gcc compilers on different OS's complain about it. Change-Id: I5a7d4565990c985822d61ce75bde0b45f9870540	8 years ago
Alexander Traud	4b25cde9b3	res_rtp_asterisk: Allow OpenSSL configured with no-deprecated. Furthermore, allow OpenSSL configured with no-dh. Additionally, this change allows auto-negotiation of the elliptic curve/group for servers, not only with OpenSSL 1.0.2 but also with OpenSSL 1.1.0 and newer. This enables X25519 (since OpenSSL 1.1.0) and X448 (since OpenSSL 1.1.1) as a side-effect. ASTERISK-27910 Change-Id: I5b0dd47c5194ee17f830f869d629d7ef212cf537	8 years ago
Alexander Traud	60242e1418	tcptls.h: Repair ./configure --with-ssl=PATH. asterisk/tcptls.h was included (explicitly, implicitly, or transitively). Those inclusions got replaced by forward declarations. As side effect, the inclusions got completed. ASTERISK-27878 Change-Id: I9d102728e30336d6522e5e4ae9e964013a0835f7	8 years ago
Joshua Colp	eb52c70f5b	Merge "rtp: Add support for RTP extension negotiation and abs-send-time." into 15	8 years ago
Joshua Colp	d94dd27667	rtp: Add support for RTP extension negotiation and abs-send-time. When RTP was originally created it had the ability to place a single extension in an RTP packet. In practice people wanted to potentially put multiple extensions in one and so RFC 5285 (obsoleted by RFC 8285) came into existence. This allows RTP extensions to be negotiated with a unique identifier to be used in the RTP packet, allowing multiple extensions to be present in the packet. This change extends the RTP engine API to add support for this. A user of it can enable extensions and the API provides the ability to retrieve the information (to construct SDP for example) and to provide negotiated information (from SDP). The end result is that the RTP engine can then query to see if the extension has been negotiated and what unique identifier is to be used. It is then up to the RTP engine implementation to construct the packet appropriately. The first extension to use this support is abs-send-time which is defined in the REMB draft[1] and is a second timestamp placed in an RTP packet which is for when the packet has left the sending system. It is used to more accurately determine the available bandwidth. ASTERISK-27831 [1] https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03 Change-Id: I508deac557867b1e27fc7339be890c8018171588	8 years ago
Alexander Traud	ae2afa73a2	tcptls: Repair ./configure --with-ssl=PATH. SSL_OP_NO_TLSv1_1 and SSL_OP_NO_TLSv1_2 got discovered without honoring a PATH. ASTERISK-27865 Change-Id: I8cd358eed7411726d08fa7b01691bef122fbeb71	8 years ago
Jenkins2	2daabdeb03	Merge "rtp_engine: Avoid a typo error in Doxygen for ast_rtp_codecs_find_payload_code." into 15	8 years ago
Joshua Colp	e93cbb5d29	Merge "pjsip: Rewrite OPTIONS support with new eyes." into 15	8 years ago
Alexander Traud	62b80c0d3a	rtp_engine: Avoid a typo error in Doxygen for ast_rtp_codecs_find_payload_code. Change-Id: Ica089d4507a27ddfc4ce3a88d697ffbef378de48	8 years ago
Jenkins2	a3e1bf8fe8	Merge "stream: Make the topology a reference counted object." into 15	8 years ago
Joshua Colp	0ad4d2301c	stream: Make the topology a reference counted object. The stream topology has no lock of its own resulting in another lock protecting it in some way (for example the channel lock). If multiple channels are being juggled at the same time this can be problematic. This change makes the topology a reference counted object instead which guarantees it will remain valid even without the channel lock being held. Change-Id: I4f4d3dd856a033ed55fe218c3a4fab364afedb03	8 years ago
Sean Bright	0295559c4a	iostreams: Add some documentation for the ast_iostream_* functions Change-Id: Id71b87637f0a484eb5a1cd26c3d1c7c15c7dcf26	8 years ago
Jenkins2	f0c785ee25	Merge "BuildSystem: Add DragonFly BSD." into 15	8 years ago
Joshua Colp	228bc1890b	pjsip: Rewrite OPTIONS support with new eyes. The OPTIONS support in PJSIP has organically grown, like many things in Asterisk. It has been tweaked, changed, and adapted based on situations run into. Unfortunately this has taken its toll. Configuration file based objects have poor performance and even dynamic ones aren't that great. This change scraps the existing code and starts fresh with new eyes. It leverages all of the APIs made available such as sorcery observers and serializers to provide a better implementation. 1. The state of contacts, AORs, and endpoints relevant to the qualify process is maintained. This state can be updated by external forces (such as a device registering/unregistering) and also the reload process. This state also includes the association between endpoints and AORs. 2. AORs are scheduled and not contacts. This reduces the amount of work spent juggling scheduled items. 3. Manipulation of which AORs are being qualified and the endpoint states all occur within a serializer to reduce the conflict that can occur with multiple threads attempting to modify things. 4. Operations regarding an AOR use a serializer specific to that AOR. 5. AORs and endpoint state act as state compositors. They take input from lower level objects (contacts feed AORs, AORs feed endpoint state) and determine if a sufficient enough change has occurred to be fed further up the chain. 6. Realtime is supported by using observers to know when a contact has been registered. If state does not exist for the associated AOR then it is retrieved and becomes active as appropriate. The end result of all of this is best shown with a configuration file of 3000 endpoints each with an AOR that has a static contact. In the old code it would take over a minute to load and use all 8 of my cores. This new code takes 2-3 seconds and barely touches the CPU even while dealing with all of the OPTIONS requests. ASTERISK-26806 Change-Id: I6a5ebbfca9001dfe933eaeac4d3babd8d2e6f082	8 years ago
Jenkins2	d6ce4fb8cb	Merge "bridge_softmix: Forward TEXT frames" into 15	8 years ago
Joshua Colp	07a535af05	Merge "streams: Add string metadata capability" into 15	8 years ago
Alexander Traud	cabe201515	BuildSystem: Add DragonFly BSD. ASTERISK-27820 Change-Id: I310896143e94d65da1c2be3bb448204a8b86d557	8 years ago
Jenkins2	8e1420ad6d	Merge "bridge_softmix / app_confbridge: Add support for REMB combining." into 15	8 years ago
George Joseph	0455fb2062	Merge "utils: Add ast_assert_return" into 15	8 years ago
Joshua Colp	5c2120f5f5	bridge_softmix / app_confbridge: Add support for REMB combining. This change adds the ability for multiple REMB reports in bridge_softmix to be combined according to a configured behavior into a single report. This single report is sent back to the sender of video, which adjusts the encoding bitrate to be at or below the bitrate of the report. The available behaviors are: lowest, highest, and average. Lowest uses the lowest received bitrate. Highest uses the highest received bitrate. Average goes through the received bitrates adding them to the previous average and creates a new average. Other behaviors can be added in the future and the existing average one may be adjusted, but this provides the foundation to do so. Support for configuring which behavior to use has been added to app_confbridge. ASTERISK-27804 Change-Id: I9eafe4e7c1f72d67074a8d6acb26bfcf19322b66	8 years ago
Jenkins2	648b40e770	Merge "res_rtp_asterisk: Add support for receiving and handling NACK requests." into 15	8 years ago
George Joseph	8cca1501a6	bridge_softmix: Forward TEXT frames Core bridging and, more specifically, bridge_softmix have been enhanced to relay received frames of type TEXT or TEXT_DATA to all participants in a softmix bridge. res_pjsip_messaging and chan_pjsip have been enhanced to take advantage of this so when res_pjsip_messaging receives an in-dialog MESSAGE message from a user in a conference call, it's relayed to all other participants in the call. res_pjsip_messaging already queues TEXT frames to the channel when it receives an in-dialog MESSAGE from an endpoint and chan_pjsip will send an MESSAGE when it gets a TEXT frame. On a normal point-to-point call, the frames are forwarded between the two correctly. bridge_softmix was not though so messages weren't getting forwarded to conference bridge participants. Even if they were, the bridging code had no way to tell the participants who sent the message so it would look like it came from the bridge itself. * The TEXT frame type doesn't allow storage of any meta data, such as sender, on the frame so a new TEXT_DATA frame type was added that uses the new ast_msg_data structure as its payload. A channel driver can queue a frame of that type when it receives a message from outside. A channel driver can use it for sending messages by implementing the new send_text_data channel tech callback and setting the new AST_CHAN_TP_SEND_TEXT_DATA flag in its tech properties. If set, the bridging/channel core will use it instead of the original send_text callback and it will get the ast_msg_data structure. Channel drivers aren't required to implement this. Even if a TEXT_DATA enabled driver uses it for incoming messages, an outgoing channel driver that doesn't will still have it's send_text callback called with only the message text just as before. * res_pjsip_messaging now creates a TEXT_DATA frame for incoming in-dialog messages and sets the "from" to the display name in the "From" header, or if that's empty, the caller id name from the channel. This allows the chat client user to set a friendly name for the chat. * bridge_softmix now forwards TEXT and TEXT_DATA frames to all participants (except the sender). * A new function "ast_sendtext_data" was added to channel which takes an ast_msg_data structure and calls a channel's send_text_data callback, or if that's not defined, the original send_text callback. * bridge_channel now calls ast_sendtext_data for TEXT_DATA frame types and ast_sendtext for TEXT frame types. * chan_pjsip now uses the "from" name in the ast_msg_data structure (if it exists) to set the "From" header display name on outgoing text messages. Change-Id: Idacf5900bfd5f22ab8cd235aa56dfad090d18489	8 years ago
Ben Ford	8eb0622971	res_rtp_asterisk: Add support for receiving and handling NACK requests. Adds the ability to receive and handle incoming NACK requests if retransmissions are enabled. If retransmissions are enabled, a data buffer is allocated that stores packets being sent. If a NACK request is received, the packet requested for retransmission is sent if it is still in the buffer. In the same request, if any of the following 16 packets are marked as not received, those will be sent as well if available, as outlined in RFC4585. Also changes RTCP RR and SR to use media source SSRC instead of packet source SSRC when determining which instance to use for RTCP reports. For more information, refer to the wiki page: https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements ASTERISK-27806 #close Change-Id: I7f7f124af3b9d5d2fd9cffc6ba8cb48a6fff06ec	8 years ago
George Joseph	471be2b2c6	streams: Add string metadata capability Replaces the never used opaque data array. Updated stream tests to include get/set metadata and stream clone with metadata. Added stream metadata dump to "core show channel" Change-Id: Id7473aa4b374d7ab53046c20e321037ba9a56863	8 years ago
George Joseph	711fa178df	Merge "pjsip_scheduler.c: Add ability to trace scheduled tasks." into 15	8 years ago
George Joseph	00bdee4e62	utils: Add ast_assert_return Similar to pjproject's PJ_ASSERT_RETURN macro, this one will do the following... If the assert passes... NoOp If the assert fails and AST_DEVMODE is defined, execute ast_assert() then, if DO_CRASH isn't set, return from the calling function with the supplied value. If the assert fails and AST_DEVMODE is not defined, return from the calling function with the supplied value. The macro will execute a return without a value if one isn't suppled. Change-Id: I0003844affeab550d5ff5bca7aa7cf8a559b873e	8 years ago
Richard Mudgett	a7770fabd7	pjsip_scheduler.c: Add ability to trace scheduled tasks. When a scheduled task is created you can pass in the AST_SIP_SCHED_TASK_TRACK flag. This new flag causes scheduling events to be logged. Change-Id: I91967eb3d5a220915ce86881a28af772f9a7f56b	8 years ago
Richard Mudgett	571ccf92b2	res_pjsip.c: Split ast_sip_push_task_synchronous() to fit expectations. ast_sip_push_task_synchronous() did not necessarily execute the passed in task under the specified serializer. If the current thread is any registered pjsip thread then it would execute the task immediately instead of under the specified serializer. Reentrancy issues could result if the task does not execute with the right serializer. The original reason ast_sip_push_task_synchronous() checked to see if the current thread was a registered pjsip thread was because of a deadlock with masquerades and the channel technology's fixup callback (ASTERISK_22936). A subsequent masquerade deadlock fix (ASTERISK_24356) involving call pickups avoided the original deadlock situation entirely. The PJSIP channel technology's fixup callback no longer needed to call ast_sip_push_task_synchronous(). However, there are a few places where this unexpected behavior is still required to avoid deadlocks. The pjsip monitor thread executes callbacks that do calls to ast_sip_push_task_synchronous() that would deadlock if the task were actually pushed to the specified serializer. I ran into one dealing with the pubsub subscriptions where an ao2 destructor called ast_sip_push_task_synchronous(). * Split ast_sip_push_task_synchronous() into ast_sip_push_task_wait_servant() and ast_sip_push_task_wait_serializer(). ast_sip_push_task_wait_servant() has the old behavior of ast_sip_push_task_synchronous(). ast_sip_push_task_wait_serializer() has the new behavior where the task is always executed by the specified serializer or a picked serializer if one is not passed in. Both functions behave the same if the current thread is not a SIP servant. * Redirected ast_sip_push_task_synchronous() to ast_sip_push_task_wait_servant() to preserve API for released branches. ASTERISK_26806 Change-Id: Id040fa42c0e5972f4c8deef380921461d213b9f3	8 years ago
Richard Mudgett	10f4c68f70	pjsip_scheduler.c: Fix some corner cases. * Fix the periodic interval wander because it may take significant time between the sched thread queueing the task in the serializer and the serializer actually executing the task. The time it takes to actually execute the task was already taken into account. * Pass a schtd ref to the serializer when we queue a scheduled task on the serializer. We don't want it going away on us while it is in the serializer queue. * Skip the scheduled task if the task was canceled between queueing the task to the serializer and the serializer actually executing the task. * Reorder struct ast_sip_sched_task to avoid unnecessary padding. Removed task_id and added next_periodic. * Hold a ref to the passed in serializer so the serializer cannot go away on the scheduled task. ASTERISK_26806 Change-Id: I6c8046b75f6953792c8c30e55b836a4291143f24	8 years ago
Jenkins2	2cf078c475	Merge "pjsip_scheduler.c: Fix ao2 usage errors." into 15	8 years ago
Jenkins2	7f02d86c9d	Merge "res_pjsip_refer/chan_sip: Fix INVITE with replaces transfer to ConfBridge" into 15	8 years ago
Richard Mudgett	6b9933fe44	pjsip_scheduler.c: Fix ao2 usage errors. * Removed several invalid uses of OBJ_NOLOCK. These uses resulted in the 'tasks' container being accessed without a lock in a multi-threaded environment. A recipe for crashes. * Removed needlessly obtaining schtd object references. If the caller providing you a pointer to an object doesn't have a valid reference then you cannot safely get one from it. * Getting a ref to 'tasks' when you aren't copying the pointer into another location is useless. The 'tasks' container pointer is global. * Removed many unnecessary uses of RAII_VAR. * Make ast_sip_schedule_task() name parameter const. ASTERISK_26806 Change-Id: I5c62488e651314e2a1dbc01f5b078a15512d73db	8 years ago
Joshua Colp	f2a9129d8c	Merge "app_confbridge / bridge_softmix: Add ability to configure REMB interval." into 15	8 years ago
Joshua Colp	0e370c0915	Merge "res_rtp_asterisk: Queue video update on picture loss indication." into 15	8 years ago

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3908 Commits (97f111ebbbb5e4e611ed70002ff76da226a9db07)