3707 Commits (44a0edcf71f09258673491154601ebd0f36cc3f0)

Author	SHA1	Message	Date
Joshua Colp	861984eac0	res_pjsip: Add support for returning only reachable contacts and use it. ... This introduces the ability for PJSIP code to specify filtering flags when retrieving PJSIP contacts. The first flag for use causes the query code to only retrieve contacts that are not unreachable. This change has been leveraged by both the Dial() process and the PJSIP_DIAL_CONTACTS dialplan function so they will now only attempt calls to contacts which are not unreachable. ASTERISK-26281 Change-Id: I8233b4faa21ba3db114f5a42e946e4b191446f6c	9 years ago
Jenkins2	452e6315bb	Merge "format: Reintroduce smoother flags"	9 years ago
Joshua Colp	1a24543124	Merge "Confbridge: Add "sfu" video mode to bridge profile options."	9 years ago
Jenkins2	bb2f6234da	Merge "Add primitive SFU support to bridge_softmix."	9 years ago
Joshua Colp	97abf6d475	Merge "res_srtp: Add support for libsrtp2"	9 years ago
Jenkins2	48d047ad5a	Merge "res_pjsip: New endpoint option "refer_blind_progress""	9 years ago
Sean Bright	5c27fe2187	format: Reintroduce smoother flags ... In review 4843 (ASTERISK-24858), we added a hack that forced a smoother creation when sending signed linear so that the byte order was adjusted during transmission. This was needed because smoother flags were lost during the new format work that was done in Asterisk 13. Rather than rolling that same hack into res_rtp_multicast, re-introduce smoother flags so that formats can dictate their own options. Change-Id: I77b835fba0e539c6ce50014a984766f63cab2c16	9 years ago
Mark Michelson	39d14834f8	Confbridge: Add "sfu" video mode to bridge profile options. ... A previous commit added plumbing to bridge_softmix to allow for an SFU experience with Asterisk. This commit adds an option to app_confbridge that allows for a confbridge to actually make use of the SFU video mode. SFU mode is implemented in a "set it and forget it" kind of way. That is, when the bridge is created, if SFU mode is enabled, then the video mode gets set to SFU and cannot be changed. Future improvements may allow for a hybrid experience (e.g. forward multiple video streams, specifically those of the most recent talkers), but for this addition, no such capability is present. Change-Id: I87bbcb63dec6dbbb42488f894871b86f112b2020	9 years ago
Mark Michelson	2da869408a	Add primitive SFU support to bridge_softmix. ... This sets up the "plumbing" in bridge_softmix to be able to accommodate Asterisk asking as an SFU (selective forwarding unit) for conferences. The way this works is that whenever a channel enters or leaves a conference, all participants in the bridge get sent a stream topology change request. The topologies consist of the channels' original topology, along with video destination streams corresponding to each participants' source video streams. So for instance, if Alice, Bob, and Carol are in the conference, and each supplies one video stream, then the topologies for each would look like so: Alice: Audio, Source video(Alice), Destination Video(Bob), Destination video (Carol) Bob: Audio, Source video(Bob) Destination Video(Alice), Destination video (Carol) Carol: Audio, Source video(Carol) Destination Video(Alice), Destination video (Bob) This way, video that arrives from a source video stream can then be copied out to the destination video streams on the other participants' channels. Once the bridge gets told that a topology on a channel has changed, the bridge constructs a map in order to get the video frames routed to the proper destination streams. This is done using the bridge channel's stream_map. This change is bare-bones with regards to SFU support. Some key features are missing at this point: * Stream limits. This commit makes no effort to limit the number of streams on a specific channel. This means that if there were 50 video callers in a conference, bridge_softmix will happily send out topology change requests to every channel in the bridge, requesting 50+ streams. * Configuration. The plumbing has been added to bridge_softmix, but there has been nothing added as of yet to app_confbridge to enable SFU video mode. * Testing. Some functions included here have unit tests. However, the functionality as a whole has only been verified by hand-tracing the code. * Selectivenss. For a "selective" forwarding unit, this does not currently have any means of being selective. * Features. Presumably, someone might wish to only receive video from specific sources. There are no external-facing functions at the moment that allow for users to select who they receive video from. * Efficiency. The current scheme treats all video streams as being unidirectional. We could be re-using a source video stream as a desetnation, too. But to simplify things on this first round, I did it this way. Change-Id: I7c44a829cc63acf8b596a337b2dc3c13898a6c4d	9 years ago
Sean Bright	1f136fe885	res_srtp: Add support for libsrtp2 ... ASTERISK-25294 #close Reported by: Tzafrir Cohen ASTERISK-26976 #close Reported by: Alex Change-Id: I789b1c3d1ed31365bbd9339fa58ef36f48833c40	9 years ago
Jenkins2	56b6a71548	Merge "asterisk: Audit locking of channel when manipulating flags."	9 years ago
George Joseph	08edd54c1b	unittests: Add a unit test that causes a SEGV and... ... ...that can only be run by explicitly calling it with 'test execute category /DO_NOT_RUN/ name RAISE_SEGV' This allows us to more easily test CI and debugging tools that should do certain things when asterisk coredumps. To allow this a new member was added to the ast_test_info structure named 'explicit_only'. If set by a test, the test will be skipped during a 'test execute all' or 'test execute category ...'. Change-Id: Ia3a11856aae4887df9a02b6b081cc777b36eb6ed	9 years ago
Kevin Harwell	51375686f7	core/conversions: Added string to unsigned integer and long conversions ... Added functions that convert a string to an unsigned integer or unsigned long. A couple of unit test were also created to test the routines. The reasons for adding these conversion utilities (and hopefully eventually more) are as follows: * Conversion routines are functionally contained with consistent and better error checking * The function names offer a better description of what is happening * It encourages code reuse for easier bug fixing at a single source * It's simpler to use * It's unit testable For instance, currently in a lot of places when converting to an integer or similar the "sscanf" function is used. When using "sscanf" it may not be immediately clear what's happening as it lacks semantic naming. Limited error checking is usually done as well. For example, most of the time a check is done to make sure the value converted, but does not check for overflows or negative valued conversions when converting unsigned numbers. Why use/wrap "strtoul" and not "sscanf" then? Primarily, it lacks some of the built in error handling that "strtoul" has. For instance "strtoul" contains overflow checks. Less so, but can still factor as reasons, "sscanf" is slightly more complex in its use. And maybe a bit controversial, but it may be ("big if") potentially slower than "strtoul" in some cases. Change-Id: If7eaca4a48f8c7b89cc8b5a1f4bed2852fca82bb	9 years ago
Joshua Colp	5a7af00e80	asterisk: Audit locking of channel when manipulating flags. ... When manipulating flags on a channel the channel has to be locked to guarantee that nothing else is also manipulating the flags. This change introduces locking where necessary to guarantee this. It also adds helper functions that manipulate channel flags and lock to reduce repeated code. ASTERISK-26789 Change-Id: I489280662dba0f4c50981bfc5b5a7073fef2db10	9 years ago
George Joseph	ce4d8dac91	Merge changes from topic 'sdp_api_adjustments' ... * changes: SDP: Make process possible multiple fmtp attributes per rtpmap. SDP: Explicitly stop a RTP instance before destoying it. SDP: Rework merge_capabilities(). SDP: Update ast_get_topology_from_sdp() to keep RTP map.	9 years ago
Jenkins2	f09e079294	Merge "SDP: Add interface_address to specify our address to use."	9 years ago
Jenkins2	542dd7d795	Merge "logger: Added logger_queue_limit to the configuration options."	9 years ago
Alexei Gradinari	808f299808	res_pjsip: New endpoint option "refer_blind_progress" ... This option was added to turn off notifying the progress details on Blind Transfer. If this option is not set then the chan_pjsip will send NOTIFY "200 OK" immediately after "202 Accepted". Some SIP phones like Mitel/Aastra or Snom keep the line busy until receive "200 OK". ASTERISK-26333 #close Change-Id: Id606fbff2e02e967c02138457badc399144720f2	9 years ago
Richard Mudgett	b8659be9b0	SDP: Make process possible multiple fmtp attributes per rtpmap. ... Change-Id: Ie7511008d82b59590e0eb520a21b5e1da4bd7349	9 years ago
Richard Mudgett	16785c0908	SDP: Add interface_address to specify our address to use. ... When we optionally set the interface_address we are forcing the media to go out a specific interface address. This allows us to optionally have the media go out the interface that SIP signalling came in on or if we are configured to have the media always go out a specific address. Change-Id: I160d9fac322a075bd2557b430632544178196189	9 years ago
Richard Mudgett	367042bd3e	SDP: Explicitly stop a RTP instance before destoying it. ... * Made sdp_add_m_from_rtp_stream() and sdp_add_m_from_udptl_stream() handle generating disabled/declined streams. * Added /main/sdp/sdp_merge_asymmetric unit test. It currently does not check the offerer side negotiated SDP because that isn't the purpose of this patch and there is much to be done to handle declined/dummy streams. * Added T.38 image streams to the /main/sdp/sdp_merge_symmetric and /main/sdp/sdp_merge_crisscross unit tests. Change-Id: Ib4dcb3ca4f9a9133b376f4e3302f9a1f963f2b31	9 years ago
Richard Mudgett	ae7689f093	SDP: Update ast_get_topology_from_sdp() to keep RTP map. ... * Add failure exits to ast_get_topology_from_sdp(). Change-Id: I4cc85c1ede8d712766ed20f544dbcef04c8c1049	9 years ago
Joshua Colp	c62b5721b3	Merge "stream: ast_stream_clone() cannot copy the opaque user data."	9 years ago
George Joseph	201346fb7d	logger: Added logger_queue_limit to the configuration options. ... All log messages go to a queue serviced by a single thread which does all the IO. This setting controls how big that queue can get (and therefore how much memory is allocated) before new messages are discarded. The default is 1000. Should something go bezerk and log tons of messages in a tight loop, this will prevent memory escalation. When the limit is reached, a WARNING is logged to that effect and messages are discarded until the queue is empty again. At that time another WARNING will be logged with the count of discarded messages. There's no "low water mark" for this queue because the logger thread empties the entire queue and processes it in 1 batch before going back and waiting on the queue again. Implementing a low water mark would mean additional locking as the thread processes each message and it's not worth it. A "test" was added to test_logger.c but since the outcome is non-deterministic, it's really just a cli command, not a unit test. Change-Id: Ib4520c95e1ca5325dbf584c7989ce391649836d1	9 years ago
Joshua Colp	552e6d81ef	Merge "bridge: Fix returning to dialplan when executing Bridge() from AMI."	9 years ago
Richard Mudgett	56c5c51076	stream: ast_stream_clone() cannot copy the opaque user data. ... ast_stream_clone() cannot copy the opaque user data stored on a stream. We don't know how to clone the data so it isn't copied into the clone. Change-Id: Ia51321bf38ecbfdcc53787ca77ea5fd2cabdf367	9 years ago
Jenkins2	a20db27c56	Merge "SDP: Replace SDP telephone_event option with dtmf option"	9 years ago
Joshua Colp	c90d81ef51	bridge: Fix returning to dialplan when executing Bridge() from AMI. ... When using the Bridge AMI action on the same channel multiple times it was possible for the channel to return to the wrong location in the dialplan if the other party hung up. This happened because the priority of the channel was not preserved across each action invocation and it would fail to move on to the next priority in other cases. This change makes it so that the priority of a channel is preserved when taking control of it from another thread and it is incremented as appropriate such that the priority reflects where the channel should next be executed in the dialplan, not where it may or may not currently be. The Bridge AMI action was also changed to ensure that it too starts the channels at the next location in the dialplan. ASTERISK-24529 Change-Id: I52406669cf64208aef7252a65b63ade31fbf7a5a	9 years ago
Kevin Harwell	7b0e3b92fd	bridge_simple: Added support for streams ... This patch is the first cut at adding stream support to the bridging framework. Changes were made to the framework that allows mapping of stream topologies to a bridge's supported media types. The first channel to enter a bridge initially defines the media types for a bridge (i.e. a one to one mapping is created between the bridge and the first channel). Subsequently added channels merge their media types into the bridge's adding to it when necessary. This allows channels with different sized topologies to map correctly to each other according to media type. The bridge drops any frame that does not have a matching index into a given write stream. For now though, bridge_simple will align its two channels according to size or first to join. Once both channels join the bridge the one with the most streams will indicate to the other channel to update its streams to be the same as that of the other. If both channels have the same number of streams then the first channel to join is chosen as the stream base. A topology change source was also added to a channel when a stream toplogy change request is made. This allows subsystems to know whether or not they initiated a change request. Thus avoiding potential recursive situations. ASTERISK-26966 #close Change-Id: I1eb5987921dd80c3cdcf52accc136393ca2d4163	9 years ago
Richard Mudgett	cd272da7a8	SDP: Replace SDP telephone_event option with dtmf option ... The telephone_event option was used as a flag and a bit mapped value in different places when it is a boolean. It is also inadequate to configure the DTMF operation of the RTP instance created for the stream. Change-Id: Ib1addeaf0ce86f07039f2f979cab29405dc5239b	9 years ago
Joshua Colp	1d6429b269	Merge "SDP: Make SDP translation to/from internal representation more const."	9 years ago
Joshua Colp	090c6b702e	Merge "stream: Make ast_stream_topology_create_from_format_cap() allow NULL cap."	9 years ago
Jenkins2	9af53d3563	Merge "SDP: Make ast_sdp_state_set_remote_sdp() return error."	9 years ago
Jenkins2	74134a03bc	Merge "SDP: Misc cleanups (Mostly memory leaks)"	9 years ago
Jenkins2	94b97e0835	Merge "SDP API: Add SSRC-level attributes"	9 years ago
Richard Mudgett	ede90e4aa5	SDP: Make SDP translation to/from internal representation more const. ... Change-Id: I473a174b869728604b37c60853896b0c458bc504	9 years ago
Richard Mudgett	5c1851cbc0	stream: Make ast_stream_topology_create_from_format_cap() allow NULL cap. ... Change-Id: Ie29760c49c25d7022ba2124698283181a0dd5d08	9 years ago
Richard Mudgett	d71c6e3bfd	SDP: Make ast_sdp_state_set_remote_sdp() return error. ... Change-Id: I7707c9d872c476d897ff459008652b35142a35e1	9 years ago
Richard Mudgett	176123e76c	SDP: Misc cleanups (Mostly memory leaks) ... Change-Id: I74431b385da333f2c5f5a6d7c55e70b69a4f05d2	9 years ago
Jenkins2	528e238447	Merge "channel: Add ability to request an outgoing channel with stream topology."	9 years ago
Jenkins2	066659a383	Merge "res_pjsip_session: Add cleanup to ast_sip_session_terminate"	9 years ago
Jenkins2	49b2d1bde5	Merge "vector: defaults and indexes"	9 years ago
Mark Michelson	d6535c0080	SDP API: Add SSRC-level attributes ... RFC 5576 defines how SSRC-level attributes may be added to SDP media descriptions. In general, this is useful for grouping related SSRCes, indicating SSRC-level format attributes, and resolving collisions in RTP SSRC values. These attributes are used widely by browsers during WebRTC communications, including attributes defined by documents outside of RFC 5576. This commit introduces the addition of SSRC-level attributes into SDPs generated by Asterisk. Since Asterisk does not tend to use multiple SSRCs on a media stream, the initial support is minimal. Asterisk includes an SSRC-level CNAME attribute if configured to do so. This at least gives browsers (and possibly others) the ability to resolve SSRC collisions at offer-answer time. In order to facilitate this, the RTP engine API has been enhanced to be able to retrieve the SSRC and CNAME on a given RTP instance. res_rtp_asterisk currently does not provide meaningful CNAME values in its RTCP SDES items, and therefore it currently will always return an empty string as the CNAME value. A task in the near future will result in res_rtp_asterisk generating more meaningful CNAMEs. Change-Id: I29e7f23e7db77524f82a3b6e8531b1195ff57789	9 years ago
George Joseph	d6b2a58736	res_pjsip_session: Add cleanup to ast_sip_session_terminate ... If you use ast_request to create a PJSIP channel but then hang it up without causing a transaction to be sent, the session will never be destroyed. This is due ot the fact that it's pjproject that triggers the session cleanup when the transaction ends. app_chanisavail was doing this to get more granular channel state and it's also possible for this to happen via ARI. * ast_sip_session_terminate was modified to explicitly call the cleanup tasks and unreference session if the invite state is NULL AND invite_tsx is NULL (meaning we never sent a transaction). * chan_pjsip/hangup was modified to bump session before it calls ast_sip_session_terminate to insure that session stays valid while it does its own cleanup. * Added test events to session_destructor for a future testsuite test. ASTERISK-26908 #close Reported-by: Richard Mudgett Change-Id: I52daf6f757184e5544c261f64f6fe9602c4680a9	9 years ago
Joshua Colp	2b22c3c84b	channel: Add ability to request an outgoing channel with stream topology. ... This change extends the ast_request functionality by adding another function and callback to create an outgoing channel with a requested stream topology. Fallback is provided by either converting the requested stream topology into a format capabilities structure if the channel driver does not support streams or by converting the requested format capabilities into a stream topology if the channel driver does support streams. The Dial application has also been updated to request an outgoing channel with the stream topology of the calling channel. ASTERISK-26959 Change-Id: Ifa9037a672ac21d42dd7125aa09816dc879a70e6	9 years ago
Joshua Colp	78eb08e7ba	Merge "sdp: Add support for T.38"	9 years ago
Joshua Colp	ed69471f94	Merge "SDP: Ensure SDPs "merge" properly."	9 years ago
Kevin Harwell	cf3429b934	vector: defaults and indexes ... Added an pre-defined integer vector declaration. This makes integer vectors easier to declare and pass around. Also, added the ability to default a vector up to a given size with a default value. Lastly, added functionality that returns the "nth" index of a matching value. Also, updated a unit test to test these changes. Change-Id: Iaf4b51b2540eda57cb43f67aa59cf1d96cdbcaa5	9 years ago
Jenkins2	e478d2eb94	Merge "res_pjsip_sdp_rtp: No rtpmap for static RTP payload IDs in SDP."	9 years ago
Joshua Colp	19a79ae12c	sdp: Add support for T.38 ... This change adds a T.38 format which can be used in a stream topology to specify that a UDPTL stream needs to be created. The SDP API has been changed to understand T.38 and create the UDPTL session, add the attributes, and parse the attributes. This change does not change the boundary of the T.38 state machine. It is still up to the channel driver to implement and act on it (such as queueing control frames or reacting to them). ASTERISK-26949 Change-Id: If28956762ccb8ead562ac6c03d162d3d6014f2c7	9 years ago