6193 Commits (a507c73a784d35cd35a67efc85e44714bee9fb60)

Author	SHA1	Message	Date
Joshua Colp	a507c73a78	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
Kevin Harwell	357654313f	Merge "rtp_engine: Allow Media Formats with add_static_payload(-1) on egress again."	8 years ago
Joshua Colp	60ce5d0003	Merge "cli: Display correct unit for HTTP timeout in "manager show settings"."	8 years ago
Joshua Colp	195af35026	Merge "Fix GCC 8 build issues."	8 years ago
Alexander Traud	71d1e8d8c8	rtp_engine: Remove the double assigned RTP payload ID of H.263+. ... Mantis-3709 (Commit 68ff3c3, Asterisk 1.2) added support for the video format H.263+. For this, the RTP payload ID 103 got assigned statically. Commit f1aadc8 assigned another payload ID 98 for this format in Asterisk 1.6. Change-Id: I90e35b158487f8f1f8187da6241b54cd3b74e667	8 years ago
Corey Farrell	4722a653f4	cli: Display correct unit for HTTP timeout in "manager show settings". ... HTTP timeout is in seconds, not minutes. ASTERISK-27852 #close Change-Id: Ie6640835cb07307555741f9b559c2eb876d9343e	8 years ago
Corey Farrell	b5914d90ac	Fix GCC 8 build issues. ... This fixes build warnings found by GCC 8. In some cases format truncation is intentional so the warning is just suppressed. ASTERISK-27824 #close Change-Id: I724f146cbddba8b86619d4c4a9931ee877995c84	8 years ago
Alexander Traud	919b0eb3f2	rtp_engine: Allow Media Formats with add_static_payload(-1) on egress again. ... This issue affected only installations with rtp_use_dynamic=yes in asterisk.conf which is the default since Asterisk 15. Codec 2 and SiLK were built-in examples of media formats which were affected. ASTERISK-27850 Reported by: Dinis Brazão, Selene Feigl Change-Id: I08c1e76433a67e4350141d38cacf3a1cb5086496	8 years ago
Jaco Kroon	9f1e1d153a	manager: fix digest auth for ami/http mechanism. ... Due to a fixed size buffer the digest authentication could be incorrectly calculated if a large URI was provided, causing authentication failure. The buffer is now dynamically allocated to allow any size URI within the normal limits of the HTTP request size. ASTERISK-27841 Change-Id: I660609db13b8f9e5f9567f339dd804f4985d41b3	8 years ago
Jenkins2	d83a37f0cc	Merge "stream: Make the topology a reference counted object."	8 years ago
Jenkins2	dcaaae6cd1	Merge "iostreams: Add some documentation for the ast_iostream_* functions"	8 years ago
Joshua Colp	7528b86cad	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	069a0b7593	iostreams: Add some documentation for the ast_iostream_* functions ... Change-Id: Id71b87637f0a484eb5a1cd26c3d1c7c15c7dcf26	8 years ago
Gaurav Khurana	0827d5cc53	Add the ability to read the media file type from HTTP header for playback ... How it works today: media_cache tries to parse out the extension of the media file to be played from the URI provided to Asterisk while caching the file. What's expected: Better will be to have Asterisk get extension from other ways too. One of the common ways is to get the type of content from the CONTENT-TYPE header in the HTTP response for fetching the media file using the URI provided. Steps to Reproduce: Provide a URL of the form: http://host/media/1234 to Asterisk for media playback. It fails to play and logs show the following error line: [Sep 15 15:48:05] WARNING [29148] [C-00000092] file.c: File http://host/media/1234 does not exist in any format Scenario this issue is blocking: In the case where the media files are stored in some cloud object store, following can block the media being played via Asterisk: Cloud storage generally needs authenticated access to the storage. The way to do that is by using signed URIs. With the signed URIs there's no way to preserve the name of the file. In most cases Cloud storage returns a key to access the object and preserving file name is also not a thing there ASTERISK-27286 Reporter: Gaurav Khurana Change-Id: I1b14692a49b2c1ac67688f58757184122e92ba89	8 years ago
George Joseph	3bad41257b	Merge "BuildSystem: Add DragonFly BSD."	8 years ago
Jenkins2	8e368d0eaf	Merge "translate: generic plc not filled in after translation"	8 years ago
Jenkins2	9c430569d4	Merge "bridge_softmix: Forward TEXT frames"	9 years ago
Richard Mudgett	661fec4b59	core: Remove unused/incomplete SDP modules. ... Change-Id: Icc28fbdc46f58e54a21554e6fe8b078f841b1f86	9 years ago
Joshua Colp	1dedc73951	Merge "streams: Add string metadata capability"	9 years ago
Jenkins2	56a9338fc1	Merge "Build System: Add missing ASTMM_LIBC to flex output."	9 years ago
Kevin Harwell	ff652711c7	translate: generic plc not filled in after translation ... If during translation a codec could not handle a given frame the translation core would return NULL, thus not passing along the "missing" frame. Due to this there was no frame to apply generic plc to, thus rendering it useless. This patch makes it so the translation core produces an interpolated slin frame in the cases where an attempt was made to translate to slin, but failed. This interpolated frame is then passed along and can be used by the generic plc algorithms to fill in the frame. ASTERISK-27814 #close Change-Id: I133d084da87adef913bf2ecc9c9240e3eaf4f40a	9 years ago
Alexander Traud	efe40ff671	BuildSystem: Add DragonFly BSD. ... ASTERISK-27820 Change-Id: I310896143e94d65da1c2be3bb448204a8b86d557	9 years ago
Jenkins2	6ccf08c543	Merge "stringfields: Collect extended stringfields into the stringfield section."	9 years ago
Corey Farrell	179ae87cf4	Build System: Add missing ASTMM_LIBC to flex output. ... Redirect libc allocation functions to use Asterisk functions for main/ast_expr2f.c and res/ael/ael_lex.c. This will resolve errors produced by astmm.h when these files are regenerated, though other issues still remain. ASTERISK~27813 Change-Id: I7263e9e4217a17bde4ffaa2087a8f8aeb2a8588c	9 years ago
Joshua Colp	8de3fa2b56	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	9 years ago
George Joseph	f79a372941	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	9 years ago
George Joseph	4fb7967c73	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	9 years ago
Richard Mudgett	d50d637764	stringfields: Collect extended stringfields into the stringfield section. ... Use of extended stringfields is a temporary mechanism to avoid ABI breakage in released branches without resorting to more inconvienient methods. * Collect existing extended stringfields into the parent stringfield section of the struct. Change-Id: I8d46d037801b4518837c3ea4b6df95ceadc9436b	9 years ago
Jenkins2	fabfe701bb	Merge "res_pjsip_refer/chan_sip: Fix INVITE with replaces transfer to ConfBridge"	9 years ago
Richard Mudgett	0c03eab962	res_pjsip_refer/chan_sip: Fix INVITE with replaces transfer to ConfBridge ... There is a problem when an INVITE-with-Replaces transfer targets a channel in a ConfBridge. The transfer will unconditionally swap out the ConfBridge channel. Unfortunately, the ConfBridge state will not be aware of this change. Unexpected behavior will happen as a result since ConfBridge channels currently can only be replaced by a masquerade and not normal bridge channel moves. * We just need to pretend that the channel isn't in a bridge (like other transfer methods already do) so the transfer channel will masquerade into the ConfBridge channel. Change-Id: I209beb0e748fa4f4b92a576f36afa8f495ba4c82	9 years ago
Joshua Colp	0f6431e8e4	app_confbridge / bridge_softmix: Add ability to configure REMB interval. ... This change adds a configuration option to app_confbridge which can be used to set the interval at which we will send a combined REMB (remote estimated maximum bitrate) frame to sources of video. The bridging API has also been extended slightly to allow setting this so bridge_softmix can use it. ASTERISK-27786 Change-Id: I0e49eae60f369c86434414f3cb8278709c793c82	9 years ago
George Joseph	df7277ac5d	Merge "main: Update copyright notice with year 2018"	9 years ago
Jenkins2	c66fde8247	Merge "BuildSystem: With external editline, do not require libs for internal editline."	9 years ago
Jenkins2	0718964e32	Merge "core: Create main/options.c."	9 years ago
Kevin Harwell	48ef239a01	Merge "res_rtp_asterisk: Add support for raising additional RTCP messages."	9 years ago
Kevin Harwell	d8a2ced3ad	Merge "main/indications: Use ast_cli_completion_add for all completions."	9 years ago
Jenkins2	7b744bac60	Merge "Add data buffer API to store packets."	9 years ago
Ben Ford	138e0eff4e	Add data buffer API to store packets. ... Adds a data buffer with a configurable size that can store different kinds of packets (like RTP packets for retransmission). Given a number it will store a data packet at that position relative to the others. Given a number it will retrieve the given data packet if it is present. This is purposely a storage of arbitrary things so it can be used not just for RTP packets but also Asterisk frames in the future if needed. The API does not internally use a lock, so it will be up to the user of the API to properly protect the data buffer. For more information, refer to the wiki page: https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements Change-Id: Iff13c5d4795d52356959fe2a57360cd57dfade07	9 years ago
Joshua Colp	e14b0e960d	res_rtp_asterisk: Add support for raising additional RTCP messages. ... This change extends the existing AST_FRAME_RTCP frame type to be able to contain additional RTCP message types, such as feedback messages. The payload type is contained in the subclass which allows knowing what is in the frame itself. The RTCP feedback message type is now handled and REMB[1] messages are raised with their containing information. This also fixes a bug where all feedback messages were triggering video updates instead of just FIR and FUR. Finally RTCP frames are now passed up through the Asterisk core to what is handling the channel, mapped appropriately in the case of bridging, and written to an outgoing stream. Since RTCP frames are on a per-stream basis this is only done on multistream capable channels. [1] https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03 ASTERISK-27758 ASTERISK-26366 Change-Id: I680da0ad8d5059d5e9655d896fb9d92e9da8491e	9 years ago
Florian Floimair	455cee99ae	main: Update copyright notice with year 2018 ... Change-Id: I2d80bc5edf940fab914cba3d8a0fa0b5eb2a3148	9 years ago
Guido Falsi	48190c7f93	core: fix getopt(3) usage ... Setting optind = 0 is forced to 1 in glibc implementation, but causes option parsing to be flawed in other implementations, for example on FreeBSD. ASTERISK-27773 #close Change-Id: Ia548e69f8302e9754dbbedb6bc451c0700c66f61	9 years ago
Corey Farrell	318bf45928	main/indications: Use ast_cli_completion_add for all completions. ... Change-Id: I371be01f178fb542a9fbe8d97e7ae21aa4d82c36	9 years ago
Alexander Traud	d6fda173a4	BuildSystem: With external editline, do not require libs for internal editline. ... ASTERISK-27761 Change-Id: Ib17a7415297a210cfcdbf149e4df9b6edadbfab6	9 years ago
Corey Farrell	a6d58c518a	core: Create main/options.c. ... This creates a separate source to 'own' symbols related to options.h and paths.h. This significantly reduces the number of exports created by main/asterisk.o. This change is required to eventually be able to link unmodified Asterisk sources to utilities and/or stand-alone tests. ASTERISK~26245 Change-Id: I5cf184f4757f9363b80c9e678bdc35c477122380	9 years ago
Jenkins2	fa892d8dfd	Merge "core: Stop using AST_INLINE_API for allocator functions."	9 years ago
Joshua Colp	f03b984724	Merge "core: Remove additional symbols."	9 years ago
Jenkins2	60a06fbd1d	Merge "core: Remove dead symbols from asterisk.exports.in."	9 years ago
Jenkins2	f863f1f91b	Merge "channel.c: Allow generic plc then channel formats are equal"	9 years ago
Jenkins2	fcad222d7f	Merge "BuildSystem: Instead of $PJPROJECT_LIBS with s, use $PJPROJECT_LIB everywhere."	9 years ago
Jenkins2	b68dae189d	Merge "main/sounds: Use ast_cli_completion_add."	9 years ago