29467 Commits (b3590c96025c7f12c2c8ed36dd9412c806aeb11e)

 

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Author	SHA1	Message	Date
George Joseph	8170793be6	res_pjsip_outbound_authenticator_digest: Add context to log messages ... There was no context info in this module's log messages so it was impossible to toubleshoot. Added endpoint or host to all messages and added the realms in the challenge for the "No auth credentials for any realm" message. Change-Id: Ifeed2786f35fbea7d141237ae15625e472acff9b	8 years ago
Jenkins2	09cde039a3	Merge "chan_vpb.cc: Fix compile error."	8 years ago
George Joseph	07164d04a6	Merge "chan_sip: Trigger reinvite if the SDP answer is included in the SIP ACK"	8 years ago
Richard Mudgett	48566b8c66	res_sdp_translator_pjmedia.c: Add TODO notes. ... Change-Id: If27ca61f79accc882c3376d2e876d2b44aa1347b	8 years ago
Richard Mudgett	ede90e4aa5	SDP: Make SDP translation to/from internal representation more const. ... Change-Id: I473a174b869728604b37c60853896b0c458bc504	8 years ago
Richard Mudgett	5c1851cbc0	stream: Make ast_stream_topology_create_from_format_cap() allow NULL cap. ... Change-Id: Ie29760c49c25d7022ba2124698283181a0dd5d08	8 years ago
Richard Mudgett	d71c6e3bfd	SDP: Make ast_sdp_state_set_remote_sdp() return error. ... Change-Id: I7707c9d872c476d897ff459008652b35142a35e1	8 years ago
Richard Mudgett	176123e76c	SDP: Misc cleanups (Mostly memory leaks) ... Change-Id: I74431b385da333f2c5f5a6d7c55e70b69a4f05d2	8 years ago
Richard Mudgett	bad091b317	chan_vpb.cc: Fix compile error. ... Change-Id: I6d9edd34d8b2474222c86f44e379ead61e57a54f	8 years ago
Jenkins2	528e238447	Merge "channel: Add ability to request an outgoing channel with stream topology."	8 years ago
Jenkins2	16089ae1c9	Merge "frame: Better handle interpolated frames."	8 years ago
Jenkins2	066659a383	Merge "res_pjsip_session: Add cleanup to ast_sip_session_terminate"	8 years ago
Jenkins2	175297fe34	Merge "res_pjsip/res_pjsip_callerid: NULL check on caller id name string"	8 years ago
Jenkins2	49b2d1bde5	Merge "vector: defaults and indexes"	8 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	8 years ago
Joshua Colp	413de95ab2	Merge "cleanup: Fix fread() and fwrite() error handling"	8 years ago
Jenkins2	623bbcbed2	Merge "pjproject_bundled: Add --disable-libwebrtc to configure"	8 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	8 years ago
Jenkins2	54e27cad3c	Merge "res_rtp_asterisk.c: Fix crash in RTCP DTLS operation."	8 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	8 years ago
Joshua Colp	78eb08e7ba	Merge "sdp: Add support for T.38"	8 years ago
Joshua Colp	ed69471f94	Merge "SDP: Ensure SDPs "merge" properly."	8 years ago
Kevin Harwell	c6b757fa05	res_pjsip/res_pjsip_callerid: NULL check on caller id name string ... It's possible for a name in a party id structure to be marked as valid, but the name string itself be NULL (for instance this is possible to do by using the dialplan CALLERID function). There were a couple of places where the name was validated, but the string itself was not checked before passing it to functions like 'strlen'. This of course caused a crashed. This patch adds in a NULL check before attempting to pass it into a function that is not NULL tolerant. ASTERISK-25823 #close Change-Id: Iaa6ffe9d92f598fe9e3c8ae373fadbe3dfbf1d4a	8 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	8 years ago
Joshua Colp	985a5fd7aa	frame: Better handle interpolated frames. ... Interpolated frames are frames which contain a number of samples but have no actual data. Audiohooks did not handle this case when translating an incoming frame into signed linear. It assumed that a frame would always contain media when it may not. If this occurs audiohooks will now immediately return and not act on the frame. As well for users of ast_trans_frameout the function has been changed to be a bit more sane and ensure that the data pointer on a frame is set to NULL if no data is actually on the frame. This allows the various spots in Asterisk that check for an interpolated frame based on the presence of a data pointer to work as expected. ASTERISK-26926 Change-Id: I7fa22f631fa28d540722ed789ce28e84c7f8662b	8 years ago
Jenkins2	e478d2eb94	Merge "res_pjsip_sdp_rtp: No rtpmap for static RTP payload IDs in SDP."	8 years ago
Yasin CANER	99dea9ba84	res_pjsip_session : fixed wrong From Header number On Re-invite ... ASTERISK-26964 #close Change-Id: I55a9caa7dc90e6c4c219cb09b5c2ec08af84a302	8 years ago
George Joseph	858ed60446	pjproject_bundled: Add --disable-libwebrtc to configure ... Without the disable, pjproject tries to build it's internal webrtc implementation which requires sse2. This fails on platforms without sse2. ASTERISK-26930 #close Reported-by: abelbeck Change-Id: I07231f9160c35cfa42b194d3aad4e7d51fd9a410	8 years ago
George Joseph	b382919017	Merge "alembic: Add table for 'resource_list' PJSIP RLS type."	8 years ago
Thierry Magnien	585f9405b1	channels/chan_sip.c: use binding IP address for outgoing TCP SIP connections ... For outgoing TCP connections, Asterisk uses the first IP address of the interface instead of the IP address we asked him to bind to. ASTERISK-26922 #close Reported-by: Ksenia Change-Id: I43c71ca89211dbf1838e5bcdb9be8d06d98e54eb	8 years ago
Jenkins2	5a987fc5e9	Merge "res_pjsip_session.c: Send 100 Trying out earlier to prevent retransmissions."	8 years ago
George Joseph	a3a77890dc	Merge "res_hep: Add additional config initialization and validation"	8 years ago
Sean Bright	f5b67871df	cleanup: Fix fread() and fwrite() error handling ... Cleaned up some of the incorrect uses of fread() and fwrite(), mostly in the format modules. Neither of these functions will ever return a value less than 0, which we were checking for in some cases. I've introduced a fair amount of duplication in the format modules, but I plan to change how format modules work internally in a subsequent patch set, so this is simply a stop-gap. Change-Id: I8ca1cd47c20b2c0b72088bd13b9046f6977aa872	8 years ago
George Joseph	8df729517e	Merge "res_pjsip_session.c: Restructure ast_sip_session_alloc()"	8 years ago
Joshua Colp	199d4776c0	alembic: Add table for 'resource_list' PJSIP RLS type. ... This change adds an Alembic migration which adds a ps_resource_list table that can contain resource_list RLS configuration objects. ASTERISK-26929 Change-Id: I7c888fafc67b3e87012de974f71ca7a5b8b1ec05	8 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	8 years ago
Mark Michelson	32b3e36c68	SDP: Ensure SDPs "merge" properly. ... The gist of this work ensures that when a remote SDP is received, it is merged properly with the local capabilities. The remote SDP is converted into a stream topology. That topology is then merged with the current local topology on the SDP state. That new merged topology is then used to create an SDP. Finally, adjustments are made to RTP instances based on knowledge gained from the remote SDP. There are also a battery of tests in this commit that ensure that some basic SDP merges work as expected. While this may not sound like a big change, it has the property that it caused lots of ancillary changes. * The remote SDP is no longer stored on the SDP state. Biggest reason: there's no need for it. The remote SDP is used at the time it is being set and nowhere else. * Some new SDP APIs were added in order to find attributes and convert generic SDP attributes into rtpmap structures. * Writing tests made me realize that retrieving a value from an SDP options structure, the SDP options needs to be made const. * The SDP state machine was essentially gutted by a previous commit. Initially, I attempted to reinstate it, but I found that as it had been defined, it was not all that useful. What was more useful was knowing the role we play in SDP negotiation, so the SDP state machine has been transformed into an indicator of role. * Rather than storing separate local and joint stream state capabilities, it makes more sense to keep track of current stream state and update it as things change. Change-Id: I5938c2be3c6f0a003aa88a39a59e0880f8b2df3d	8 years ago
Sean Bright	0611f2ca17	res_hep: Add additional config initialization and validation ... * Initialize hepv3_runtime_data.sockfd to -1 so that our ao2 destructor does not close fd 0 * Add logging output when the required option - capture_address - is not specified. * Remove a no longer relevant #define and correct related documentation * Pass appropriate flags to aco_option_register so that capture_address cannot be the empty string. ASTERISK-26953 #close Change-Id: Ief08441bc6596d6f1718fa810e54a5048124f076	8 years ago
Sean Bright	59203c51cc	core: Use eventfd for alert pipes on Linux when possible ... The primary win of switching to eventfd when possible is that it only uses a single file descriptor while pipe() will use two. This means for each bridge channel we're reducing the number of required file descriptors by 1, and - if you're using timerfd - we also now have 1 less file descriptor per Asterisk channel. The API is not ideal (passing int arrays), but this is the cleanest approach I could come up with to maintain API/ABI. I've also removed what I believe to be an erroneous code block that checked the non-blocking flag on the pipe ends for each read. If the file descriptor is 'losing' its non-blocking mode, it is because of a bug somewhere else in our code. In my testing I haven't seen any measurable difference in performance. Change-Id: Iff0fb1573e7f7a187d5211ddc60aa8f3da3edb1d	8 years ago
George Joseph	dc6654d969	Merge "pbx: Use same thread if AST_OUTGOING_WAIT_COMPLETE specified"	8 years ago
George Joseph	cebfe85aff	Merge "rtp_engine/res_rtp_asterisk: Fix RTP struct reentrancy crashes."	8 years ago
Richard Mudgett	f1d20c84a1	res_pjsip_session.c: Send 100 Trying out earlier to prevent retransmissions. ... If ICE is enabled and a STUN server does not respond then we will block until we give up on the STUN response. This will take nine seconds. In the mean time the peer that sent the INVITE will send retransmissions. * Restructure res_pjsip_session.c:new_invite() to send a 100 Trying out earlier to prevent these retransmissions. ASTERISK-26890 Change-Id: Ie3fc611e53a0eff6586ad55e4aacad81cf6319a8	8 years ago
Richard Mudgett	835c209445	res_pjsip_session.c: Restructure ast_sip_session_alloc() ... * Restructure ast_sip_session_alloc() to need less cleanup on off nominal error paths. * Made ast_sip_session_alloc() and ast_sip_session_create_outgoing() avoid unnecessary ref manipulation to return a session. This is faster than calling a function. That function may do logging of the ref changes with REF_DEBUG enabled. Change-Id: I2a0affc4be51013d3f0485782c96b8fee3ddb00a	8 years ago
George Joseph	d373ebc428	Merge "build: Update config.guess and config.sub"	8 years ago
George Joseph	dd239e9f91	Merge "res_stun_monitor: Don't fail to load if DNS resolution fails"	8 years ago
George Joseph	b18bfb7672	Merge "make ari-stubs so doc periodic jobs can run"	8 years ago
Jean Aunis	b4b1943c5d	chan_sip: Trigger reinvite if the SDP answer is included in the SIP ACK ... Some equipments may send a re-INVITE containing an SDP in the final ACK request. If this happens in the context of direct media, the remote end should be updated with a re-INVITE. This patch queues an "update RTP peer" frame to trigger the re-INVITE, instead of the "source change" frame wich was used previously. ASTERISK-26951 Change-Id: I3644d2025f20e086ea9f8f62b486172c52b5b2e6	8 years ago
Sean Bright	c47b3e74d2	pbx: Use same thread if AST_OUTGOING_WAIT_COMPLETE specified ... Both ast_pbx_outgoing_app() and ast_pbx_outgoing_exten() cause the core to spawn a new thread to perform the dial. When AST_OUTGOING_WAIT_COMPLETE is passed to these functions, the calling thread will be blocked until the newly created channel has been hung up. After this patch, we run the dial on the current thread rather than spawning a new one. The only in-tree code that passes AST_OUTGOING_WAIT_COMPLETE is pbx_spool, so you should see reduced thread usage if you are using .call files. Change-Id: I512735d243f0a9da2bcc128f7a96dece71f2d913	8 years ago
Richard Mudgett	afad2ffd9f	res_rtp_asterisk.c: Fix crash in RTCP DTLS operation. ... Occasionally a crash happens when processing the RTCP DTLS timeout handler. The RTCP DTLS timeout timer could be left running if we have not completed the DTLS handshake before we place the call on hold or we attempt direct media. * Made ast_rtp_prop_set() stop the RTCP DTLS timer when disabling RTCP. * Made some sanity tweaks to ast_rtp_prop_set() when switching from standard RTCP mode to RTCP multiplexed mode. ASTERISK-26692 #close Change-Id: If6c64c79129961acfa4b3d63a864e8f6b664acc0	8 years ago
Richard Mudgett	d165079cbc	rtp_engine/res_rtp_asterisk: Fix RTP struct reentrancy crashes. ... The struct ast_rtp_instance has historically been indirectly protected from reentrancy issues by the channel lock because early channel drivers held the lock for really long times. Holding the channel lock for such a long time has caused many deadlock problems in the past. Along comes chan_pjsip/res_pjsip which doesn't necessarily hold the channel lock because sometimes there may not be an associated channel created yet or the channel pointer isn't available. In the case of ASTERISK-26835 a pjsip serializer thread was processing a message's SDP body while another thread was reading a RTP packet from the socket. Both threads wound up changing the rtp->rtcp->local_addr_str string and interfering with each other. The classic reentrancy problem resulted in a crash. In the case of ASTERISK-26853 a pjsip serializer thread was processing a message's SDP body while another thread was reading a RTP packet from the socket. Both threads wound up processing ICE candidates in PJPROJECT and interfering with each other. The classic reentrancy problem resulted in a crash. * rtp_engine.c: Make the ast_rtp_instance_xxx() calls lock the RTP instance struct. * rtp_engine.c: Make ICE and DTLS wrapper functions to lock the RTP instance struct for the API call. * res_rtp_asterisk.c: Lock the RTP instance to prevent a reentrancy problem with rtp->rtcp->local_addr_str in the scheduler thread running ast_rtcp_write(). * res_rtp_asterisk.c: Avoid deadlock when local RTP bridging in bridge_p2p_rtp_write() because there are two RTP instance structs involved. * res_rtp_asterisk.c: Avoid deadlock when trying to stop scheduler callbacks. We cannot hold the instance lock when trying to stop a scheduler callback. * res_rtp_asterisk.c: Remove the lock in struct dtls_details and use the struct ast_rtp_instance ao2 object lock instead. The lock was used to synchronize two threads to prevent a race condition between starting and stopping a timeout timer. The race condition is no longer present between dtls_perform_handshake() and __rtp_recvfrom() because the instance lock prevents these functions from overlapping each other with regards to the timeout timer. * res_rtp_asterisk.c: Remove the lock in struct ast_rtp and use the struct ast_rtp_instance ao2 object lock instead. The lock was used to synchronize two threads using a condition signal to know when TURN negotiations complete. * res_rtp_asterisk.c: Avoid deadlock when trying to stop the TURN ioqueue_worker_thread(). We cannot hold the instance lock when trying to create or shut down the worker thread without a risk of deadlock. This patch exposed a race condition between a PJSIP serializer thread setting up an ICE session in ice_create() and another thread reading RTP packets. * res_rtp_asterisk.c:ice_create(): Set the new rtp->ice pointer after we have re-locked the RTP instance to prevent the other thread from trying to process ICE packets on an incomplete ICE session setup. A similar race condition is between a PJSIP serializer thread resetting up an ICE session in ice_create() and the timer_worker_thread() processing the completion of the previous ICE session. * res_rtp_asterisk.c:ast_rtp_on_ice_complete(): Protect against an uninitialized/null remote_address after calling update_address_with_ice_candidate(). * res_rtp_asterisk.c: Eliminate the chance of ice_reset_session() destroying and setting the rtp->ice pointer to NULL while other threads are using it by adding an ao2 wrapper around the PJPROJECT ice pointer. Now when we have to unlock the RTP instance object to call a PJPROJECT ICE function we will hold a ref to the wrapper. Also added some rtp->ice NULL checks after we relock the RTP instance and have to do something with the ICE structure. ASTERISK-26835 #close ASTERISK-26853 #close Change-Id: I780b39ec935dcefcce880d50c1a7261744f1d1b4	8 years ago