29899 Commits (eb224fea5e02ce4073a48032a7030e129118a49c)

 

All Branches   

Search

Author	SHA1	Message	Date
Richard Mudgett	7c93982e9d	res_rtp_asterisk.c: Fix bundled SSRC handling. ... Assertions in the v15+ AST-2017-008 patches found that we were not handling the case if the incoming SDP did not specify the required SSRC attributes for bundled to work. * Be strict on matching SSRC for bundled instances including the parent instance. If the SSRC doesn't match then discard the packet. Bundled has to tell us in the SDP signaling what SSRC to expect. Otherwise, we will not know how to find the bundled instance structure. Change-Id: I152830bbff71c662408909042068fada39e617f9	8 years ago
Joshua Colp	b9f7b03c0b	Merge "AST-2017-008: Improve RTP and RTCP packet processing."	8 years ago
Jenkins2	f01cd0b409	Merge "res_config_pgsql: Add missing \n in debug log and update copyright year"	8 years ago
Joshua Colp	357a2f353b	Merge "res_pjsip_session: Check for removed stream state."	8 years ago
Joshua Colp	f2985e3106	bridge: Change participant SFU streams when source streams change. ... Some endpoints do not like a stream being reused for a new media stream. The frame/jitterbuffer can rely on underlying attributes of the media stream in order to order the packets. When a new stream takes its place without any notice the buffer can get confused and the media ends up getting dropped. This change uses the SSRC change to determine that a new source is reusing an existing stream and then bridge_softmix renegotiates each participant such that they see a new media stream. This causes the frame/jitterbuffer to start fresh and work as expected. ASTERISK-27277 Change-Id: I30ccbdba16ca073d7f31e0e59ab778c153afae07	8 years ago
Jenkins2	5ff46578aa	Merge "res_srtp: lower log level of auth failures"	8 years ago
Jenkins2	b9da3d643c	Merge "chan_sip: Expose read-only access to the full SIP INVITE Request-URI"	8 years ago
Joshua Colp	7ff464bc0a	Merge "bridge : Fix one-way direct-media when early bridging with native_rtp"	8 years ago
George Joseph	971548405b	res_pjsip_session: Change some asserts to warning/debug messages ... There was an issue reported where an SDP received on a 183 Session Progress message caused a crash because the pending streams had already been processed when the OK was received. In that case the pending topology was legitimately NULL. There was an assert for an incorrect number of streams in the topology but not one for topology being NULL. In any case, if you're not in dev-mode the asserts don't do anything and since the scenario is legit, the asserts weren't appropriate anyway. * Changed several asserts to warning or debug messages and return codes as appropriate. ASTERISK-27264 Reported by: Daniel Heckl Change-Id: I58daaa9d2938fa980857ab3ec41925ab5ff9c848	8 years ago
Rodrigo Ramírez Norambuena	cad68137a7	res_config_pgsql: Fix removed support to previous for versions PostgreSQL 9.1 ... In PostgreSQL 9.1 the backslash are string literals and not the escape of characters. In previous issue ASTERISK_26057 was fixed the use of escape LIKE but the support for old version of Postgresql than 9.1 was dropped. The sentence before make was "ESCAPE '\'" but in version before than 9.1 need it to be as follow "ESCAPE '\\'". ASTERISK-27283 Change-Id: I96d9ee1ed7693ab17503cb36a9cd72847165f949	8 years ago
Ben Ford	e666051d79	res_pjsip_session: Check for removed stream state. ... When a sip session is refreshed, the stream topology is looped through, checking each stream for compatible formats. This would cause a crash if the stream state was AST_STREAM_STATE_REMOVED, since the formats would never be set for this stream, causing a NULL value to be returned from ast_stream_get_formats. This commit adds a check for streams with removed states. Also removed a stray semicolon. Change-Id: Ic86f8b65a4a26a60885b28b8b1a0b22e1b471d42	8 years ago
George Joseph	b6aa728a58	chan_pjsip: Ignore AST_CONTROL_STREAM_TOPOLOGY_CHANGED for now ... chan_pjsip_indicate was missing a case for the recently added AST_CONTROL_STREAM_TOPOLOGY_CHANGED condition and was returning an error and causing the call to be hung up instead of just ignoring it. ASTERISK-27260 Reported by: Daniel Heckl Change-Id: I4fecbb00a0b8a853da85155065c1a6bddf235e80	8 years ago
Jean Aunis	6b7d5671d1	bridge : Fix one-way direct-media when early bridging with native_rtp ... When two channels were early bridged in a native_rtp bridge, the RTP description on one side was not updated when the other side answered. This patch forbids non-answered channels to enter a native_rtp bridge, and triggers a bridge reconfiguration when an ANSWER frame is received. ASTERISK-27257 Change-Id: If1aaee1b4ed9658a1aa91ab715ee0a6413b878df	8 years ago
Joshua Colp	8830cc0541	Merge "res_calendar: Plug memory leak and micro-optimization"	8 years ago
Jenkins2	f2fe5bb3d1	Merge "res_pjsip_pubsub: Check for Content-Type header in rx_notify_request"	8 years ago
Alexander Traud	1e4c1cec7f	res_srtp: lower log level of auth failures ... Previously, sRTP authentication failures were reported on log level WARNING. When such failures happen, each RT(C)P packet is affected, spamming the log. Now, those failures are reported at log level VERBOSE 2. Furthermore, the amount is further reduced (previously all two seconds, now all three seconds). Additionally, the new log entry informs whether media (RTP) or statistics (RTCP) are affected. ASTERISK-16898 #close Change-Id: I6c98d46b711f56e08655abeb01c951ab8e8d7fa0	8 years ago
George Joseph	b748038230	res_pjsip_pubsub: Check for Content-Type header in rx_notify_request ... pubsub_on_rx_notify_request wasn't checking for a null Content-Type header before checking that it was application/simple-message-summary. ASTERISK-27279 Reported by: Ross Beer Change-Id: Iec2a6c4d2e74af37ff779ecc9fd35644c5c4ea52	8 years ago
David J. Pryke	a5f1d58fe1	chan_sip: Expose read-only access to the full SIP INVITE Request-URI ... Provide a way to get the contents of the the Request URI from the initial SIP INVITE in dial plan function call. (In this case "${CHANNEL(ruri)}") ASTERISK-27278 Reported by: David J. Pryke Tested by: David J. Pryke Change-Id: I1dd4d6988eed1b6c98a9701e0e833a15ef0dac3e	8 years ago
Joshua Colp	6fd3db51e8	app_confbridge: Only create a channel that records audio. ... This change makes it so that the conference recorder channel that is created only contains audio formats and an audio stream. This is because the underlying application used by ConfBridge to record, MixMonitor, only allows recording audio. Having additional streams (and in particular a video stream) can result in clients needlessly renegotiating to add a video stream that will never receive video. Change-Id: I89d38aedc9205eca7741d5435e73e73bb9de97a0	8 years ago
Rodrigo Ramírez Norambuena	56f0d5fc0f	res_config_pgsql: Add missing \n in debug log and update copyright year ... Change-Id: I4ba338ecbdecc6a814a902eddc4121c8ef3cda58	8 years ago
Sean Bright	55567ee1d8	res_calendar: Plug memory leak and micro-optimization ... ast_variables_destroy is NULL safe, so there is no need to check its argument before passing it. ASTERISK-25524 #close Reported by: Jesper Change-Id: Ib0f8057642e9d471960f1a79fd42e5a3ce587d3b	8 years ago
alex	1199927fc0	cdr_mysql.c: Apply cdrzone to start and answer ... Change-Id: I7de0a5adc89824a5f2b696fc22c80fc22dff36b0	8 years ago
Richard Mudgett	087f667ab1	AST-2017-008: Improve RTP and RTCP packet processing. ... Validate RTCP packets before processing them. * Validate that the received packet is of a minimum length and apply the RFC3550 RTCP packet validation checks. * Fixed potentially reading garbage beyond the received RTCP record data. * Fixed rtp->themssrc only being set once when the remote could change the SSRC. We would effectively stop handling the RTCP statistic records. * Fixed rtp->themssrc to not treat a zero value as special by adding rtp->themssrc_valid to indicate if rtp->themssrc is available. ASTERISK-27274 Make strict RTP learning more flexible. Direct media can cause strict RTP to attempt to learn a remote address again before it has had a chance to learn the remote address the first time. Because of the rapid relearn requests, strict RTP could latch onto the first remote address and fail to latch onto the direct media remote address. As a result, you have one way audio until the call is placed on and off hold. The new algorithm learns remote addresses for a set time (1.5 seconds) before locking the remote address. In addition, we must see a configured number of remote packets from the same address in a row before switching. * Fixed strict RTP learning from always accepting the first new address packet as the new stream. * Fixed strict RTP to initialize the expected sequence number with the last received sequence number instead of the last transmitted sequence number. * Fixed the predicted next sequence number calculation in rtp_learning_rtp_seq_update() to handle overflow. ASTERISK-27252 Change-Id: Ia2d3aa6e0f22906c25971e74f10027d96525f31c	8 years ago
Jenkins2	317b62c8b4	Merge "res_pjsip: Filter out non SIP(S) requests"	8 years ago
Joshua Colp	e1837aca0c	Merge "res_calendar: Various fixes"	8 years ago
George Joseph	d178f497d2	res_pjsip: Filter out non SIP(S) requests ... Incoming requests with non sip(s) URIs in the Request, To, From or Contact URIs are now rejected with PJSIP_SC_UNSUPPORTED_URI_SCHEME (416). This is performed in pjsip_message_filter (formerly pjsip_message_ip_updater) and is done at pjproject's "TRANSPORT" layer before a request can even reach the distributor. URIs read by res_pjsip_outbound_publish from pjsip.conf are now also checked for both length and sip(s) scheme. Those URIs read by outbound registration and aor were already being checked for scheme but their error messages needed to be updated to include scheme failure as well as length failure. Change-Id: Ibb2f9f1d2dc7549da562af4cbd9156c44ffdd460	8 years ago
Jenkins2	e9455d2264	Merge "chan_rtp: Use μ-law by default instead of signed linear"	8 years ago
Joshua Colp	ba27a6508a	Merge "tcptls: Change error message to debug."	8 years ago
Joshua Colp	01f2220bec	tcptls: Change error message to debug. ... The Websocket implementation will steal the underlying stream of TCP/TLS sessions. This results in an error message being output about a stream not being present when in reality this is actually fine. This change moves it to a debug message instead. Change-Id: I66cc639080b4b4599beadb4faa7d313f2721d094	8 years ago
Sean Bright	d8112cd98b	res_calendar: Various fixes ... * The way that we were looking at XML elements for CalDAV was extremely fragile, so use SAX2 for increased robustness. * Don't complain about a 'channel' not be specified if autoreminder is not set. Assume that if 'channel' is not set, we don't want to be notified. * Fix some truncated CLI output in 'calendar show calendar' and make the 'Autoreminder' description a bit more clear ASTERISK-24588 #close Reported by: Stefan Gofferje ASTERISK-25523 #close Reported by: Jesper Change-Id: I200d11afca6a47e7d97888f286977e2e69874b2c	8 years ago
Sean Bright	eec0396395	chan_rtp: Use μ-law by default instead of signed linear ... Multicast/Unicast RTP do not use SDP so we need to use a format that cleanly maps to one of the static RTP payload types. Without this change, an Originate to a Multicast or Unicast channel without a format specified would produce no audio on the receiving device. ASTERISK-21399 #close Reported by: Tzafrir Cohen Change-Id: I97e332b566e85da04b0004b9b0daae746cfca0e3	8 years ago
George Joseph	446d48fd49	res_pjsip: Add handling for incoming unsolicited MWI NOTIFY ... A new endpoint parameter "incoming_mwi_mailbox" allows Asterisk to receive unsolicited MWI NOTIFY requests and make them available to other modules via the stasis message bus. res_pjsip_pubsub has a new handler "pubsub_on_rx_mwi_notify_request" that parses a simple-message-summary body and, if endpoint->incoming_mwi_account is set, calls ast_publish_mwi_state with the voice-message counts from the message. Change-Id: I08bae3d16e77af48fcccc2c936acce8fc0ef0f3c	8 years ago
Jenkins2	ec940f4fec	Merge "alembic: Fix typo in add_auto_info_to_endpoint_dtmf_mode"	8 years ago
Richard Mudgett	4889574ff5	res_rtp_asterisk.c: Add doxygen to RTCP payload types. ... Change-Id: I3f20ce428777cc4ce9c13b2f808d29ff8c873998	8 years ago
Joshua Colp	619008f41c	Merge "cdr_pgsql: Refactor magic number by definition for version"	8 years ago
Jenkins2	97a7760b7a	Merge "alembic: Add support for MS-SQL"	8 years ago
George Joseph	f9bad3bd61	alembic: Fix typo in add_auto_info_to_endpoint_dtmf_mode ... The downgrade function was missing "_v2" at the end of the alter column type. Change-Id: Iaa9bcef48d6f3590ce07a61342d8e66f00263d8e	8 years ago
Walter Doekes	680aba21ec	res/res_pjsip: Fix localnet checks in pjsip, part 2. ... In 45744fc53, I mistakenly broke SDP media address rewriting by misinterpreting which address was checked in the localnet comparison. Instead of checking the remote peer address to decide whether we need media address rewriting, we check our local media address: if it's local, then we rewrite. This feels awkward, but works and even made directmedia work properly if you set local_net. (For the record: for local peers, the SDP media rewrite code is not called, so the comparison does no harm there.) ASTERISK-27248 #close Change-Id: I566be1c33f4d0a689567d451ed46bab9c3861d4f	8 years ago
Rodrigo Ramírez Norambuena	c8d53a1638	cdr_pgsql: Refactor magic number by definition for version ... Change-Id: I43f25976aa3069793ddbe0086833965a6fb0a518	8 years ago
Florian Floimair	e9a81157ac	alembic: Add support for MS-SQL ... MS-SQL has no native Enum-type support and therefore needs to work with constraints. Since these constraints need unique names the suggested approach referenced in the following alembic documentation has been applied: http://bit.ly/2x9r8pb ASTERISK-27255 #close Change-Id: I8b579750dae0c549f1103ee50172644afb9b2f95	8 years ago
Jenkins2	68b506caaa	Merge "chan_sip: when getting sip pvt return failure if not found"	8 years ago
Jenkins2	e40be6c3ce	Merge "app_waitforsilence: Cleanup & don't treat missing frames as 'noise'"	8 years ago
Joshua Colp	973885df0d	Merge "res_srtp: Add support for libsrtp2.1."	8 years ago
Jenkins2	b294a48a08	Merge "chan_sip: Do not change IP address in SDP origin line (o=) in SIP reINVITE"	8 years ago
Jenkins2	bee342fdd2	Merge "res_pjsip_session: Preserve stream name during renegotiation."	8 years ago
Jenkins2	fba2f65027	Merge "func_cdr: honour 'u' flag on dummy channel"	8 years ago
Jenkins2	2f03e11b0e	Merge "stasis/control.c: Fix set_interval_hook() ref leak."	8 years ago
Jacek Konieczny	525f84bb35	func_cdr: honour 'u' flag on dummy channel ... Fixes ${CDR(...,u)} when used in cdr_custom.conf ASTERISK-27165 #close Change-Id: Ia4e0b6ba93e03d27886354c279737790e2cd6a83	8 years ago
Sean Bright	2b3f903e6f	app_waitforsilence: Cleanup & don't treat missing frames as 'noise' ... * WaitForSilence completes successfully if it receives no media in the specified timeout, but when acting as WaitForNoise that logic needs to be reversed. * Use standard argument parsing macros and add some error checking for invalid values. * The documentation indicated that the first argument to both WaitForSilence and WaitForNoise was required when it was not. Update the documentation to reflect that. * Wrap up some behavior in structs to avoid boolean checks all over the place. ASTERISK-24066 #close Reported by: M vd S Change-Id: I01d40adc5b63342bb5018a1bea2081a0aa191ef9	8 years ago
Scott Griepentrog	5553644284	chan_sip: when getting sip pvt return failure if not found ... In handle_request_invite, when processing a pickup, a call is made to get_sip_pvt_from_replaces to locate the pvt for the subscription. The pvt is assumed to be valid when zero is returned indicating no error, and is dereferenced which can cause a crash if it was not found. This change checks the not found case and returns -1 which allows the calling code to fail appropriately. ASTERISK-27217 #close Reported-by: Bryan Walters Change-Id: I6bee92b8b8b85fcac3fd66f8c00ab18bc1765612	8 years ago