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Author	SHA1	Message	Date
Sean Bright	c9e972a26a	res_rtp_asterisk: Trim trailing byte off of SDES packet ... This could have been fixed by subtracting 1 from the final value of 'len' but the way the packet was being constructed was confusing so I took the opportunity to (I think) make it more clear. We were sending 1 extra byte at the end of the SDES RTCP packet which caused Chrome to complain (in its debug log): Too little data (1 byte) remaining in buffer to parse RTCP header (4 bytes). We now send the correct number of bytes. Change-Id: I9dcf087cdaf97da0374ae0acb7d379746a71e81b	8 years ago
Joshua Colp	c8a8933558	Merge "webrtc: Allow 'webrtc' to be set on endpoints without dtls_ca_file"	8 years ago
Joshua Colp	f9bcd3287d	Merge "channel.c: Fix invalid reference in conditionaled out code."	8 years ago
Joshua Colp	2a09b3b76a	Merge "app_queue: Only do announcement logic between ringing cycles"	8 years ago
Joshua Colp	87c4a72f16	Merge "res_pjsip_session: Reduce (and improve) SDP renegotiation."	8 years ago
Sean Bright	721947ebae	webrtc: Allow 'webrtc' to be set on endpoints without dtls_ca_file ... If using a legitimate certificate from a trusted certificate authority, you don't need to provide CA file. Change-Id: I8623973b4209b44889243716d7880274caed8a6d	8 years ago
Sean Bright	0cbeaa5589	pjproject: Patch to correct STUN FINGERPRINT usage ... Change-Id: I0e453253dff1388b0186b36c754457c1d0d12db6	8 years ago
Joshua Colp	0133d29c83	Merge "build: A few gcc 7 error fixes"	8 years ago
Kevin Harwell	b74cbadd05	res_pjsip_session: outgoing call did not offer all configured codecs ... For some scenarios when an outgoing call was made only a subset of the configured codecs were offered. If the codecs being offered happened to not have a codec supported by the phone then the call would fail. For instance Alice and Bob both are configured in Asterisk for g722 and ulaw( allow=!all,g722,ulaw). Alice's endpoint however only supports g722 while Bob's only supports ulaw. When Alice calls Bob, Alice negotiates g722 fine with Asterisk. But when Asterisk sends the outgoing offer to Bob it only contains g722 and not both g722 and ulaw, so the call ends. This patch makes it so all the audio codecs configured on the endpoint always get sent, and not just a subset. However priority is given to those codecs that are compatible with the "other side". ASTERISK-27259 #close Change-Id: Iffabc373bd94cd1dc700925dcfe406e12918c696	8 years ago
Joshua Colp	360bd11c30	Merge "res_pjsip_session: Don't end session when receiving a 500 on a reinvite"	8 years ago
Richard Mudgett	08e67f814b	channel.c: Fix invalid reference in conditionaled out code. ... ASTERISK-27289 Change-Id: I7a415948116493050614d9f4fa91ffbe0c21ec4c	8 years ago
Joshua Colp	9cb5899781	Merge "app_stream_echo: Don't echo declined streams"	8 years ago
George Joseph	4275ca16a1	build: A few gcc 7 error fixes ... Change-Id: I7b5300fbf1af7d88d47129db13ad6dbdc9b553ec	8 years ago
Joshua Colp	1b468b543d	Merge "res_pjsip: Use ast_sip_is_content_type() where appropriate"	8 years ago
StefanEng86	c3c73b3511	app_queue: Only do announcement logic between ringing cycles ... This patch reverts the change by patch 2263 from old reviewboard. Note that reverting that 2263-patch still preserves the behaviour that the commit log of the 2263-patch claimed to add. The reason for this is: The function wait_for_answer is only called from try_calling which in turn is only called from the main for loop in queue_exec, and earlier in that loop we already check the things that's removed by this patch. There's no need to check those things twice each loop iteration, and I think the proper place to check it is before each ringing cycle. By checking it in wait_for_answer, you allow the issue explained in the jira - that the head caller hears announcements while the agents' sip phones are actively ringing. Reported-by: Stefan Engström Tested-by: Stefan Engström ASTERISK-27216 #close Change-Id: Ic4290dc75256f9743900c6762ee1bb915f672db0	8 years ago
Sean Bright	0fad11f21c	app_stream_echo: Don't echo declined streams ... Discovered while experimenting with Cyber Mega Phone 2K Ultimate Dynamic Edition after accepting the audio request but declining the video one. Change-Id: Iaa86d41fccfbc1b559a30ccf740d78a3b5f8a98c	8 years ago
Joshua Colp	601e0c563f	res_pjsip_session: Reduce (and improve) SDP renegotiation. ... When pruning a request to change the topology of a channel be more intelligent about the resulting topology that is actually used for SDP renegotiation. In a case where a stream has not already been negotiated we don't need to renegotiate and offer a declined stream. This can occur if something in Asterisk (such as ConfBridge) requests to add video to a PJSIP channel that has no video codecs configured. In this case since the stream did not already exist we can safely remove the stream from the requested topology, resulting in no renegotiation occurring. In a case where a renegotiation is requested with a codec that is not supported we can reuse the formats of the existing stream if it exists to ensure that the stream continues to flow, instead of removing it. Change-Id: I636540798d55922377318fe619c510fb6ed125fb	8 years ago
Kevin Harwell	36690c26f8	res_pjsip_session: Don't end session when receiving a 500 on a reinvite ... During a reinvite, if a remote endpoint error occurs and it returns a 500 the session would end. This patch makes it so the session is not terminated, but continues as it was. The reason for this is because some endpoints may send non session terminating "server errors" like a failed codec negotiation. So in this case instead of ending the call it can hopefully continue. In the case of a real server error the session is already "doomed", will be known soon enough and appropriately ended by Asterisk later. Change-Id: Ifeedae86b8cb44b92d52c79046522ec5f0aff1d5	8 years ago
Joshua Colp	f91a905003	Merge "res_pjsip_session/BUNDLE: Handle no audio codecs on endpoint"	8 years ago
Joshua Colp	feadd654af	Merge "res_pjsip_session: Change some asserts to warning/debug messages"	8 years ago
Sean Bright	ebd0a4bebf	res_pjsip: Use ast_sip_is_content_type() where appropriate ... Change-Id: If3ab0d73d79ac4623308bd48508af2bfd554937d	8 years ago
George Joseph	6c0e13da22	res_pjsip_session/BUNDLE: Handle no audio codecs on endpoint ... When an INVITE came in with both audio and video streams but there were no audio codecs defined for the endpoint, we weren't declining the audio stream. Since it's usually the first/transport stream, when the video stream was processed and tried to use the transport, it was empty and caused a crash. We now decline the the stream if there are no matching codecs so when the video stream is processed, it's now the first/transport stream and processes normally. Change-Id: Ic854eda54c95031e66b076ecfae3041d34daa692	8 years ago
Joshua Colp	63a3d17f0b	Merge "res_rtp_asterisk.c: Fix bundled SSRC handling."	8 years ago
Joshua Colp	8b7aed0025	Merge "res_config_pgsql: Fix removed support to previous for versions PostgreSQL 9.1"	8 years ago
Joshua Colp	f77a2367e7	Merge "bridge: Change participant SFU streams when source streams change."	8 years ago
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