asterisk

Commit Graph

Author	SHA1	Message	Date
Daniel Tryba	a7d977d986	res_pjsip_session: Prevent user=phone being added to anonimized URIs. Move ast_sip_add_usereqphone to be called after anonymization of URIs, to prevent the user_eq_phone adding "user=phone" to URIs containing a username that is not a phonenumber (RFC3261 19.1.1). An extra call to ast_sip_add_usereqphone on the saved version before anonymization is added to add user=phone" to the PAI. ASTERISK-27047 #close Change-Id: Ie5644bc66341b86dc08b1f7442210de2e6acdec6	9 years ago
Daniel Tryba	3a692949b7	res_pjsip: Prevent "user=phone" being added multiple times to header ast_sip_add_usereqphone adds "user=phone" to the header every time is is called without checking whether the param already exists. Preventing this by searching to string representation of header for "user=phone". ASTERISK-26988 #close Change-Id: Ib84383b07254de357dc6a98d91fc1d2c2c3719e6	9 years ago
Jenkins2	defa7084c7	Merge "res_pjsip_registrar.c: Update remove_existing AOR contact handling." into 15	9 years ago
Corey Farrell	d106e63258	res_pjproject: Fix cleanup of buildopts vector. ASTERISK-27306 Change-Id: I3bed0edf3f55b1d4adcbabb25ec14f11dc766c72	9 years ago
Jenkins2	75525ab378	Merge "res_pjsip: Fix issues that prevented shutdown of modules." into 15	9 years ago
Jenkins2	e25bf840a5	Merge "res_pjsip_session: Fix format_cap leak." into 15	9 years ago
Jenkins2	2ca39642b7	Merge "res_pjsip: Fix leak of persistent endpoint references." into 15	9 years ago
Richard Mudgett	6014a8743d	res_pjsip_registrar.c: Update remove_existing AOR contact handling. When "rewrite_contact" is enabled, the "max_contacts" count option can block re-registrations because the source port from the endpoint can be random. When the re-registration is blocked, the endpoint may give up re-registering and require manual intervention. * The "remove_existing" option now allows a registration to succeed by displacing any existing contacts that now exceed the "max_contacts" count. Any removed contacts are the next to expire. The behaviour change is beneficial when "rewrite_contact" is enabled and "max_contacts" is greater than one. The removed contact is likely the old contact created by "rewrite_contact" that the device is refreshing. ASTERISK-27192 Change-Id: I64c107a10b70db1697d17136051ae6bf22b5314b	9 years ago
Joshua Colp	5a7ca93119	Merge "res_sdp_translator_pjmedia: Fix test unregistration." into 15	9 years ago
hajekd	335c46185a	res/res_ari.c Fix: Memory leaks in ARI when using Content-Type: application/json ASTERISK-27305 Reported by: David Hajek Tested by: David Hajek Change-Id: Ife3e289062e6cf7d0e7d342dbf79ed96feff441e	9 years ago
Corey Farrell	10d0ffb0e0	res_pjsip_session: Fix format_cap leak. ASTERISK-27306 Change-Id: I2c8d3fc148f9f53715c958314e1146f9611741f3	9 years ago
Corey Farrell	0c0f6da482	res_pjsip: Fix leak of persistent endpoint references. Do not manually call sip_endpoint_apply_handler from load_all_endpoints. This is not necessary and causes memory leaks. Additionally reinitialize persistent->aors when we reuse a persistent object with a new endpoint. ASTERISK-27306 Change-Id: I59bbfc8da8a14d5f4af8c5bb1e71f8592ae823eb	9 years ago
Jenkins2	6e14b6dc3a	Merge "res_pjsip: Fix leak of fake_auth references." into 15	9 years ago
Corey Farrell	70efaf8f3e	res_sdp_translator_pjmedia: Fix test unregistration. ASTERISK-27306 Change-Id: Ib3ed47167cb697ab7bd0a56cab589893f491651b	9 years ago
Corey Farrell	4ad9ddfc75	res_pjsip: Fix leak of fake_auth references. pjsip_distributor leaks references to fake_auth when the default realm has not changed. ASTERISK-27306 Change-Id: I3fcf103b3680ad2d1d4610dcd6738eeaebf4d202	9 years ago
Daniel Tryba	ba37dd65d3	res_pjsip_caller_id chan_sip: Comply to RFC 3323 values for privacy Currently privacy requests are only granted if the Privacy header value is exactly "id" (defined in RFC 3325). It ignores any other possible value (or a combination there of). This patch reverses the logic from testing for "id" to grant privacy, to testing for "none" and granting privacy for any other value. "none" must not be used in combination with any other value (RFC 3323 section 4.2). ASTERISK-27284 #close Change-Id: If438a21f31a962da32d7a33ff33bdeb1e776fe56	9 years ago
Corey Farrell	9293d85f70	res_pjsip: Fix issues that prevented shutdown of modules. res_pjsip and res_pjsip_session had circular references, preventing both modules from shutting down. * Move session supplement registration to res_pjsip. * Use create internal functions for use by pjsip_message_filter.c. ASTERISK-27306 Change-Id: Ifbd5c19ec848010111afeab2436f9699da06ba6b	9 years ago
krells	2e1424363a	res_calendar_icalendar: Filter out occurrences superceded by another VEVENT When we are loading the calendars, we call libical's icalcomponent_foreach_recurrence method for each VEVENT component that we have in our calendar. That method has no knowledge concerning the existence of the other VEVENT components and will feed our callback with all ocurrences matching the requested time span. The occurrences generated by icalcomponent_foreach_recurrence while expanding a recurring VEVENT's RRULE and RDATE properties can be superceded by an other VEVENT sharing the same UID. I use an external iterator (in libical terminology) to avoid messing with the internal ones from the calling function, and search for VEVENTS which could supersede the current occurrence. The event which can invalidate this occurence needs to have: - the same UID as our recurrent component (comp) - a RECURRENCE-ID property, which represents the start time of this occurrence If one component is found, just clean and return. ASTERISK-27296 #close Reported by: Benoît Dereck-Tricot Change-Id: I8587ae3eaa765af7cb21eda3b6bf84e8a1c87af8	9 years ago
Jenkins2	b4e511c41b	Merge "pjsip_message_filter: Fix regression causing bad contact address" into 15	9 years ago
Jenkins2	d610cd59e6	Merge "res_stasis: Add 'video_sfu' as a requested bridge type." into 15	9 years ago
Jenkins2	88372c7844	Merge "res_pjsip_session: outgoing call did not offer all configured codecs" into 15	9 years ago
Jenkins2	c9b3b89ed6	Merge "res_rtp_asterisk.c: Fix bridge_p2p_rtp_write() reentrancy potential." into 15	9 years ago
Joshua Colp	17629e5e19	Merge "res_rtp_asterisk: Trim trailing byte off of SDES packet" into 15	9 years ago
Joshua Colp	ce08a5b3f6	res_stasis: Add 'video_sfu' as a requested bridge type. This change adds 'video_sfu' as a requested bridge type when creating a bridge. By specifying this a mixing type bridge is created that exchanges video in an SFU fashion. Change-Id: I2ada47cf5f3fc176518b647c0b4aa39d55339606	9 years ago
Richard Mudgett	7192cacfea	res_pjsip_outbound_publish.c: Fix misplaced parenthesis. The pjsip_publishc_init() call was referenced with a misplaced parentheses. As a result, outbound publication messages went out with an expiration of 1 second. ASTERISK-27298 Change-Id: I93622eabc8ee83e7a22e98c107f921284c605a08	9 years ago
George Joseph	ab15481553	pjsip_message_filter: Fix regression causing bad contact address The "res_pjsip: Filter out non SIP(S) requests" commit moved the filtering of messages to pjproject's PJSIP_MOD_PRIORITY_TRANSPORT_LAYER in order to filter out incoming bad uri schemes as early as possible. Since the change affected outgoing messages as well and the TRANSPORT layer is the last to be run on outgoing messages, we were overwriting the setting of external_signaling_address (which is set earlier by res_pjsip_nat) with an internal address. * pjsip_message_filter now registers itself as a pjproject module twice. Once in the TSX layer for the outgoing messages (as it was originally), then a second time in the TRANSPORT layer for the incoming messages to catch the invalid uri schemes. ASTERISK-27295 Reported by: Sean Bright Change-Id: I2c90190c43370f8a9d1c4693a19fd65840689c8c	9 years ago
Richard Mudgett	1eb84bdb8a	res_rtp_asterisk.c: Fix bridge_p2p_rtp_write() reentrancy potential. The bridge_p2p_rtp_write() has potential reentrancy problems. * Accessing the bridged RTP members must be done with the instance1 lock held. The DTMF and asymmetric codec checks must be split to be done with the correct RTP instance struct locked. i.e., They must be done when working on the appropriate side of the point to point bridge. * Forcing the RTP mark bit was referencing the wrong side of the point to point bridge. The set mark bit is used everywhere else to set the mark bit when sending not receiving. The patches for ASTERISK_26745 and ASTERISK_27158 did not take into account that not everything carried by RTP uses a codec. The telephony DTMF events are not exchanged with a codec. As a result when RFC2833/RFC4733 sent digits you would crash if "core set debug 1" is enabled, the DTMF digits would always get passed to the core even though the local native RTP bridge is active, and the DTMF digits would go out using the wrong SSRC id. * Add protection for non-format payload types like DTMF when updating the lastrxformat and lasttxformat. Also protect against non-format payload types when checking for asymmetric codecs. ASTERISK-27292 Change-Id: I6344ab7de21e26f84503c4d1fca1a41579364186	9 years ago
Sean Bright	a3583cabbd	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	9 years ago
Joshua Colp	4e4520854a	Merge "webrtc: Allow 'webrtc' to be set on endpoints without dtls_ca_file" into 15	9 years ago
Jenkins2	629e9bb6a0	Merge "res_pjsip_session: Reduce (and improve) SDP renegotiation." into 15	9 years ago
Sean Bright	a89f14f2d6	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	9 years ago
Kevin Harwell	3d658e3be4	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	9 years ago
Joshua Colp	132faf9922	Merge "res_pjsip_session: Don't end session when receiving a 500 on a reinvite" into 15	9 years ago
Joshua Colp	3c37ef567d	Merge "res_pjsip: Use ast_sip_is_content_type() where appropriate" into 15	9 years ago
Joshua Colp	d80b7e1604	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	9 years ago
Kevin Harwell	08440df3e3	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	9 years ago
Joshua Colp	fc2d3e7e81	Merge "res_pjsip_session/BUNDLE: Handle no audio codecs on endpoint" into 15	9 years ago
Joshua Colp	c96a5ffa6e	Merge "res_pjsip_session: Change some asserts to warning/debug messages" into 15	9 years ago
Sean Bright	9eefce503d	res_pjsip: Use ast_sip_is_content_type() where appropriate Change-Id: If3ab0d73d79ac4623308bd48508af2bfd554937d	9 years ago
George Joseph	962865d4ee	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	9 years ago
Joshua Colp	919f744fa8	Merge "res_config_pgsql: Fix removed support to previous for versions PostgreSQL 9.1" into 15	9 years ago
George Joseph	4df07df040	Merge "res_rtp_asterisk.c: Fix bundled SSRC handling." into 15	9 years ago
Joshua Colp	1fd68fb253	Merge "bridge: Change participant SFU streams when source streams change." into 15	9 years ago
Joshua Colp	95cb45cfc8	Merge "AST-2017-008: Improve RTP and RTCP packet processing." into 15	9 years ago
Joshua Colp	00c625a858	Merge "res_pjsip_session: Check for removed stream state." into 15	9 years ago
Joshua Colp	693c06f8a3	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	9 years ago
Rodrigo Ramírez Norambuena	365003a444	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	9 years ago
George Joseph	40de3a12e0	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	9 years ago
Ben Ford	90b68dd54b	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	9 years ago
Alexander Traud	f5ef14597b	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	9 years ago

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4254 Commits (6fdb992d12fe4783ae6553afa2bd9c8b3b7f27f4)