29711 Commits (0442fcb5869df3ba9e37fac5428897d2374f7e77)

 

All Branches   

Search

Author	SHA1	Message	Date
George Joseph	0442fcb586	Merge "res_pjsip_outbound_registration.c: Re-REGISTER on transport shutdown." into 15	8 years ago
George Joseph	6a736c3c0d	Merge changes from topic 'ASTERISK-27147' into 15 ... * changes: res_pjsip: Remove ephemeral registered contacts on transport shutdown. res_pjsip: PJSIP Transport state monitor refactor.	8 years ago
Jenkins2	c5618d7c29	Merge "res_pjsip_transport_management.c: Rename some variables." into 15	8 years ago
Jenkins2	260c27ef99	Merge "STUN/netsock2: Fix some valgrind uninitialized memory findings." into 15	8 years ago
Richard Mudgett	89a92b1f75	STUN/netsock2: Fix some valgrind uninitialized memory findings. ... * netsock2.c: Test the addr->len member first as it may be the only member initialized in the struct. * stun.c:ast_stun_handle_packet(): The combinded[] local array could get used uninitialized by ast_stun_request(). The uninitialized string gets copied to another location and could overflow the destination memory buffer. These valgrind findings were found for ASTERISK_27150 but are not necessarily a fix for the issue. Change-Id: I55f8687ba4ffc0f69578fd850af006a56cbc9a57	9 years ago
Richard Mudgett	aef38f07d3	res_pjsip_outbound_registration.c: Re-REGISTER on transport shutdown. ... The fix for the issue is broken up into three parts. This is part three which handles the client side of REGISTER requests. The registered contact may no longer be valid on the server when the transport used is reliable and the connection is broken. * Re-REGISTER our contact if the reliable transport is broken after registration completes. We attempt to re-REGISTER immediately to minimize the time we are unreachable. Time may have already passed between the connection being broken and the loss being detected. * Reorder sip_outbound_registration_state_alloc() so the STATSD_GUAGE's are still correct if an allocation failure happens. ASTERISK-27147 Change-Id: I3668405b1ee75dfefb07c0d637826176f741ce83	9 years ago
Richard Mudgett	b9f6697f06	res_pjsip: Remove ephemeral registered contacts on transport shutdown. ... The fix for the issue is broken up into three parts. This is part two which handles the server side of REGISTER requests when rewrite_contact is enabled. Any registered reliable transport contact becomes invalid when the transport connection becomes disconnected. * Monitor the rewrite_contact's reliable transport REGISTER contact for shutdown. If it is shutdown then the contact must be removed because it is no longer valid. Otherwise, when the client attempts to re-REGISTER it may be blocked because the invalid contact is there. Also if we try to send a call to the endpoint using the invalid contact then the endpoint is not likely to see the request. The endpoint either won't be listening on that port for new connections or a NAT/firewall will block it. * Prune any rewrite_contact's registered reliable transport contacts on boot. The reliable transport no longer exists so the contact is invalid. * Websockets always rewrite the REGISTER contact address and the transport needs to be monitored for shutdown. * Made the websocket transport set a unique name since that is what we use as the ao2 container key. Otherwise, we would not know which transport we find when one of them shuts down. The names are also used for PJPROJECT debug logging. * Made the websocket transport post the PJSIP_TP_STATE_CONNECTED state event. Now the global keep_alive_interval option, initially idle shutdown timer, and the server REGISTER contact monitor can work on wetsocket transports. * Made the websocket transport set the PJSIP_TP_DIR_INCOMING direction. Now initially idle websockets will automatically shutdown. ASTERISK-27147 Change-Id: I397a5e7d18476830f7ffe1726adf9ee6c15964f4	9 years ago
Richard Mudgett	fe630930c9	res_pjsip_transport_management.c: Rename some variables. ... * Use monitored instead of the misleading keepalive name. Change-Id: I9e5bcbb4ab2b82d49bcd0f06dfe85d15e0b552b6	9 years ago
Richard Mudgett	0de033c9c6	res_pjsip: PJSIP Transport state monitor refactor. ... The fix for the issue is broken up into three parts. This is part one which refactors the transport state monitor code to allow more modules to be able to monitor transports. * Pull the management of PJPROJECT's transport state callback code from res_pjsip_transport_management.c into res_pjsip. Now other modules can dynamically add and remove themselves from transport monitoring without worrying about breaking PJPROJECT's callback chain. * Add the ability for other modules to get a callback whenever a specific transport is shutdown. ASTERISK-27147 Change-Id: I7d9a31371eb1487c9b7050cf82a9af5180a57912	9 years ago
Richard Mudgett	d0bc274f54	UPGRADE notes: Fixup for the 15 branch ... Change-Id: I4ca2f07ed62d77f1fdd10c3b216f6a28dd75720c	9 years ago
Scott Griepentrog	9d2844fd24	res_pjsip_messaging: IPv6 receive address needs brackets ... When handling an incoming SIP MESSAGE, PJSIP attaches the IP address that the message was received from to the message in the variable PJSIP_RECVADDR. When the IP address is IPv6 the :PORT appended results in an unparseable mess. By using an additional bit flag on the pj_sockaddr_print call, the conventional use of brackets around the address is achieved. ASTERISK-27193 #close Change-Id: I12342521f2ce87a5b6e4883d480a3fd957aa9fd9	9 years ago
Jenkins2	ca1807ed0a	Merge "Make --with-pjproject-bundled the default for Asterisk 15" into 15	9 years ago
Joshua Colp	f27710ef5b	Merge "res_rtp_asterisk: Make P2P bridge Asymmetric codec aware" into 15	9 years ago
Torrey Searle	4873e5ff33	res_rtp_asterisk: enable rtcp & QOS stats on native bridge ... Asterisk wasn't generating or forwarding RTCP packets when native bridge was activated. Also the stats weren't available via CHANNEL(qos). Now the RTCP stats are always calculated. ASTERISK-27158 #close Change-Id: I46fb8f61c95e836b9d2dda6054b0cf205c16037b	9 years ago
Torrey Searle	be8cb75dd2	res_rtp_asterisk: Make P2P bridge Asymmetric codec aware ... Introduce a new property to rtp-engine to make it aware of the desire for assymetric codecs or not. If asymmetric codecs is not allowed, the bridge will compare read/write formats and shut down the p2p bridge if needed ASTERISK-26745 #close Change-Id: I0d9c83e5356df81661e58d40a8db565833501a6f	9 years ago
Joshua Colp	709b837c35	Merge "res_pjsip_session/_sdp_rtp: Handling of 'msid' is incorrect" into 15	9 years ago
George Joseph	82f8c19a58	Make --with-pjproject-bundled the default for Asterisk 15 ... '--with-pjproject-bundled' is now the default when running ./configure. It can be disabled with '--without-pjproject-bundled'. To make building without an internet connection easier, a new ./configure option '--with-download-cache' was added that sets the cache for externals (like pjproject, the codecs and the DPMA), AND the sounds files. It can also be specified as an environment variable named "AST_DOWNLOAD_CACHE". The existing '--with-sounds-cache' option / SOUNDS_CACHE_DIR env variable and '--with-externals-cache' option / EXTERNALS_CACHE_DIR env variable remain and if specified, will override '--with-downloads-cache'. ASTERISK-27189 Change-Id: Ifa9783fddf44aafadb060c9feba713dfa81d38ce	9 years ago
Jenkins2	11d1d51844	Merge "res_pjsip_session: Release media resources on session end quicker." into 15	9 years ago
Joshua Colp	9685a704a6	res_pjsip_session: Release media resources on session end quicker. ... A change was made long ago where the session was kept around until the underlying INVITE session had been destroyed. This had the side effect of also keeping the underlying media resources around for this time as well. This change ensures that when we are told to terminate the session we immediately release any media sessions associated with it. ASTERISK-27110 Change-Id: I643e431d5c3bf05cda220c1d39e824a505a29b82	9 years ago
Jenkins2	b24854e16b	Merge "bridge: Fix stream topology/participant locking and video misrouting." into 15	9 years ago
Jenkins2	866dd2c816	Merge "channel: Fix leak on successful call to chan->tech->requester." into 15	9 years ago
Joshua Colp	027f3f665b	Merge "res_pjsip_nat.c: Remove unnecessary CMP_STOP." into 15	9 years ago
Jenkins2	f339069d09	Merge "Support GMIME 3.0" into 15	9 years ago
Joshua Colp	0e352ec510	bridge: Fix stream topology/participant locking and video misrouting. ... This change fixes a few locking issues and some video misrouting. 1. When accessing the stream topology of a channel the channel lock must be held to guarantee the topology remains valid. 2. When a channel was joined to a bridge the bridge specific implementation for stream mapping was not invoked, causing video to be misrouted for a brief period of time. ASTERISK-27182 Change-Id: I5d2f779248b84d41c5bb3896bf22ba324b336b03	9 years ago
Corey Farrell	3dbb1b9f48	channel: Fix leak on successful call to chan->tech->requester. ... joint_cap needs to be released unconditionally as chan->tech->requester does not steal the reference even on success. ASTERISK-27180 #close Change-Id: I647728992559bdb0a9c7357c20be1b36400d68b6	9 years ago
Kevin Harwell	09995fae53	res_pjsip_session/_sdp_rtp: Handling of 'msid' is incorrect ... Currently, the handling of the msid attribute is not quite right. According to the spec the msid's between the offer/answer are not dependent upon one another. Meaning the same msid's given in an offer do not have to be returned in the answer for a given stream. And they probably shouldn't be (copied/reused) since this can potentially cause some browser side confusion. This patch generates new msids when both an offer and answer are sent from Asterisk. However, Asterisk does reuse the original msid it sent out for a reinvite. Also audio+video streams are paired together by sharing the same stream id, but a different track id. ASTERISK-27179 #close Change-Id: Ifaec06dc7e65ad841633a24ebec8c8a9302d6643	9 years ago
Corey Farrell	418e062ce5	app_privacy: remove unused header asterisk/image.h ... Change-Id: I56ed530633a642633b18383821069e806c92ae82	9 years ago
Jenkins2	2014a695f7	Merge "alembic/res_pjsip: Add "webrtc" configuration option" into 15	9 years ago
Joshua Colp	3517551b3a	Merge "Fix compile error for old versions of GCC." into 15	9 years ago
Jenkins2	3ac2da8974	Merge "Correct some leaks in unit tests." into 15	9 years ago
Jenkins2	4c7f81af6e	Merge "res_pjsip_transport_websocket.c: Fix serializer ref leak." into 15	9 years ago
Jenkins2	230cf2ee59	Merge "res_pjsip_outbound_registration.c: Misc fixes." into 15	9 years ago
Tzafrir Cohen	0cbfb95b7f	Support GMIME 3.0 ... Support building the Asterisk httpd with version 3.0 of gmime as well as earlier versions of that library. ASTERISK-27173 Change-Id: I7e13dd05a3083ccb0df2dabf83110223f6a9fa8f	9 years ago
Corey Farrell	1a3e08326e	Correct some leaks in unit tests. ... * chan_sip: channel in test_sip_rtpqos_1. * test_config: config hook, config info and global config holder. * test_core_format: format in format_attribute_set_without_interface. * test_stream: unneeded frame duplication. * test_taskprocessor: task_data. Change-Id: I94d364d195cf3b3b5de2bf3ad565343275c7ad31	9 years ago
Richard Mudgett	563264c9b5	res_pjsip_transport_websocket.c: Fix serializer ref leak. ... Change-Id: Ib5a19bfd597f63d9021baeb645fc11153b3afa57	9 years ago
Richard Mudgett	347e148aa0	res_pjsip_outbound_registration.c: Misc fixes. ... * Remove unnecessary CMP_STOP. * In handle_client_registration() use DEBUG_ATLEAST() to only do work needed for the debug log message when the debug log message is needed. * In sip_outbound_registration_state_destroy() check state->registration for NULL. Change-Id: I656d0fa11dda0b00048103efb1558e67a426fd80	9 years ago
Richard Mudgett	db6b883087	res_pjsip_nat.c: Remove unnecessary CMP_STOP. ... Change-Id: I6279b0d723bc3b75b8d65e81e02da9ea9bc0c3da	9 years ago
Richard Mudgett	2cd7bec061	res_pjsip_registrar.c: Remove unnecessary CMP_STOP. ... Most uses of CMP_STOP are superfluous and are only respected when OBJ_MULTIPLE is used to search the container. Change-Id: I20571a202ec0aa1098bb2749eeba18de7ca110b8	9 years ago
Kevin Harwell	b0c016cf6e	alembic/res_pjsip: Add "webrtc" configuration option ... When the "webrtc" option was added in res_pjsip it was not added to the alembic scripts. This patch adds the option for alembic. Also, changed the sorcery configuration type to an OPT_YESNO_T value instead of an OPT_BOOL_T so if this field is ever written to a database it will write out the correct value. ASTERISK-27119 #close Change-Id: I3e199f060aea25e193c439fc5cf96be4d3ed1c7b	9 years ago
Corey Farrell	cd79a15b2f	Fix compile error for old versions of GCC. ... Use -Wno-format-truncation only if supported by compiler. ASTERISK-27171 #close Change-Id: Iac0aed7a5bcaa16c21b7d62c4e4678d244c4ccb6	9 years ago
George Joseph	6edefc9480	Revert "res_pjsip_session: Release media resources on session end quicker." ... This reverts commit 6c374d863c420d2519a52fffba3de5f935a30cdb. Change-Id: Ia89836005df66ef540885afd6f171886de0db442	9 years ago
Joshua Colp	8f51443a08	res_pjsip_session: Release media resources on session end quicker. ... A change was made long ago where the session was kept around until the underlying INVITE session had been destroyed. This had the side effect of also keeping the underlying media resources around for this time as well. This change ensures that when we know the session is ending we release the media resources immediately. ASTERISK-27110 Change-Id: I3c6a82fe7d2c50b9dc9197cb12ef22f20d337501	9 years ago
Sean Bright	ef39998169	res_pjsip_pidf_eyebeam_body_supplement: Correct status presentation ... This change fixes PIDF content generation when the underlying device state is considered in use. Previously it was incorrectly marked as closed meaning they were offline/unavailable. The code now correctly marks them as open. Additionally: * Generate an XML element for our activity instead of a using a text node. * Consider every extension state other than "unavailable" to be 'open' status. * Update the XML namespaces and structure to reflect those documented in RFC 4480 * Use 'on-the-phone' (defined in RFC 4880) instead of 'busy' as the "in use" activity. This change results in eyeBeam using the appropriate icon for the watched user. This was tested on eyeBeam 1.5.20.2 build 59030 on Windows. ASTERISK-26659 #close Reported by: Abraham Liebsch patches: ASTERISK-26659.diff submitted by snuffy (license 5024) Change-Id: I6e5ad450f91106029fb30517b8c0ea0c2058c810	9 years ago
Joshua Colp	448317ddc7	res_pjsip: Add support for dnsmgr to external_media_address. ... The "external_media_address" option on transports is now resolved using dnsmgr. This allows it to be automatically refreshed regularly if refreshes are enabled in dnsmgr. If the system is using a dynamic IP address a dynamic DNS hostname can be provided to keep the IP address up to date. Change-Id: Ia54771720dff0105bde55d5bbb81a3ba437e05b2	9 years ago
Corey Farrell	7c3cdc81f7	Fix compiler warnings on Fedora 26 / GCC 7. ... GCC 7 has added capability to produce warnings, this fixes most of those warnings. The specific warnings are disabled in a few places: * app_voicemail.c: truncation of paths more than 4096 chars in many places. * chan_mgcp.c: callid truncated to 80 chars. * cdr.c: two userfields are combined to cdr copy, fix would break ABI. * tcptls.c: ignore use of deprecated method SSLv3_client_method(). ASTERISK-27156 #close Change-Id: I65f280e7d3cfad279d16f41823a4d6fddcbc4c88	9 years ago
Sean Bright	1078d4f7f6	app_queue: Add announce-position-only-up option ... Setting this option will cause the Queue application to only announce the caller's position if it has improved since the last time that we announced it. Change-Id: I173a124121422209485b043e2bf784f54242fce6	9 years ago
George Joseph	63224d5d9c	bundled_pjproject: Improve SSL/TLS error handling ... OpenSSL has 2 levels or error processing. It's possible for the top layer to return SSL_ERROR_SYSCALL but the lower layer return no error, in which case processing should continue. Only the top layer was being examined though so connections were being torn down when they didn't need to be. This patch adds the examination of the lower level codes, and if they return no errors, allows processing to continue. ASTERISK-27001 Reported-by: Ian Gilmour patches: pjproject-2.6.patch submitted by Ian Gilmour (license 6889) Updated-by: George Joseph and Sauw Ming (Teluu) Merged to upstream pjproject on 7/27/2017 (commit 5631) Change-Id: I23844ca0c68ef1ee550f14d46f6dae57d33b7bd2	9 years ago
Torrey Searle	c24cdbf4ae	chan_pjsip: add a new function PJSIP_DTMF_MODE ... This function is a replica of SIPDtmfMode, allowing the DTMF mode of a PJSIP call to be modified on a per-call basis ASTERISK-27085 #close Change-Id: I20eef5da3e5d1d3e58b304416bc79683f87e7612	9 years ago
Sean Bright	d056f6b2fe	res_rtp_asterisk: Fix mapping of pjsip's ICE roles to ours ... Change-Id: Ia578ede1a55b21014581793992a429441903278b	9 years ago
Joshua Colp	1c07c80eef	Merge "bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues." into 15	9 years ago