152 Commits (dac5e3a0df5d76967ef581984e79e0d5543c6a89)

Author	SHA1	Message	Date
Richard Mudgett	22810fc635	chan_pjsip/res_pjsip: Add CHANNEL(pjsip,request_uri) ... This patch does three things associated with the initial incoming INVITE request URI. 1) Add access to the full initial incoming INVITE request URI. 2) We were not setting DNID on incoming PJSIP channels. The DNID is the user portion of the initial incoming INVITE Request-URI. The value is accessed by reading CALLERID(dnid). 3) Fix CHANNEL(pjsip,target_uri) documentation. * The initial incoming INVITE request URI is now available using CHANNEL(pjsip,request_uri). * Set the DNID on PJSIP channel creation so CALLERID(dnid) can return the initial incoming INVITE request URI user portion. * CHANNEL(pjsip,target_uri) now correctly documents that the target URI is the contact URI. * Refactored print_escaped_uri() out of channel_read_pjsip() to handle pjsip_uri_print() error condition when the buffer is too small. ASTERISK-27478 Change-Id: I512e60d1f162395c946451becb37af3333337b33	8 years ago
Joshua Colp	d83d96dba6	Merge "res_pjsip_session: Fix multiple leaks."	8 years ago
Joshua Colp	fe23c48081	Merge "res_pjsip_session: Check for errors from ast_stream_topology_set_stream."	8 years ago
Kevin Harwell	dd1a914495	AST-2017-011 - res_pjsip_session: session leak when a call is rejected ... A previous commit made it so when an invite session transitioned into a disconnected state destruction of the Asterisk pjsip session object was postponed until either a transport error occurred or the event timer expired. However, if a call was rejected (for instance a 488) before the session was fully established the event timer may not have been initiated, or it was canceled without triggering either of the session finalizing states mentioned above. Really the only time destruction of the session should be delayed is when a BYE is being transacted. This is because it's possible in some cases for the session to be disconnected, but the BYE is still transacting. This patch makes it so the session object always gets released (no more memory leak) when the pjsip session is in a disconnected state. Except when the method is a BYE. Then it waits until a transport error occurs or an event timeout. ASTERISK-27345 #close Reported by: Corey Farrell Change-Id: I1e724737b758c20ac76d19d3611e3d2876ae10ed	8 years ago
Corey Farrell	ecb81ae4de	res_pjsip_session: Fix multiple leaks. ... * Pre-initialize cloned media state vectors to final size to ensure vector errors cannot happen later in the clone initialization. * Release session_media on vector replace failure in ast_sip_session_media_state_add. * Release clone and media_state in ast_sip_session_refresh if we fail to append to the stream topology, return an error. Change-Id: Ib5ffc9b198683fa7e9bf166d74d30c1334c23acb	8 years ago
Corey Farrell	eba1179795	res_pjsip_session: Check for errors from ast_stream_topology_set_stream. ... Free memory and return error if ast_stream_topology_set_stream fails. Change-Id: I9f4dbf44bed627243d2f1dd8aea2eab6c38a028d	8 years ago
Joshua Colp	4c535f5c30	core / pjsip: Add support for grouping streams together. ... In WebRTC streams (or media tracks in their world) can be grouped together using the mslabel. This informs the browser that each should be synchronized with each other. This change extends the stream API so this information can be stored with streams. The PJSIP support has been extended to use the mslabel to determine grouped streams and store this association on the streams. Finally when creating the SDP the group information is used to cause each media stream to use the same mslabel. ASTERISK-27379 Change-Id: Id6299aa031efe46254edbdc7973c534d54d641ad	8 years ago
Alexander Traud	8f65d91dfd	res_pjsip_session: Rewrite o= with external_media_address. ... PJSIP allows a domain name as external_media_address. This allows chan_pjsip to be used behind a NAT with changing IP addresses. The IP address of that domain is resolved to the c= line already. This change sets also the o= line to that domain. ASTERISK-27341 #close Change-Id: I690163b6e762042ec38b3995aa5c9bea909d8ec4	8 years ago
Daniel Tryba	6576e4320a	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	8 years ago
Jenkins2	89859f33b2	Merge "res_pjsip: Fix issues that prevented shutdown of modules."	8 years ago
Corey Farrell	eb224fea5e	res_pjsip_session: Fix format_cap leak. ... ASTERISK-27306 Change-Id: I2c8d3fc148f9f53715c958314e1146f9611741f3	8 years ago
Corey Farrell	7d04544986	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	8 years ago
Joshua Colp	dd4c573e6d	Merge "res_pjsip_session: outgoing call did not offer all configured codecs"	8 years ago
Joshua Colp	87c4a72f16	Merge "res_pjsip_session: Reduce (and improve) SDP renegotiation."	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
Joshua Colp	1b468b543d	Merge "res_pjsip: Use ast_sip_is_content_type() where appropriate"	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	f77a2367e7	Merge "bridge: Change participant SFU streams when source streams change."	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
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
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
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
Jenkins2	bee342fdd2	Merge "res_pjsip_session: Preserve stream name during renegotiation."	8 years ago
Walter Doekes	f856d9b42b	res/res_pjsip: Standardize/fix localnet checks across pjsip. ... In 2dee95cc (ASTERISK-27024) and 776ffd77 (ASTERISK-26879) there was confusion about whether the transport_state->localnet ACL has ALLOW or DENY semantics. For the record: the localnet has DENY semantics, meaning that "not in the list" means ALLOW, and the local nets are in the list. Therefore, checks like this look wrong, but are right: /* See if where we are sending this request is local or not, and if not that we can get a Contact URI to modify */ if (ast_apply_ha(transport_state->localnet, &addr) != AST_SENSE_ALLOW) { ast_debug(5, "Request is being sent to local address, " "skipping NAT manipulation\n"); (In the list == localnet == DENY == skip NAT manipulation.) And conversely, other checks that looked right, were wrong. This change adds two macro's to reduce the confusion and uses those instead: ast_sip_transport_is_nonlocal(transport_state, addr) ast_sip_transport_is_local(transport_state, addr) ASTERISK-27248 #close Change-Id: Ie7767519eb5a822c4848e531a53c0fd054fae934	8 years ago
Joshua Colp	68bcfccd52	res_pjsip_session: Preserve stream name during renegotiation. ... Stream names within Asterisk can have meaning so when an externally initiated renegotiation occurs we need to preserve the name of the stream if it already exists. Change-Id: I29f50d0cc7f3238287d6d647777e76e1bdf8c596	8 years ago
Torrey Searle	33a648d4c6	res/res_pjsip_session: allow SDP answer to be regenerated ... If an SDP answer hasn't been sent yet, it's legal to change it. This is required for PJSIP_DTMF_MODE to work correctly, and can also have use in the future for updating codecs too. ASTERISK-27209 #close Change-Id: Idbbfb7cb3f72fbd96c94d10d93540f69bd51e7a1	8 years ago
Richard Mudgett	f2c14f00b8	res_pjsip_session.c: Fix crash when declining an active stream. ... If a previously active stream is declined we could crash because the channel's thread is still using the stream while we are updating the topology in the serializer thread. * Defer removing any declined stream's handler until we have blocked the channel's thread with the channel lock. ASTERISK-27212 Change-Id: I50e1d3ef26f8e41948f4c411ee329aa3b960a420	8 years ago
Jenkins2	df4bcdda2a	Merge "res_pjsip_session/_sdp_rtp: Handling of 'msid' is incorrect"	8 years ago
Joshua Colp	62092bc114	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	8 years ago
Kevin Harwell	104a8047a5	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	8 years ago
Joshua Colp	2a4283f3e7	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	8 years ago
Torrey Searle	65c560894d	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	8 years ago
Joshua Colp	680c491a62	bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues. ... This change does a few things to improve packet loss and renegotiation: 1. On outgoing RTP streams we will now properly reflect out of order packets and packet loss in the sequence number. This allows the remote jitterbuffer to better reorder things. 2. Video updates can now be discarded for a period of time after one has been sent to prevent flooding of clients. 3. For declined and removed streams we will now release any media session resources associated with them. This was not previously done and caused an issue where old state was being used for a new stream. 4. RTP bundling was not actually removing bundled RTP instances from the parent. This has been resolved by removing based on the RTP instance itself and not the SSRC. 5. The code did not properly handle explicitly unbundling an RTP instance from its parent. This now works as expected. ASTERISK-27143 Change-Id: Ibd91362f0e4990b6129638e712bc8adf0899fd45	8 years ago
Kevin Harwell	7da6ddda30	res_pjsip: Add "webrtc" configuration option ... This patch creates a new configuration option called "webrtc". When enabled it defaults and enables the following options that are needed in order for webrtc to work in Asterisk: rtcp-mux, use_avpf, ice_support, and use_received_transport=enabled media_encryption=dtls dtls_verify=fingerprint dtls_setup=actpass When "webrtc" is enabled, this patch also parses the "msid" media level attribute from an SDP. It will also appropriately add it onto the outgoing session when applicable. Lastly, when "webrtc" is enabled h264 RTCP FIR feedback frames are now sent. ASTERISK-27119 #close Change-Id: I5ec02e07c5d5b9ad86a34fdf31bf2f9da9aac6fd	8 years ago
Joshua Colp	065c3005ad	res_rtp_asterisk / res_pjsip: Add support for BUNDLE. ... BUNDLE is a specification used in WebRTC to allow multiple streams to use the same underlying transport. This reduces the number of ICE and DTLS negotiations that has to occur to 1 normally. This change implements this by adding support for it to the RTP SDP module in PJSIP. BUNDLE can be turned on using the "bundle" option and on an offer we will offer to bundle streams together. On an answer we will accept any bundle groups provided. Once accepted each stream is bundled to another RTP instance for transport. For the res_rtp_asterisk changes the ability to bundle an RTP instance to another based on the SSRC received from the remote side has been added. For outgoing traffic if an RTP instance is bundled to another we will use the other RTP instance for any transport related things. For incoming traffic received from the transport instance we look up the correct instance based on the SSRC and use it for any non-transport related data. ASTERISK-27118 Change-Id: I96c0920b9f9aca7382256484765a239017973c11	8 years ago
Mark Michelson	45df25a579	chan_pjsip: Add support for multiple streams of the same type. ... The stream topology (list of streams and order) is now stored with the configured PJSIP endpoints and used during the negotiation process. Media negotiation state information has been changed to be stored in a separate object. Two of these objects exist at any one time on a session. The active media state information is what was previously negotiated and the pending media state information is what the media state will become if negotiation succeeds. Streams and other state information is stored in this object using the index (or position) of each individual stream for easy lookup. The ability for a media type handler to specify a callback for writing has been added as well as the ability to add file descriptors with a callback which is invoked when data is available to be read on them. This allows media logic to live outside of the chan_pjsip module. Direct media has been changed so that only the first audio and video stream are directly connected. In the future once the RTP engine glue API has been updated to know about streams each individual stream can be directly connected as appropriate. Media negotiation itself will currently answer all the provided streams on an offer within configured limits and on an offer will use the topology created as a result of the disallow/allow codec lines. If a stream has been removed or declined we will now mark it as such within the resulting SDP. Applications can now also request that the stream topology change. If we are told to do so we will limit any provided formats to the ones configured on the endpoint and send a re-invite with the new topology. Two new configuration options have also been added to PJSIP endpoints: max_audio_streams: determines the maximum number of audio streams to offer/accept from an endpoint. Defaults to 1. max_video_streams: determines the maximum number of video streams to offer/accept from an endpoint. Defaults to 1. ASTERISK-27076 Change-Id: I8afd8dd2eb538806a39b887af0abd046266e14c7	8 years ago
Jenkins2	2c3c862cee	Merge "res_pjsip_refer/session: Calls dropped during transfer"	9 years ago
George Joseph	ea3f8c6889	res_pjsip_session: Correct inverted test in session_outgoing_nat_hook ... There was a typo introduced in commit 776ffd77 which was preventing the transport's external media address from being used. ASTERISK-27024 #close Reported-by: Christopher van de Sande patches: patch.diff submitted by Florian Floimair (license 6892) Change-Id: I7ec617171eaa2d86d2680b00cf37d5088adafc27	9 years ago
Kevin Harwell	9e53c30610	res_pjsip_refer/session: Calls dropped during transfer ... When doing an attended transfer it's possible for the transferer, after receiving an accepted response from Asterisk, to send a BYE to Asterisk, which can then be processed before Asterisk has time to start and/or complete the transfer process. This of course causes the transfer to not complete successfully, thus dropping the call. This patch makes it so any BYEs received from the transferer, after the REFER, that initiate a session end are deferred until the transfer is complete. This allows the channel that would have otherwise been hung up by Asterisk to remain available throughout the transfer process. ASTERISK-27053 #close Change-Id: I43586db79079457d92d71f1fd993be9a3b409d5a	9 years ago
Joshua Colp	861984eac0	res_pjsip: Add support for returning only reachable contacts and use it. ... This introduces the ability for PJSIP code to specify filtering flags when retrieving PJSIP contacts. The first flag for use causes the query code to only retrieve contacts that are not unreachable. This change has been leveraged by both the Dial() process and the PJSIP_DIAL_CONTACTS dialplan function so they will now only attempt calls to contacts which are not unreachable. ASTERISK-26281 Change-Id: I8233b4faa21ba3db114f5a42e946e4b191446f6c	9 years ago
Joshua Colp	dece2eb892	Merge "res_pjsip_session : fixed wrong From Header number On Re-invite"	9 years ago
Richard Mudgett	30fbed65f1	res_pjsip_session.c: Process initial INVITE sooner. (key exists) ... Retransmissions of an initial INVITE could be queued in the serializer before we have processed the first INVITE message. If the first INVITE message doesn't get completely processed before the retransmissions are seen then we could try to setup the same call from the retransmissions. A symptom of this is seeing a (key exists) message associated with an INVITE. An earlier change attempted to address this kind of problem by calculating a distributor serializer to use for unassociated messages. Part of that change also made incoming calls keep using that distributor serializer. (ASTERISK-26088) However, some leftover code was still deferring the INVITE processing to the session's serializer even though we were already in that serializer. This not only is unnecessary but would cause the same call resetup problem. * Removed the code to defer processing the initial INVITE to the session's serializer because we are already running in that serializer. ASTERISK-26998 #close Change-Id: I1e822d82dcc650e508bc2d40d545d5de4f3421f6	9 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	9 years ago
Yasin CANER	99dea9ba84	res_pjsip_session : fixed wrong From Header number On Re-invite ... ASTERISK-26964 #close Change-Id: I55a9caa7dc90e6c4c219cb09b5c2ec08af84a302	9 years ago