360 Commits (8a281776d5e92183489624da7e1993c560e2fecc)

Author	SHA1	Message	Date
Jenkins2	8a281776d5	Merge "pjsip_options: dynamic contact's fields not updated on reload"	8 years ago
Sean Bright	ec1f4bf48d	res_pjsip: Add TLSv1.1 and TLSv1.2 support ... Support for these protocols was added in the same commit as the 'proto' field, so we can safely use the same ./configure check. For reference: https://trac.pjsip.org/repos/changeset/4968 Change-Id: Icf4975d785d6bfb8f30ac7ffa695a0adf9382dac	8 years ago
Jenkins2	cd741136c5	Merge "pjsip: Improve CLI completion performance"	8 years ago
Sean Bright	521f741b04	pjsip: Improve CLI completion performance ... Use the new ast_cli_completion_add() function to improve completion performance for commands like 'pjsip show endpoint.' Change-Id: I76d802294d2ac1766110dc75f7d117c8541ce348	8 years ago
Sean Bright	dbb376f166	pjsip_configuration: Add correct file header ... Change-Id: I25348c386a222bb704aff07f54375108a6402906	8 years ago
Kevin Harwell	bd2218ce63	pjsip_options: dynamic contact's fields not updated on reload ... Dynamic contacts were not being properly updated on reload. As a matter of fact any changes to the AOR that a dynamic contact was associated with were not being applied. On reload, this patch makes it so for each dynamic contact, the associated AOR is now retrieved and the AOR's fields are applied to the contact. ASTERISK-27467 Change-Id: I8e3165dc6a745218c1c9db837f77fafa0516985d	8 years ago
Richard Mudgett	8536a09b86	security-events: Fix SuccessfulAuth using_password declaration. ... The SuccessfulAuth using_password field was declared as a pointer to a uint32_t when the field was later read as a uint32_t value. This resulted in unnecessary casts and a non-portable field value reinterpret in main/security_events.c:add_json_object(). i.e., It would work on a 32 bit architecture but not on a 64 bit big endian architecture. Change-Id: Ia08bc797613a62f07e5473425f9ccd8d77c80935	8 years ago
Joshua Colp	a2caade298	Merge "res_pjsip: Use sorcery prefix operation for contact lookup"	8 years ago
Corey Farrell	53f42cc052	res_pjsip: Fix warning by deferring implicit type cast. ... Mac doesn't like the comparison of -1 to an enum, so store the result of ast_sip_str_to_dtmf to an int so we can check for the negative return value. ast_sip_str_to_dtmf returns an int so this is only delaying the implicit type cast. Change-Id: I0c262c1719ee951aae1f437d733a301cf5f8ad29	8 years ago
Sean Bright	6c53fb5d21	res_pjsip: Use sorcery prefix operation for contact lookup ... This improves performance for registrations assuming that res_config_astdb is not in use. Change-Id: I86f37aa9ef07a4fe63448cb881bbadd996834bb1	8 years ago
Jenkins2	d6213397f0	Merge "res_pjsip: Fix leak on error in ast_sip_auth_vector_init."	8 years ago
Jenkins2	fdba5fce58	Merge "res_pjsip: Avoid crash when contact uri is empty string"	8 years ago
Jenkins2	06746b0af5	Merge "res_pjsip: Ignore empty TLS configuration"	8 years ago
Aaron An	a36d8cc533	res_pjsip: Avoid crash when contact uri is empty string ... Asterisk will crash if contact uri is invalid, so contact_apply_handler should check if the uri is NULL or empty. ASTERISK-27393 #close Reported-by: Aaron An Tested-by: AaronAn Change-Id: Ia0309bdc6b697c73c9c736e1caec910b77ca69f5	8 years ago
Corey Farrell	29205e7adc	res_pjsip: Fix leak on error in ast_sip_auth_vector_init. ... Change-Id: Ib0fc7a18f3135ca8990c3984c9e15f6d26e556e8	8 years ago
Joshua Colp	637b37fb98	Merge "dtls: Add support for ephemeral DTLS certificates."	8 years ago
Sean Bright	ca4e6b568f	res_pjsip: Ignore empty TLS configuration ... When using realtime, fields that are not explicitly set by an administrator are still presented to sorcery as empty strings. Handle this case explicitly. In this particular case, if any of these fields are required for TLS support, their existence should be validated in the 'apply' handler once we have a complete transport definition. ASTERISK-27032 #close Reported by: seanchann.zhou Change-Id: Ie3b5fb421977ccdb33e415d4ec52c3fd192601b7	8 years ago
Sean Bright	04d3785a79	dtls: Add support for ephemeral DTLS certificates. ... This mimics the behavior of Chrome and Firefox and creates an ephemeral X.509 certificate for each DTLS session. Currently, the only supported key type is ECDSA because of its faster generation time, but other key types can be added in the future as necessary. ASTERISK-27395 Change-Id: I5122e5f4b83c6320cc17407a187fcf491daf30b4	8 years ago
Ben Ford	f8e0f9be22	res_pjsip: Add to list of valid characters for from_user. ... Fixes a regression where some characters were unable to be used in the from_user field of an endpoint. Additionally, the backtick was removed from the list of valid characters, since it is not valid, and it was replaced with a single quote, which is a valid character. ASTERISK-27387 Change-Id: Id80c10a644508365c87b3182e99ea49da11b0281	8 years ago
Sean Bright	1bfd1cf640	pjsip_message_filter: Only do interface lookup for wildcard addresses. ... Change-Id: Ie083987e69dc43b6861671c218cacacc11b2072f	8 years ago
Joshua Colp	9e1fbab382	res_pjsip: Add 'ip' as a valid option to 'identify_by' on endpoint. ... When the identify_by option on an endpoint is set to ip it will only be identified using the res_pjsip_endpoint_identifier_ip module. This ensures that it is not mistakenly matched using the username of the From header. To ensure behavior has not changed the default has been changed to "username,ip" for the identify_by option. ASTERISK-27206 Change-Id: I2170b86a7f7e221b4f00bf14aa1ef1ac5b050bbd	8 years ago
Jenkins2	6fc222a1d2	Merge "res_pjsip: Add REF_DEBUG info to module references."	8 years ago
Jenkins2	89859f33b2	Merge "res_pjsip: Fix issues that prevented shutdown of modules."	8 years ago
Corey Farrell	a68a91f722	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	8 years ago
Corey Farrell	b35ac9e566	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	8 years ago
Corey Farrell	65399a5eda	res_pjsip: Add REF_DEBUG info to module references. ... This provides better information to REF_DEBUG log for troubleshooting when the system is unable to unload res_pjsip.so during shutdown due to module references. ASTERISK-27306 Change-Id: I63197ad33d1aebe60d12e0a6561718bdc54e4612	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
George Joseph	61ea872233	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	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
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
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
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
George Joseph	f78f5278ff	pjsip_message_ip_updater: Fix issue handling "tel" URIs ... sanitize_tdata was assuming all URIs were SIP URIs so when a non SIP uri was in the From, To or Contact headers, the unconditional cast of a non-pjsip_sip_uri structure to pjsip_sip_uri caused a segfault when trying to access uri->other_param. * Added PJSIP_URI_SCHEME_IS_SIP(uri) || PJSIP_URI_SCHEME_IS_SIPS(uri) checks before attempting to cast or use the returned uri. ASTERISK-27152 Reported-by: Ross Beer Change-Id: Id380df790e6622c8058a96035f8b8f4aa0b8551f	8 years ago
Richard Mudgett	9e2b2a9837	res_pjsip: Fix prune_on_boot to remove only contacts for the host. ... * Check that the contact's reg_server matches the host's name before deleting any prune_on_boot contacts. We don't want to delete reliable transport contacts made with other servers if the ps_contacts database table is shared with other servers. Thanks to Ross Beer for pointing out that the original prune logic would delete reliable transport contacts from other servers. ASTERISK-27147 Change-Id: I8e439d0d1c266ffdfd7b73d1e5e466180a689bd0	8 years ago
Richard Mudgett	82f4ade959	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	8 years ago
Richard Mudgett	1dcb92bba8	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	8 years ago
Kevin Harwell	521b6fed12	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	8 years ago
Sean Bright	2be8d91c0f	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	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
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
Jenkins2	0f45c979a3	Merge "res_rtp_asterisk / res_pjsip: Add support for BUNDLE."	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
Benjamin Keith Ford	8f72128e66	res_pjsip: Fix crash with from_user containing invalid characters. ... If the from_user field contains certain characters (like @, {, ^, etc.), PJSIP will return a null value for the URI when attempting to parse it. This causes a crash when trying to dial out through a trunk that contains these invalid characters in its from_user field. This change checks the configuration and ensures that an endpoint will not be created if the from_user contains an invalid character. It also adds a null check to the PJSIP URI parsing as a backup. ASTERISK-27036 #close Reported by: Maxim Vasilev Change-Id: I0396fdb5080604e0bdf1277464d5c8a85db913d0	8 years ago
George Joseph	a10bc3e23f	Merge "pjsip_distributor.c: Fix deadlock with TCP type transports."	8 years ago
Richard Mudgett	b485f6c59c	pjsip_distributor.c: Fix deadlock with TCP type transports. ... When a SIP message comes in on a transport, pjproject obtains the lock on the transport and pulls the data out of the socket. Unlike UDP, the TCP transport does not allow concurrent access. Without concurrency the transport lock is not released when the transport's message complete callback is called. The processing continues and eventually Asterisk starts processing the SIP message. The first thing Asterisk tries to do is determine the associated dialog of the message to determine the associated serializer. To get the associated serializer safely requires us to get the dialog lock. To send a request or response message for a dialog, pjproject obtains the dialog lock and then obtains the transport lock. Deadlock can result because of the opposite order the locks are obtained. * Fix the deadlock by obtaining the serializer associated with the dialog another way that doesn't involve obtaining the dialog lock. In this case, we use an ao2 container to hold the associated endpoint and serializer. The new locks are held a brief time and won't overlap other existing lock times. ASTERISK-27090 #close Change-Id: I9ed63f4da9649e9db6ed4be29c360968917a89bd	8 years ago
Richard Mudgett	65a5ac0168	pjsip_distributor.c: Fix unidentified_requests hash functions. ... The OBJ_SEARCH_xxx defines should not be used as if they were individual bits. They represent a multi-bit enumeration value field. Change-Id: I32abc9a475396dab02402a7014357dd94284e17b	8 years ago
Torrey Searle	fb7247c57c	res_pjsip: Add DTMF INFO Failback mode ... The existing auto dtmf mode reverts to inband if 4733 fails to be negotiated. This patch adds a new mode auto_info which will switch to INFO instead of inband if 4733 is not available. ASTERISK-27066 #close Change-Id: Id185b11e84afd9191a2f269e8443019047765e91	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
Alexei Gradinari	7a46309d3d	res_pjsip: New endpoint option "notify_early_inuse_ringing" ... This option was added to control whether to notify dialog-info state 'early' or 'confirmed' on Ringing when already INUSE. The value "yes" is useful for some SIP phones (Cisco SPA) to be able to indicate and pick up ringing devices. ASTERISK-26919 #close Change-Id: Ie050bc30023543c7dfb4365c5be3ce58c738c711	8 years ago