asterisk

Commit Graph

Author	SHA1	Message	Date
Kevin Harwell	06dada3f01	codec negotiation: add incoming_call_offer_prefs option Add a new option, incoming_call_offer_pref, to res_pjsip endpoints that specifies the preferred order of codecs after receiving an offer. This patch does the following: Adds a new enumeration, ast_sip_call_codec_pref, used by the the new configuration option that's added to the endpoint media structure. Adds a new ast_sip_session_caps structure that's set for each session media object. Creates a new file, res_pjsip_session_caps that "implements" the new structure and option, and is compiled into the res_pjsip_session library. ASTERISK-28756 #close Change-Id: I35e7a2a0c236cfb6bd9cdf89539f57a1ffefc76f	7 years ago
Joshua C. Colp	87fda066ea	res_rtp_asterisk: Improve video performance in certain networks. The receive buffer will now grow if we end up flushing the receive queue after not receiving the expected packet in time. This is done in hopes that if this is encountered again the extra buffer size will allow more time to pass and any missing packets to be received. The send buffer will now grow if we are asked for packets and can't find them. This is done in hopes that the packets are from the past and have simply been expired. If so then in the future with the extra buffer space the packets should be available. Sequence number cycling has been handled so that the correct sequence number is calculated and used in various places, including for sorting packets and for determining if a packet is old or not. NACK sending is now more aggressive. If a substantial number of missing sequence numbers are added a NACK will be sent immediately. Afterwards once the receive buffer reaches 25% a single NACK is sent. If the buffer continues to grow and reaches 50% or greater a NACK will be sent for each received future packet to aggressively ask the remote endpoint to retransmit. ASTERISK-28764 Change-Id: I97633dfa8a09a7889cef815b2be369f3f0314b41	7 years ago
Kevin Harwell	f8a852605d	Merge "res/res_pjsip_sdp_rtp: Fix MOH transitions"	7 years ago
Kevin Harwell	a3b3a9d2dc	Merge "pjsip: Update ACLs on named ACL changes."	7 years ago
Torrey Searle	77c9ba8e63	res/res_pjsip_sdp_rtp: Fix MOH transitions Update the state of remote_hold immediately on receipt of remote SDP so that the information is available when building the SDP answer ASTERISK-28754 #close Change-Id: I7026032a807e9c95081cb8f060400b05deb4836f	7 years ago
George Joseph	3854b561a5	Merge "bridging: Add better support for adding/removing streams."	7 years ago
George Joseph	12540a46a1	Merge "RTP/ICE: Send on first valid pair."	7 years ago
Joshua C. Colp	d6712790cd	pjsip: Update ACLs on named ACL changes. This change extends the Sorcery API to allow a wizard to be told to explicitly reload objects or a specific object type even if the wizard believes that nothing has changed. This has been leveraged by res_pjsip and res_pjsip_acl to reload endpoints and PJSIP ACLs when a named ACL changes. ASTERISK-28697 Change-Id: Ib8fee9bd9dd490db635132c479127a4114c1ca0b	7 years ago
George Joseph	78b01f41ae	res_pjsip_outbound_registration: Fix SRV failover on timeout In order to retry outbound registrations for some situations, we need access to the tdata from the original request. For instance, for 401/407 responses we need it to properly construct the subsequent request with the authentication. We also need it if we're iterating over a DNS SRV response record set so we can skip entries we've already tried. We've been getting the tdata from the server response rdata and transaction but that only works for the failures where there was actually a response (4XX, 5XX, etc). For timeouts there's no response and therefore no rdata or transaction from which to get the tdata. When processing a single A/AAAA record for a server, this wasn't an issue as we just retried that same server after the retry timer expired. If we got an SRV record set for the server though, without the state from the tdata, we just kept trying the first entry in the set repeatedly instead of skipping to the next one in the list. * Added a "last_tdata" member to the client state structure to keep track of the sent tdata. * Updated registration_client_send() to save the tdata it used into the client_state. * Updated sip_outbound_registration_response_cb() to use the tdata saved in client_state when we don't get a response from the server. We still use the tdata from the transaction when we DO get a response from the server so we can properly handle 4XX responses where our new request depends on it. General note on timeouts: Although res_pjsip_outbound_registration skips to the next record immediately when a timeout occurs during SRV set traversal, it's pjproject that determines how long to wait before a timeout is declared. As with other SIP message types, pjproject will continue trying the same server at an interval specified by "timer_t1" until "timer_b" expires. Both of those timers are set in the pjsip.conf "system" section. ASTERISK-28746 Change-Id: I199b8274392d17661dd3ce3b4d69a3968368fa06	7 years ago
Joshua C. Colp	5a5be92b79	bridging: Add better support for adding/removing streams. This change adds support to bridge_softmix to allow the addition and removal of additional video source streams. When such a change occurs each participant is renegotiated as needed to reflect the update. If another video source is added then each participant gets another source. If a video source is removed then it is removed from each participant. This functionality allows you to have both your webcam and screenshare providing video if you desire, or even more streams. Mapping has been changed to use the topology index on the source channel as a unique identifier for outgoing participant streams, this will never change and provides an easy way to establish the mapping. The bridge_simple and bridge_native_rtp modules have also been updated to renegotiate when the stream topology of a party changes allowing the same behavior to occur as added to bridge_softmix. If a screen share is added then the opposite party is renegotiated. If that screen share is removed then the opposite party is renegotiated again. Some additional fixes are also included in here. Stream state is now conveyed in SDP so sendonly/recvonly/inactive streams can be requested. Removed streams now also remove previous state from themselves so consumers don't get confused. ASTERISK-28733 Change-Id: I93f41fb41b85646bef71408111c17ccea30cb0c5	7 years ago
George Joseph	a6de4497e6	Merge "res_pjsip_sdp_rtp: implement hold state handling on moh_passthrough"	7 years ago
Ben Ford	168637cc0c	RTP/ICE: Send on first valid pair. When handling ICE negotiations, it's possible that there can be a delay between STUN binding requests which in turn will cause a delay in ICE completion, preventing media from flowing. It should be possible to send media when there is at least one valid pair, preventing this scenario from occurring. A change was added to PJPROJECT that adds an optional callback (on_valid_pair) that will be called when the first valid pair is found during ICE negotiation. Asterisk uses this to start the DTLS handshake, allowing media to flow. It will only be called once, either on the first valid pair, or when ICE negotiation is complete. ASTERISK-28716 Change-Id: Ia7b68c34f06d2a1d91c5ed51627b66fd0363d867	7 years ago
Torrey Searle	bf4340f0ec	res_pjsip_sdp_rtp: implement hold state handling on moh_passthrough When moh_passthrough is used, asterisk is only generating invites of type sendonly and sendrecv instead of taking fully into account the on hold state of the local and remote parties ASTERISK-28738 #close Change-Id: Iaaad9fbc033cb14803d433b8a4071bc337047761	7 years ago
Kevin Harwell	3865b3fd6a	res_rtp_asterisk: bad audio (static) due to incomplete dtls/srtp setup There was a race condition between client initiated DTLS setup, and handling of server side ice completion that caused the underlying SSL object to get cleared during DTLS initialization. If this happened Asterisk would be left in a partial DTLS setup state. RTP packets were sent and received, but were not being encrypted and decrypted. This resulted in no audio, or static. Specifically, this occurred when '__rtp_recvfrom' was processing the handshake sequence from the client to the server, and then 'ast_rtp_on_ice_complete' gets called from another thread and clears the SSL object when calling the 'dtls_perform_setup' function. The timing had to be just right in the sense that from the external SSL library perspective SSL initialization completed (rtp recv), Asterisk clears/resets the SSL object (ice done), and then checks to see if SSL is intialized (rtp recv). Since it was cleared, Asterisk thinks it is not finished, thus not completing 'dtls_srtp_setup'. This patch removes calls to 'dtls_perform_setup', which clears the SSL object, in 'ast_rtp_on_ice_complete'. When ice completes, there is no reason to clear the underlying SSL object. If an ice candidate changes a full protocol level renegotiation occurs. Also, in the case of bundled ICE candidates are reused when a stream is added. So no real reason to have to clear, and reset in this instance. Also, this patch adds a bit of extra logging to aid in diagnosis of any future problems. ASTERISK-28742 #close Change-Id: I34c9e6bad5a39b087164646e2836e3e48fe6892f	7 years ago
Joshua Colp	519f21ecb2	Merge "res_pjsip_session: Fix off-nominal session refreshes."	7 years ago
Sean Bright	aeff1f2c53	res_musiconhold: Avoid spurious warning when 'format' is the empty string The change to res_config_odbc that allowed empty strings to be returned to realtime consumers¹ causes a warning to be emitted when loading MoH classes. So we need to treat an empty 'format' as if it was not specified to avoid the warning. ASTERISK-28735 #close Reported by: Ross Beer [1] https://gerrit.asterisk.org/c/asterisk/+/13722 Change-Id: I9a271d721e1a0973e80ebe7d75b46a0d8fa0e5a5	7 years ago
Joshua C. Colp	ac155decae	res_pjsip_session: Fix off-nominal session refreshes. Given a scenario where session refreshes occur close to each other while another is finishing it was possible for the session refreshes to occur out of order. It was also possible for session refreshes to be delayed for quite some time if a session refresh did not result in a topology change. For the out of order session refreshes the first session refresh would be queued due to a transaction in progress. This transaction would then finish. When finished a separate task to process the delayed requests queue would be queued for handling. A second refresh would be requested internally before this delayed request queued task was processed. As no transaction was in progress this session refresh would be immediately handled before the queued session refresh. The code will now check if any delayed requests exist before allowing a session refresh to immediately occur. If any exist then the session refresh is queued. For the delayed session refreshes if a session refresh did not result in a topology change the attempt would be immediately stopped and no other delayed requests would be processed. The code will now go through the entire delayed requests queue until a delayed request results in a request actually being sent. ASTERISK-28730 Change-Id: Ied640280133871f77d3f332be62265e754605088	7 years ago
Joshua Colp	49809cb078	Merge "res_rtp_asterisk: Don't produce transport-cc if no packets."	7 years ago
Friendly Automation	dda7f986d0	Merge "res_config_odbc: Preserve empty strings returned by the database"	7 years ago
Friendly Automation	a41c971371	Merge "res_stasis_playback: Prevent media_index from going out of bounds"	7 years ago
Joshua C. Colp	1b53d329ac	res_rtp_asterisk: Don't produce transport-cc if no packets. The code assumed that when the transport-cc feedback function was called at least one packet will have been received. In practice this isn't always true, so now we just reschedule the sending and do nothing. Change-Id: Iabe7b358704da446fc3b0596b847bff8b8a0da6a	7 years ago
Joshua Colp	0c845063b3	Merge "res_pjsip_messaging: Allow Content-Type to be overridden"	7 years ago
Friendly Automation	d4b9529a58	Merge "res_stasis: trigger cleanup after update"	7 years ago
Friendly Automation	87ab83d03f	Merge "stasis/app: don't lock an app before a call to send"	7 years ago
Friendly Automation	499f41051e	Merge "res_pjsip_pubsub: Increment persistence data ref when recreating."	7 years ago
Sean Bright	eb9252ea27	res_config_odbc: Preserve empty strings returned by the database When res_config_odbc (and perhaps other realtime backends) reads a SQL NULL from the database, it coalesces the value to the empty string which prevents it from being returned to the realtime core. However, if it instead reads the empty string from the database, it needs a way to encode that fact without having the value omitted entirely. It does this by changing the value to a string with a single space. The realtime code in main/config.c recognizes this special case and _turns the string back into the empty string_ before passing it to realtime API consumers. For all of this to work, we need to ensure that we actually pass the single-space-string back to the realtime core, which is currently failing because we are trimming the value before checking its content. So instead we now special case the single-space-string case so that empty values are returned properly. ASTERISK-28719 #close Reported by: EDV O-TON Change-Id: I673ed8c31ad037aa224e80c78c7a1dc4e4a4e3de	7 years ago
Sean Bright	31dc904380	res_stasis_playback: Prevent media_index from going out of bounds Incrementing stasis_app_playback.media_index directly in our playback loop means that when we reach the end of our playlist the index into the vector will be outside of the bounds of the vector. Instead use a temporary variable and only assign when we're sure that we are in bounds. ASTERISK-28713 #close Reported by: Sébastien Duthil Change-Id: Ib53f7f156097e0607eb5871d9d78d246ed274928	7 years ago
Joshua C. Colp	a1f0c833ab	res_pjsip_pubsub: Increment persistence data ref when recreating. Each subscription needs to have a reference to the persisted data for it, as well as the main JSON contained within the tree. When recreating a subscription this did not occur and they both shared the same reference. ASTERISK-28714 Change-Id: I706abd49ea182ea367a4ac3feca2706460ae9f4a	7 years ago
Sean Bright	03d24ca4c1	res_pjsip_messaging: Allow Content-Type to be overridden ASTERISK-26082 #close Reported by: Alex Change-Id: I6549e90932016349bc72b0f053432dc25286f4fb	7 years ago
Kevin Harwell	cce2b0da95	stasis/app: don't lock an app before a call to send Calling 'app_send' eventually calls the app's message handler. It's possible for a handler to obtain a lock on another object, and then need/want to lock the app object. If the caller of 'app_send' locks the app object prior to calling then there's a potential for a deadlock, if another thread calls 'app_send' without locking. This patch makes it so 'app_send' is not called with the app object locked in the section of code doing such. ASTERISK-28423 #close Change-Id: I6767c6d0933c7db1b984018966eefca4c0638a27	7 years ago
Kevin Harwell	4206830a52	res_stasis: trigger cleanup after update The cleanup code in stasis shuts down applications if they are in a deactivated state, and no longer have explicit subscriptions. When registering an app the cleanup code was running before calling 'update'. When it should be executed after 'update' since a call to register may re-activate the app. We don't want it to shutdown before the 'update' otherwise the app won't be re-activated, or registered. This patch makes it so the cleanup code is executed post 'update'. ASTERISK-28679 #close Change-Id: I8f2c0b17e33bb8128441567b97fd4c7bf74a327b	7 years ago
Sean Bright	b1ca2c5d71	res_pjsip_messaging: Ensure MESSAGE_SEND_STATUS is set properly We need to wait for the message sending callback to finish to know if we succeeded or failed. ASTERISK-25421 #close Reported by: Dmitriy Serov Change-Id: I22b954398821d2caf4c6fe58f0607c8cfa378059	7 years ago
Sean Bright	c376e9f8a8	res_statsd: Document that res_statsd does nothing on its own ASTERISK-24484 #close Reported by: Dan Jenkins Change-Id: I05f298904511d6739aefb1486b6fcbee27efa9ec	7 years ago
George Joseph	0380288f7c	Merge "res_realtime: Fix 'realtime update2' argument handling"	7 years ago
Friendly Automation	4255277ffd	Merge "feat: AudioSocket channel, application, and ARI support."	7 years ago
Joshua Colp	d5fce4bc34	Merge "res_pjsip_notify: Only allow a single Event header to be added to a NOTIFY"	7 years ago
Sean Bright	094e87b0dc	res_realtime: Fix 'realtime update2' argument handling The change in `9b99ef50b5` updated the syntax of the 'realtime update2' CLI command but did not correctly update the calls to ast_update2_realtime(). The issue this addresses was originally opened because we aren't allowing a SQL NULL to be set as part of the update, but this is a limitation of the Realtime API and is not a bug. Additionally, this patch: * Corrects the example in the command documentation to reflect 'update2' instead of 'update.' * Fixes the leading spacing of the command documentation. * Checks that the required 'NULL' literal argument is present where we expect it to be. ASTERISK-21794 #close Reported by: Cédric Bassaget Change-Id: Idda63a5dc50d5f9bcb34c27ea3238d90f733b2cd	7 years ago
Seán C McCord	163efbd724	feat: AudioSocket channel, application, and ARI support. This commit adds support for [AudioSocket]( https://wiki.asterisk.org/wiki/display/AST/AudioSocket), a very simple bidirectional audio streaming protocol. There are both channel and application interfaces. A description of the protocol can be found on the above referenced GitHub page. A short talk about the reasons and implementation can be found on [YouTube](https://www.youtube.com/watch?v=tjduXbZZEgI), from CommCon 2019. ARI support has also been added via the existing "externalMedia" ARI functionality. The UUID is specified using the arbitrary "data" field. ASTERISK-28484 #close Change-Id: Ie866e6c4fa13178ec76f2a6971ad3590a3a588b5	7 years ago
Sean Bright	29d867ed67	res_pjsip_endpoint_identifier_ip: Document support for hostnames ASTERISK-25429 #close Reported by: Joshua C. Colp Change-Id: I7cdfc6026821636acc2465094b7fcde8471a3824	7 years ago
Sean Bright	90af050fa4	res_pjsip_notify: Only allow a single Event header to be added to a NOTIFY ASTERISK-27775 #close Reported by: AvayaXAsterisk Change-Id: Iad158e908e34675ad98f74d09c5e73024e50c257	7 years ago
Friendly Automation	51f811183a	Merge "ARI: Ability to inhibit COLP frames when adding channels to a bridge"	7 years ago
Friendly Automation	34746220a0	Merge "res_pjsip_pubsub: Add ability to persist generator state information."	7 years ago
Joshua C. Colp	4e7adbd8f4	res_pjsip_pubsub: Add ability to persist generator state information. Some body generators, such as dialog-info+xml, require storing state information which is then conveyed in the NOTIFY request itself. Up until now there was no way for such body generators to persist this information. Two new API calls have been added to allow body generators to set and get persisted data. This data is persisted out alongside the normal persistence information and allows the body generator to restore state information or to simply use this for normal storage of state. State is stored in the form of JSON and it is up to the body generator to interpret this as needed. The dialog-info+xml body generator has been updated to take advantage of this to persist the version number. ASTERISK-27759 Change-Id: I5fda56c624fd13c17b3c48e0319b77079e9e27de	7 years ago
Sean Bright	312abaa1fe	res_pjsip_endpoint_identifier_ip.c: Add port matching support Adds source port matching support when IP matching is used: [example] type = identify match = 1.2.3.4:5060/32, 1.2.3.4:6000/32, asterisk.org:4444 If the IP matches but the source port does not, we reject and search for alternatives. SRV lookups are still performed if enabled (srv_lookups = yes), unless the configured FQDN includes a port number in which case just a host lookup is performed. ASTERISK-28639 #close Reported by: Mitch Claborn Change-Id: I256d5bd5d478b95f526e2f80ace31b690eebba92	7 years ago
George Joseph	d66b01d3bf	Merge "res_pjsip_config_wizard: Fix change detection for wizard settings"	7 years ago
Friendly Automation	255a647c53	Merge "websocket: Consider pending SSL data when waiting for socket input"	7 years ago
Sean Bright	b40dd11afe	res_pjsip_config_wizard: Fix change detection for wizard settings ast_sorcery_changeset_create() is not commutative and will fail to detect differences between two variable lists depending on what changed, so switch to ast_variable_lists_match(). ASTERISK-28492 #close Reported by: Jean-Denis Girard Change-Id: I7b3256983ddfaa2138d3de92a444a53b5193a4e1	7 years ago
Sean Bright	7d94bdde9d	res_agi: Improve GET FULL VARIABLE documentation ASTERISK-28673 #close Reported by: Jonathan Harris Change-Id: I591afdec669622bfa19243aabec31b579652c92f	7 years ago
Sean Bright	87110c1bdf	websocket: Consider pending SSL data when waiting for socket input When TLS is in use, checking the readiness of the underlying FD is insufficient for determining if there is data available to be read. So before polling the FD, check if there is any buffered data in the TLS layer and use that first. ASTERISK-28562 #close Reported by: Robert Sutton Change-Id: I95fcb3e2004700d5cf8e5ee04943f0115b15e10d	7 years ago
Jean Aunis	034ac357ad	ARI: Ability to inhibit COLP frames when adding channels to a bridge This patch adds a new flag "inhibitConnectedLineUpdates" to the 'addChannel' operation in the Bridges REST API. When set, this flag avoids generating COLP frames when the specified channels enter the bridge. ASTERISK-28629 Change-Id: Ib995d4f0c6106279aa448b34b042b68f0f2ca5dc	7 years ago

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5000 Commits (06dada3f01c8fd43ad30dff0feb11cb248ff426d)