5008 Commits (1b01a6abfd1f323c9043fc9c65f4ab038296f77c)

Author	SHA1	Message	Date
Joshua C. Colp	e181cbde63	res_pjsip_session: Apply intention behind requested formats. ... When an outgoing channel is created a list of formats may optionally be provided which is used as a request that the formats be used if possible. If an endpoint is not configured for any of the formats we ignore this request and use what is configured. This has the side effect of also including other stream types (such as video) that were not present in the requested formats. This change makes it so that the intention of the request is preserved - that is if only an audio format is requested then even if there is no joint audio format between the request and the configuration we will still only place an audio stream in the outgoing call. ASTERISK-28787 Change-Id: Ia54c0c63e94aca176169b9bae4bb8a8380ea245f	5 years ago
Joshua C. Colp	4b2729e89d	res_rtp_asterisk: Ensure sufficient space for worst case NACK. ... ASTERISK-28790 Change-Id: I10df52f98b19ed62575f25dab36e82d136dccd99	5 years ago
Joshua C. Colp	6c4e181b2f	res_pjsip_sdp_rtp: Only do hold/unhold on default audio stream. ... When examining a stream to determine hold/unhold information we only care about the default audio stream. Other streams aren't used for hold/unhold. ASTERISK-28784 Change-Id: I7a1f10f07822c4aee1f98a38b9628849b578afe4	5 years ago
Sungtae Kim	b58dfbc68a	res_pjsip_session: Fixed wrong session termination ... When the Asterisk receives 200 OK with invalid SDP, the Asterisk/PJPROJECT terminating the session. But if the channel was in the Bridge, Asterisk tries send the Re-Invite before terminating the session. And when the Asterisk sending the Re-Invite, it doesn't check the SDP is NULL or not. This crashes the Asterisk. Fixed it to close the session correctly if the UAS sends the 200 OK with wrong SDP. ASTERISK-28743 Change-Id: Ifa864e0e125b1a7ed2f3abd4164187e1dddc56da	5 years ago
Joshua C. Colp	f0d6189032	res_pjsip_session: Don't restrict non-audio default streams to sendrecv. ... The state of the default audio stream is used for hold/unhold so we restrict it to sendrecv as the core does not handle when it changes as a result of hold/unhold. This restriction does not apply to other media types though so we now only restrict it to audio. This allows the other default streams to store their state at all values, and not just sendrecv and removed. ASTERISK-28783 Change-Id: I139740f38cea7f7d92a876ec2631ef50681f6625	5 years ago
Michael Neuhauser	580e260ff8	chan_psip, res_pjsip_sdp_rtp: ignore rtptimeout if direct-media is active ... Do not hang up a PJSIP channel on RTP timeout if that channel is in a direct-media bridge. Also reset the time of the last received RTP packet when direct-media ends (wait full rtp_timeout period before checking first time after audio came back to Asterisk). ASTERISK-28774 Reported-by: Michael Neuhauser Change-Id: I8b62012be7685849e8fb2b1c5dd39d35313ca2d1	5 years ago
Jaco Kroon	c7c52b8e7a	res_rtp_asterisk: implement ACL mechanism for ICE and STUN addresses. ... A pure blacklist is not good enough, we need a whitelist mechanism as well, and the simplest way to do that is to re-use existing ACL infrastructure. This makes it simpler to blacklist say an entire block (/24) except a smaller block (eg, a /29 or even a /32). Normally you'd need to recursively split the block, so if you want to blacklist a /24 except for a /29 you'd end up with a blacklit for a /25, /26, /27 and /28. I feel that having an ACL instead of a blacklist only is clearer. Change-Id: Id57a8df51fcfd3bd85ea67c489c85c6c3ecd7b30 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	5 years ago
Torrey Searle	b8a67fc15d	res_rtp_asterisk: Send correct sender SSRC when p2p bridge in use ... bridge_p2p_rtp_write will forward rtp to the bridged rtp instance without modifying the ssrc. However, it is not updating the SSRC in the bridged rtp. Thus, when SSRC packets are generated, they have the correct SSRC for the sender. ASTERISK-28773 #close Change-Id: I39f923bde28ebb4f0fddc926b92494aed294a478	5 years ago
George Joseph	a3599cc01d	Merge "res_pjsip_sdp_rtp: Don't wait for ICE if not negotiated" into 17	5 years ago
Torrey Searle	1c6314627d	res_pjsip_sdp_rtp: Don't wait for ICE if not negotiated ... If ICE support is enabled but not negotiated, the rtp->ice structure is not being destroyed. This leads to Asterisk waiting for ICE to complete instead of immediately starting the DTLS handshake, resulting in the call leg having no RTP. ASTERISK-28769 #close Change-Id: I17c137546dc9ecfb9583c24dcf4c2ced8bbd7a27	5 years ago
Rodrigo Ramírez Norambuena	6857ba98d5	res_rtp_asterisk: Add 'rtp show settings' cli command ... This change introduce a CLI command for the RTP to display the general configuration. In the first step add the follow fields of the configurations: - rtpstart - rtpend - dtmftimeout - rtpchecksum - strictrtp - learning_min_sequential - icesupport Change-Id: Ibe5450898e2c3e1ed68c10993aa1ac6bf09b821f	5 years ago
Kevin Harwell	2f75f071d2	Merge "res_pjsip_refer: ensure refer progress is still sent after Proceeding()" into 17	5 years ago
Joshua Colp	ada97a6724	Merge "message & stasis/messaging: make text message variables work in ARI" into 17	5 years ago
lvl	88393db74d	res_pjsip_refer: ensure refer progress is still sent after Proceeding() ... ASTERISK-28766 #close Change-Id: I5ce2210062f9325db762edbf6e46075079bb2cd1	5 years ago
Joshua C. Colp	029c3e49d4	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	5 years ago
Kevin Harwell	c47dbf1fbe	message & stasis/messaging: make text message variables work in ARI ... When a text message was received any associated variable was not written to the ARI TextMessageReceived event. This occurred because Asterisk only wrote out "send" variables. However, even those "send" variables would fail ARI validation due to a TextMessageVariable formatting bug. Since it seems the TextMessageReceived event has never been able to include actual variables it was decided to remove the TextMessageVariable object type from ARI, and simply return a JSON object of key/value pairs for variables. This aligns more with how the ARI sendMessage handles variables, and other places in ARI. That being the case, and since this is technically an API breaking change (no one should really be affected since things never really worked) the ARI version was updated to reflect that. ASTERISK-28755 #close Change-Id: Ia6051c01a53b30cf7edef84c27df4ed4479b8b6f	5 years ago
Torrey Searle	1c8c560f39	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	5 years ago
Kevin Harwell	31e9c110f5	Merge "pjsip: Update ACLs on named ACL changes." into 17	5 years ago
George Joseph	e4f5142e23	Merge "bridging: Add better support for adding/removing streams." into 17	5 years ago
George Joseph	f1896e9279	Merge "RTP/ICE: Send on first valid pair." into 17	5 years ago
Joshua C. Colp	bc3c095e9f	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	5 years ago
George Joseph	5f36196384	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	5 years ago
Joshua C. Colp	957681e08b	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	5 years ago
George Joseph	f93325ce52	Merge "res_pjsip_sdp_rtp: implement hold state handling on moh_passthrough" into 17	5 years ago
Ben Ford	7a921cfe8c	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	5 years ago
Torrey Searle	fb7cff4103	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	5 years ago
Kevin Harwell	08d8035599	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	5 years ago
Joshua Colp	f424562a41	Merge "res_pjsip_session: Fix off-nominal session refreshes." into 17	5 years ago
Sean Bright	d1a96fa9ba	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	5 years ago
Joshua C. Colp	3a8da5ece7	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	5 years ago
Friendly Automation	2f992bf800	Merge "res_rtp_asterisk: Don't produce transport-cc if no packets." into 17	5 years ago
Friendly Automation	d9e625519a	Merge "res_config_odbc: Preserve empty strings returned by the database" into 17	5 years ago
Friendly Automation	259edcf5c7	Merge "res_stasis_playback: Prevent media_index from going out of bounds" into 17	5 years ago
Joshua C. Colp	325e3f8111	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	5 years ago
Joshua Colp	3efeb147bb	Merge "res_pjsip_messaging: Allow Content-Type to be overridden" into 17	5 years ago
Friendly Automation	e7e65588e4	Merge "res_stasis: trigger cleanup after update" into 17	5 years ago
George Joseph	2c72e26478	Merge "res_pjsip_pubsub: Increment persistence data ref when recreating." into 17	5 years ago
Friendly Automation	82eb8e7097	Merge "stasis/app: don't lock an app before a call to send" into 17	5 years ago
Sean Bright	646789106f	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	5 years ago
Sean Bright	8b8c1dd07f	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	5 years ago
Joshua C. Colp	35c9332edf	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	5 years ago
Sean Bright	ef5702cef1	res_pjsip_messaging: Allow Content-Type to be overridden ... ASTERISK-26082 #close Reported by: Alex Change-Id: I6549e90932016349bc72b0f053432dc25286f4fb	5 years ago
Kevin Harwell	737bd8365e	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	5 years ago
Kevin Harwell	7031c3b7bd	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	5 years ago
Sean Bright	69cf67d8c6	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	5 years ago
Sean Bright	deed0f5706	res_statsd: Document that res_statsd does nothing on its own ... ASTERISK-24484 #close Reported by: Dan Jenkins Change-Id: I05f298904511d6739aefb1486b6fcbee27efa9ec	5 years ago
Joshua Colp	cf37de8a5d	Merge "res_realtime: Fix 'realtime update2' argument handling" into 17	5 years ago
Friendly Automation	af4e503005	Merge "res_pjsip_notify: Only allow a single Event header to be added to a NOTIFY" into 17	5 years ago
Sean Bright	3779e76b68	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	5 years ago
Sean Bright	28275609b3	res_pjsip_endpoint_identifier_ip: Document support for hostnames ... ASTERISK-25429 #close Reported by: Joshua C. Colp Change-Id: I7cdfc6026821636acc2465094b7fcde8471a3824	5 years ago