4995 Commits (certified/16.8)

Author	SHA1	Message	Date
Joshua C. Colp	7b43f9f3e9	res_pjsip_nat: Don't rewrite Contact on REGISTER responses. ... When sending a SIP response to an incoming REGISTER request we don't want to change the Contact header as it will contain the Contacts registered to the AOR and not our own Contact URI. ASTERISK-29235 Change-Id: I35a0723545281dd01fcd5cae497baab58720478c (cherry picked from commit d65041ebea)	5 years ago
George Joseph	a41edfb46f	res_pjsip_session: Make reschedule_reinvite check for NULL topologies ... When the check for equal topologies was added to reschedule_reinvite() it was assumed that both the pending and active media states would actually have non-NULL topologies. We since discovered this isn't the case. We now only test for equal topologies if both media states have non-NULL topologies. The logic had to be rearranged a bit to make sure that we cloned the media states if their topologies were non-NULL but weren't equal. ASTERISK-29215 Change-Id: I61313cca7fc571144338aac826091791b87b6e17	5 years ago
Joshua C. Colp	8713492758	res_rtp_asterisk: Force resync on SSRC change. ... When an SSRC change occurs the timestamps are likely to change as well. As a result we need to reset the timestamp mapping done in the calc_rxstamp function so that they map properly from timestamp to real time. This previously occurred but due to packet retransmission support the explicit setting of the marker bit was not effective. ASTERISK-29352 Change-Id: I2d4c8f93ea24abc1030196706de2d70facf05a5a	5 years ago
Ben Ford	b9fc07ca53	AST-2021-006 - res_pjsip_t38.c: Check for session_media on reinvite. ... When Asterisk sends a reinvite negotiating T38 faxing, it's possible a crash can occur if the response contains a m=image and zero port. The reinvite callback code now checks session_media to see if it is null or not before trying to access the udptl variable on it. ASTERISK-29305 Change-Id: I1dfc51c5fa586e38579ede4bc228edee213ccaa9	6 years ago
Ben Ford	b12fcaaaf0	res_pjsip_session.c: Check topology on re-invite. ... Removes an unnecessary check for the conditional that compares the stream topologies to see if they are equal to suppress re-invites. This was a problem when a Digium phone received an INVITE that offered codecs different than what it supported, causing Asterisk to send the re-invite. ASTERISK-29303 Change-Id: I04dc91befb2387904e28a9aaeaa3bcdbcaa7fa63	6 years ago
Kevin Harwell	bb5b336f78	AST-2021-002: Remote crash possible when negotiating T.38 ... When an endpoint requests to re-negotiate for fax and the incoming re-invite is received prior to Asterisk sending out the 200 OK for the initial invite the re-invite gets delayed. When Asterisk does finally send the re-inivite the SDP includes streams for both audio and T.38. This happens because when the pending topology and active topologies differ (pending stream is not in the active) in the delayed scenario the pending stream is appended to the active topology. However, in the fax case the pending stream should replace the active. This patch makes it so when a delay occurs during fax negotiation, to or from, the audio stream is replaced by the T.38 stream, or vice versa instead of being appended. Further when Asterisk sent the re-invite with both audio and T.38, and the endpoint responded with a declined T.38 stream then Asterisk would crash when attempting to change the T.38 state. This patch also puts in a check that ensures the media state has a valid fax session (associated udptl object) before changing the T.38 state internally. ASTERISK-29203 #close Change-Id: I407f4fa58651255b6a9030d34fd6578cf65ccf09	6 years ago
Alexander Traud	93468c531a	rtp: Enable srtp replay protection ... Add option "srtpreplayprotection" rtp.conf to enable srtp replay protection. ASTERISK-29260 Reported by: Alexander Traud Change-Id: I5cd346e3c6b6812039d1901aa4b7be688173b458	6 years ago
Sean Bright	07eddbd56a	res_rtp_asterisk.c: Fix signed mismatch that leads to overflow ... ASTERISK-29205 #close Change-Id: Ib7aa65644e8df76e2378d7613ee7cf751b9d0bea	6 years ago
Joshua C. Colp	6b5306ecc5	res_pjsip_pidf_digium_body_supplement: Support Sangoma user agent. ... This adds support for both Digium and Sangoma user agent strings for the Sangoma specific body supplement. Change-Id: Ib99362b24b91d3cbe888d8b2fce3fad5515d9482 (cherry picked from commit 7e4bb4ed11)	6 years ago
Kevin Harwell	81b48f2d10	AST-2020-001 - res_pjsip: Return dialog locked and referenced ... pjproject returns the dialog locked and with a reference. However, in Asterisk the method that handles this decrements the reference and removes the lock prior to returning. This makes it possible, under some circumstances, for another thread to free said dialog before the thread that created it attempts to use it again. Of course when the thread that created it tries to use a freed dialog a crash can occur. This patch makes it so Asterisk now returns the newly created dialog both locked, and with an added reference. This allows the caller to de-reference, and unlock the dialog when it is safe to do so. In the case of a new SIP Invite the lock, and reference are now held for the entirety of the new invite handling process. Otherwise it's possible for the dialog, or its dependent objects, like the transaction, to disappear. For example if there is a TCP transport error. ASTERISK-29057 #close Change-Id: I5ef645a47829596f402cf383dc02c629c618969e	6 years ago
Ben Ford	6057050899	AST-2020-002 - res_pjsip: Stop sending INVITEs after challenge limit. ... If Asterisk sends out an INVITE and receives a challenge with a different nonce value each time, it will continuously send out INVITEs, even if the call is hung up. The endpoint must be configured for outbound authentication for this to occur. A limit has been set on outbound INVITEs so that, once reached, Asterisk will stop sending INVITEs and the transaction will terminate. ASTERISK-29013 Change-Id: I2d001ca745b00ca8aa12030f2240cd72363b46f7	6 years ago
Joshua C. Colp	d5df3b95c8	res_pjsip_session: Fix stream name memory leak. ... When constructing a stream name based on the media type and position the allocated name was not being freed causing a leak. Change-Id: I52510863b24a2f531f0a55b440bb2c81844029de	6 years ago
Joshua C. Colp	3e54af3afb	res_pjsip_session: Fix session reference leak. ... The ast_sip_dialog_get_session function returns the session with reference count increased. This was not taken into account and was causing sessions to remain around when they should not be. ASTERISK-29089 Change-Id: I430fa721b0a824311a59effec6056e9ec528e3e8	6 years ago
George Joseph	1620e94af5	res_pjsip_session: Fix issue with COLP and 491 ... The recent 491 changes introduced a check to determine if the active and pending topologies were equal and to suppress the re-invite if they were. When a re-invite is sent for a COLP-only change, the pending topology is NULL so that check doesn't happen and the re-invite is correctly sent. Of course, sending the re-invite sets the pending topology. If a 491 is received, when we resend the re-invite, the pending topology is set and since we didn't request a change to the topology in the first place, pending and active topologies are equal so the topologies-equal check causes the re-invite to be erroneously suppressed. This change checks if the topologies are equal before we run the media state resolver (which recreates the pending topology) so that when we do the final topologies-equal check we know if this was a topology change request. If it wasn't a change request, we don't suppress the re-invite even though the topologies are equal. ASTERISK-29014 Change-Id: Iffd7dd0500301156a566119ebde528d1a9573314	6 years ago
George Joseph	f67c58aa13	debugging: Add enough to choke a mule ... Added to: * bridges/bridge_softmix.c * channels/chan_pjsip.c * include/asterisk/res_pjsip_session.h * main/channel.c * res/res_pjsip_session.c There NO functional changes in this commit. Change-Id: I06af034d1ff3ea1feb56596fd7bd6d7939dfdcc3	6 years ago
George Joseph	9e54db24b6	res_pjsip_session: Handle multi-stream re-invites better ... When both Asterisk and a UA send re-invites at the same time, both send 491 "Transaction in progress" responses to each other and back off a specified amount of time before retrying. When Asterisk prepares to send its re-invite, it sets up the session's pending media state with the new topology it wants, then sends the re-invite. Unfortunately, when it received the re-invite from the UA, it partially processed the media in the re-invite and reset the pending media state before sending the 491 losing the state it set in its own re-invite. Asterisk also was not tracking re-invites received while an existing re-invite was queued resulting in sending stale SDP with missing or duplicated streams, or no re-invite at all because we erroneously determined that a re-invite wasn't needed. There was also an issue in bridge_softmix where we were using a stream from the wrong topology to determine if a stream was added. This also caused us to erroneously determine that a re-invite wasn't needed. Regardless of how the delayed re-invite was triggered, we need to reconcile the topology that was active at the time the delayed request was queued, the pending topology of the queued request, and the topology currently active on the session. To do this we need a topology resolver AND we need to make stream named unique so we can accurately tell what a stream has been added or removed and if we can re-use a slot in the topology. Summary of changes: * bridge_softmix: * We no longer reset the stream name to "removed" in remove_all_original_streams(). That was causing multiple streams to have the same name and wrecked the checks for duplicate streams. * softmix_bridge_stream_sources_update() was checking the old_stream to see if it had the softmix prefix and not considering the stream as "new" if it did. If the stream in that slot has something in it because another re-invite happened, then that slot in old might have a softmix stream but the same stream in new might actually be a new one. Now we check the new_stream's name instead of the old_stream's. * stream: * Instead of using plain media type name ("audio", "video", etc) as the default stream name, we now append the stream position to it to make it unique. We need to do this so we can distinguish multiple streams of the same type from each other. * When we set a stream's state to REMOVED, we no longer reset its name to "removed" or destroy its metadata. Again, we need to do this so we can distinguish multiple streams of the same type from each other. * res_pjsip_session: * Added resolve_refresh_media_states() that takes in 3 media states and creates an up-to-date pending media state that includes the changes that might have happened while a delayed session refresh was in the delayed queue. * Added is_media_state_valid() that checks the consistency of a media state and returns a true/false value. A valid state has: * The same number of stream entries as media session entries. Some media session entries can be NULL however. * No duplicate streams. * A valid stream for each non-NULL media session. * A stream that matches each media session's stream_num and media type. * Updated handle_incoming_sdp() to set the stream name to include the stream position number in the name to make it unique. * Updated the ast_sip_session_delayed_request structure to include both the pending and active media states and updated the associated delay functions to process them. * Updated sip_session_refresh() to accept both the pending and active media states that were in effect when the request was originally queued and to pass them on should the request need to be delayed again. * Updated sip_session_refresh() to call resolve_refresh_media_states() and substitute its results for the pending state passed in. * Updated sip_session_refresh() with additional debugging. * Updated session_reinvite_on_rx_request() to simply return PJ_FALSE to pjproject if a transaction is in progress. This stops us from creating a partial pending media state that would be invalid later on. * Updated reschedule_reinvite() to clone both the current pending and active media states and pass them to delay_request() so the resolver can tell what the original intention of the re-invite was. * Added a large unit test for the resolver. ASTERISK-29014 Change-Id: Id3440972943c611a15f652c6c569fa0e4536bfcb	6 years ago
Joshua C. Colp	7b1ace9040	res_pjsip_session: Don't aggressively terminate on failed re-INVITE. ... Per the RFC when an outgoing re-INVITE is done we should only terminate the dialog if a 481 or 408 is received. ASTERISK-29033 Change-Id: I6c3ff513aa41005d02de0396ba820083e9b18503	6 years ago
George Joseph	d189bf9ae5	res_pjsip_session: Ensure reused streams have correct bundle group ... When a bundled stream is removed, its bundle_group is reset to -1. If that stream is later reused, the bundle parameters on session media need to be reset correctly it could mistakenly be rebundled with a stream that was removed and never reused. Since the removed stream has no rtp instance, a crash will result. Change-Id: Ie2b792220f9291587ab5f9fd123145559dba96d7	6 years ago
George Joseph	8076ed59f0	scope_trace: Add/update utilities ... * Added a AST_STREAM_STATE_END sentinel * Add ast_stream_to_str() * Add ast_stream_state_to_str() * Add ast_stream_get_format_count() * Add ast_stream_topology_to_str() * Add ast_stream_topology_get_active_count() * Add ast_format_cap_append_names() * Add ast_sip_session_get_name() Change-Id: I132eb5971ea41509c660f64e9113cda8c9013b0b	6 years ago
Joshua C. Colp	bd86f382ed	websocket / pjsip: Increase maximum packet size. ... When dealing with a lot of video streams on WebRTC the resulting SDPs can grow to be quite large. This effectively doubles the maximum size to allow more streams to exist. The res_http_websocket module has also been changed to use a buffer on the session for reading in packets to ensure that the stack space usage is not excessive. Change-Id: I31d4351d70c8e2c11564807a7528b984f3fbdd01	6 years ago
Joshua C. Colp	5eec77b452	res_pjsip: Apply AOR outbound proxy to static contacts. ... The outbound proxy for an AOR was not being applied to any statically configured Contacts. This resulted in the OPTIONS requests being sent to the wrong target. This change sets the outbound proxy on statically configured contacts once the AOR configuration is done being applied. ASTERISK-28965 Change-Id: Ia60f3e93ea63f819c5a46bc8b54be2e588dfa9e0	6 years ago
Joshua C. Colp	6777ec0340	res_pjsip_session: Preserve label on incoming re-INVITE. ... When a re-INVITE is received we create a new set of streams that are then swapped in as the active streams. We did not preserve the SDP label from the previous streams, resulting in the label getting lost. This change ensures that if an SDP label is present on the previous stream then it is set on the new stream. ASTERISK-28953 Change-Id: I9dd63b88b562fe96ce5c791a3dae5bcaca258445	6 years ago
Joshua C. Colp	744bfb049c	res_rtp_asterisk: Don't assume setting retrans props means to enable. ... The "value" passed in when setting an RTP property determines whether it should be enabled or disabled. The RTP send and receive retrans props did not examine this to know if the buffers should be enabled. They assumed they always should be. This change makes it so that the "value" passed in is respected. ASTERISK-28939 Change-Id: I9244cdbdc5fd065c7f6b02cbfa572bc55c7123dc	6 years ago
Kevin Harwell	10dad4ea3a	Compiler fixes for gcc 10 ... This patch fixes a few compile warnings/errors that now occur when using gcc 10+. Also, the Makefile.rules check to turn off partial inlining in gcc versions greater or equal to 8.2.1 had a bug where it only it only checked against versions with at least 3 numbers (ex: 8.2.1 vs 10). This patch now ensures any version above the specified version is correctly compared. Change-Id: I54718496eb0c3ce5bd6d427cd279a29e8d2825f9	6 years ago
Joshua C. Colp	02188feb32	fax: Fix crashes in PJSIP re-negotiation scenarios. ... This change fixes a few re-negotiation issues uncovered with fax. 1. The fax support uses its own mechanism for re-negotiation by conveying T.38 information in its own frames. The new support for re-negotiating when adding/removing/changing streams was also being triggered for this causing multiple re-INVITEs. The new support will no longer trigger when transitioning between fax. 2. In off-nominal re-negotiation cases it was possible for some state information to be left over and used by the next re-negotiation. This is now cleared. ASTERISK-28811 ASTERISK-28839 Change-Id: I8ed5924b53be9fe06a385c58817e5584b0f25cc2 (cherry picked from commit f3ac42b9bd)	6 years ago
Joshua Colp	40d379f59e	Revert "Revert "res_rtp_asterisk: Free payload when error on insertion to data buffer"" ... This reverts commit fe3dc091b5. Reason for revert: This will be merged once 16.8-cert1 is released. Change-Id: I5c29f96a70ed7e1fa146a69e7b48bfe31cbee929	6 years ago
Joshua Colp	491f0bae7d	Revert "Revert "res_rtp_asterisk: Resolve loop when receive buffer is flushed"" ... This reverts commit a75317ce24. Reason for revert: This will be merged once 16.8-cert1 is released. Change-Id: Ic2f4dbb19c5756fabe71b9e35c5a924214f5af08	6 years ago
Joshua Colp	fe3dc091b5	Revert "res_rtp_asterisk: Free payload when error on insertion to data buffer" ... This reverts commit fef8a04aad. Reason for revert: Waiting for future release. Change-Id: I67e9837a352b252f8a75ebaa9a5fb7e5b35d41f7	6 years ago
Joshua Colp	a75317ce24	Revert "res_rtp_asterisk: Resolve loop when receive buffer is flushed" ... This reverts commit 22bc8a7168. Reason for revert: Waiting for future release. Change-Id: If924f0b7fa02a72b52c708aa80addc361a87b490	6 years ago
Pirmin Walthert	22bc8a7168	res_rtp_asterisk: Resolve loop when receive buffer is flushed ... When the receive buffer was flushed by a received packet while it already contained a packet with the same sequence number, Asterisk never left the while loop which tried to order the packets. This change makes it so if the packet is in the receive buffer it is retrieved and freed allowing the buffer to empty. ASTERISK-28827 Change-Id: Idaa376101bc1ac880047c49feb6faee773e718b3	6 years ago
Pirmin Walthert	fef8a04aad	res_rtp_asterisk: Free payload when error on insertion to data buffer ... When the ast_data_buffer_put rejects to add a packet, for example because the buffer already contains a packet with the same sequence number, the payload will never be freed, resulting in a memory leak. The data buffer will now return an error if this situation occurs allowing the caller to free the payload. The res_rtp_asterisk module has also been updated to do this. ASTERISK-28826 Change-Id: Ie6c49495d1c921d5f997651c7d0f79646f095cf1	6 years ago
Joshua C. Colp	bdf4d159fd	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	6 years ago
Joshua C. Colp	3d22ecb1e2	res_rtp_asterisk: Ensure sufficient space for worst case NACK. ... ASTERISK-28790 Change-Id: I10df52f98b19ed62575f25dab36e82d136dccd99	6 years ago
Joshua C. Colp	aa04c3f49b	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	7 years ago
George Joseph	444df0ac71	Merge "bridging: Add better support for adding/removing streams." into certified/16.8	7 years ago
George Joseph	b5b52ae386	Merge "res_pjsip_session: Fix off-nominal session refreshes." into certified/16.8	7 years ago
Joshua C. Colp	4458b7a52a	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
Joshua C. Colp	d515dea9c6	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 (cherry picked from commit b438d1d9ad)	7 years ago
Joshua C. Colp	5fccf6d1c6	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	bb19e7feb5	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
Kevin Harwell	43ca35b831	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
George Joseph	443230f5d5	Asterisk Certified 16.8 Preparation ... * Updated .gitreview default branch to certified/16.8 * Updated .version to certified/16.8 * Set all extended support modules to be disabled by default Change-Id: I11c9b5f33865fb541192a786dc25dddf8558e09b	7 years ago
Joshua C. Colp	40e331ff90	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
Kevin Harwell	1627e8eddc	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 (cherry picked from commit dc9875815c)	7 years ago
Kevin Harwell	42c51263b9	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 (cherry picked from commit e103339f02)	7 years ago
Joshua C. Colp	939e18d63e	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 (cherry picked from commit 4d32f5747c)	7 years ago
Sean Bright	04c81f9748	res_statsd: Document that res_statsd does nothing on its own ... ASTERISK-24484 #close Reported by: Dan Jenkins Change-Id: I05f298904511d6739aefb1486b6fcbee27efa9ec	7 years ago
Joshua Colp	18b0e81b61	Merge "res_realtime: Fix 'realtime update2' argument handling" into 16	7 years ago
Friendly Automation	ddf8e6de79	Merge "res_pjsip_notify: Only allow a single Event header to be added to a NOTIFY" into 16	7 years ago
Sean Bright	fbe18165d5	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