27078 Commits (965a0eee46d24321f74c244e23c5a5f45e67e12b)

 

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Author	SHA1	Message	Date
Dade Brandon	965a0eee46	res_rtp_asterisk: Resolve further timing issues with DTLS negotiation ... Resolves an edge case dtls negotiation delay for certain networks which somehow manage to drop the rtcp side's packet when these are both sent ast_rtp_remote_address_set, causing it to have to time-out and restart the handshake. Move dtls pending bio flush in to it's own function, and call it from ast_rtp_on_ice_complete, when we're rtp->ice, rather than when ast_rtp_remote_address_set. Keep the existing flush from the recent change to res_rtp_remote_address_set if ice is not being used. ASTERISK-25614 #close Reported-by: XenCALL Tested by: XenCALL Change-Id: Ie2caedbdee1783159f375589b6fd3845c8577ba5	10 years ago
Matt Jordan	7ec9b5b98f	Merge "app_queue: update RT members when the 1st call joins a queue with no agents" into 13	10 years ago
Joshua Colp	e652a786d6	Merge "json: Audit ast_json_* usage for thread safety." into 13	10 years ago
Mark Michelson	03dab00d1d	Merge topic 'alembic_fixes' into 13 ... * changes: Alembic: Increase column size of PJSIP AOR "contact". Alembic: Add PJSIP global keep_alive_interval.	10 years ago
Carlos Oliva	ae428d8460	app_queue: update RT members when the 1st call joins a queue with no agents ... If a call enters on a queue and the members on that queue are updated in realtime (ex: using mysql inserting a new agent) the queue members are never refreshed and the call will stay in the queue until other event occurs. This happens only if this is the first call of the queue and there is no agents servicing. This patch prevent this issue, ensuring realtime members are updated if there is one call in the queue and no available agents ASTERISK-25442 #close Change-Id: If1e036d013a5c1d8b0bf60d71d48fe98694a8682	10 years ago
Joshua Colp	62e49a7e20	Merge "res_sorcery_memory_cache: Add support for a full backend cache." into 13	10 years ago
Joshua Colp	59d5bb0613	res_sorcery_memory_cache: Add support for a full backend cache. ... This change introduces the configuration option 'full_backend_cache' which changes the cache to be a full mirror of the backend instead of a per-object cache. This allows all sorcery retrieval operations to be carried out against it and is useful for object types which are used in a "retrieve all" or "retrieve some" pattern. ASTERISK-25625 #close Change-Id: Ie2993487e9c19de563413ad5561c7403b48caab5	10 years ago
Joshua Colp	51a2cab780	Merge "rtp_engine: Ignore empty filenames in DTLS configuration." into 13	10 years ago
Joshua Colp	0cefcabd58	rtp_engine: Ignore empty filenames in DTLS configuration. ... When applying an empty DTLS configuration the filenames in the configuration will be empty. This is actually valid to do and each filename should simply be ignored. Change-Id: Ib761dc235638a3fb701df337952f831fc3e69539	10 years ago
Joshua Colp	158a0a5422	chan_sip: Enable WebSocket support by default. ... Per the documentation the WebSocket support in chan_sip is supposed to be enabled by default but is not. This change corrects that. Change-Id: Icb02bbcad47b11a795c14ce20a9bf29649a54423	10 years ago
Joshua Colp	a9d6fc571d	json: Audit ast_json_* usage for thread safety. ... The JSON library Asterisk uses, jansson, is not thread safe for us in a few ways. To help with this wrappers for JSON object reference count increasing and decreasing were added which use a global lock to ensure they don't clobber over each other. This does not extend to reference count manipulation within the jansson library itself. This means you can't safely use the object borrowing specifier (O) in ast_json_pack and you can't share JSON instances between objects. This change removes uses of the O specifier and replaces them with the o specifier and an explicit ast_json_ref. Some cases of instance sharing have also been removed. ASTERISK-25601 #close Change-Id: I06550d8b0cc1bfeb56cab580a4e608ae4f1ec7d1	10 years ago
Mark Michelson	53bd5a539a	Alembic: Increase column size of PJSIP AOR "contact". ... When running the PJSIP AMI "show_endpoint" test with automatic conversion to realtime, the test would fail. This was because the AOR "contact" column was sized at 40, and the configured contact was larger than that. This commit increases the size of the contact column to 255 characters. Change-Id: Ia65bc7fd37699b7c0eaef9629a1a31eab9a24ba1	10 years ago
Mark Michelson	da17dc4d75	Alembic: Add PJSIP global keep_alive_interval. ... The keep_alive_interval option was added about a year ago, but no alembic revision was created to add the appropriate column to the database. This commit fixes the problem and adds the column. This was discovered by running the testsuite with automatic conversion to realtime enabled. Change-Id: If3ef92a7c4f4844d08f8aae170d2178aec5c4c1a	10 years ago
Matt Jordan	280adca0a5	Merge "AMI: Fixed OriginateResponse message" into 13	10 years ago
server-pandora	24ae124e4f	res_rtp_asterisk.c: Fix DTLS negotiation delays. ... - Trigger pending DTLS packets to send out, once the RTP instance's remote address is set. - Avoids locking the DTLS structure unnecessarily by only doing this if DTLS is passive. - Add DTLS locks around the structurally sensitive calls in the SSL portion of __rtp_recvfrom, since dtls_srtp_check_pending does not lock inside of itself, and we're dealing with the SSL BIO in at least two threads. WebRTC channels may receive a DTLS handshake before ast_rtp_remote_address_set is called, which causes there to be a pending response to send out. Previous to 1ad827, this was handled by calling dtls_srtp_check_pending on receipt of any RTP packet - a STUN or RTP packet could trigger the pending handshake response. Since that was rightfully removed, whenever the DTLS handshake is received before the remote address is set, we would have to wait until another SSL packet arrives. As of Chrome M47's optimizations to their handshake process, WebRTC conversations between Chrome M47+ and Asterisk, where Asterisk is passive, experience a 1 second delay without this patch, because the SSL handshake is received before ICE negotation stores the remote_address, and the next SSL packet isn't received until after a 1 second timeout in Chrome, which causes a new handshake request. ASTERISK-25614 #close Change-Id: I547f1be7e302dbf71f6553dd8cbc0657b1d0b908	10 years ago
Richard Mudgett	36097a185d	Fix sscanf() format string type mismatch. ... ASTERISK-25615 Reported by: George Joseph Change-Id: Ieff35307254ca193f3d473cff2e396ca57c7ce0b	10 years ago
Matt Jordan	77ac79b175	Merge "main/utils: Don't emit an ERROR message if the read end of a pipe closes" into 13	10 years ago
Matt Jordan	94f9927784	main/utils: Don't emit an ERROR message if the read end of a pipe closes ... An ERROR or WARNING message should generally indicate that something has gone wrong in Asterisk. In the case of writing to a file descriptor, Asterisk is not in control of when the far end closes its reading on a file descriptor. If the far end does close the file descriptor in an unclean fashion, this isn't a bug or error in Asterisk, particularly when the situation can be gracefully handled in Asterisk. Currently, when this happens, a user would see the following somewhat cryptic ERROR message: "utils.c: write() returned error: Broken pipe" There's a few problems with this: (1) It doesn't provide any context, other than 'something broke a pipe' (2) As noted, it isn't actually an error in Asterisk (3) It can get rather spammy if the thing breaking the pipe occurs often, such as a FastAGI server (4) Spammy ERROR messages make Asterisk appear to be having issues, or can even mask legitimate issues This patch changes ast_carefulwrite to only log an ERROR if we actually had one that was reasonably under our control. For debugging purposes, we still emit a debug message if we detect that the far side has stopped reading. Change-Id: Ia503bb1efcec685fa6f3017bedf98061f8e1b566	10 years ago
George Joseph	5b867fa904	pjsip/config_transport: Check pjproject version at runtime for async ops ... pjproject < 2.5.0 will segfault on a tls transport if async_operations is greater than 1. A runtime version check has been added to throw an error if the version is < 2.5.0 and async_operations > 1. To assist in the check, a new api "ast_compare_versions" was added to utils which compares 2 major.minor.patch.extra version strings. ASTERISK-25615 #close Change-Id: I8e88bb49cbcfbca88d9de705496d6f6a8c938a98 Reported-by: George Joseph Tested-by: George Joseph	10 years ago
Jonathan Rose	14b41115e3	chan_sip: Add TCP/TLS keepalive to TCP/TLS server ... Adds the TCP Keep Alive option to TCP and TLS server sockets. Previously this option was only being set on session sockets. http://www.tldp.org/HOWTO/html_single/TCP-Keepalive-HOWTO/ According to the link above, the SO_KEEPALIVE option is useful for knowing when a TCP connected endpoint has severed communication without indicating it or has become unreachable for some reason. Without this patch, keep alive is not set on the socket listening for incoming TCP sessions and in Komatsu's report this resulted in the thread listening for TCP becoming stuck in a waiting state. ASTERISK-25364 #close Reported by: Hiroaki Komatsu Change-Id: I7ed7bcfa982b367dc64b4b73fbd962da49b9af36	10 years ago
Joshua Colp	c344fb02f4	Merge "res_pjsip: Add existence and readablity checks for tls related files" into 13	10 years ago
Joshua Colp	2be0d49042	Merge "app_meetme: Set default value for audio_buffers." into 13	10 years ago
pchero	fe8011cc50	AMI: Fixed OriginateResponse message ... When the asterisk sending OriginateResponse message, it doesn't set the "Uniqueid". And it didn't support correct response message for Application originate. ASTERISK-25624 #close Change-Id: I26f54f677ccfb0b7cfd4967a844a1657fd69b74d	10 years ago
tcambron	cd119ed4a2	res_chan_stats: Fix bug to send correct statistics to StatsD ... Fixed a bug that originally would show a negative number of active calls occuring in Asterisk. A gauge is persistent so incrementing and decrementing it results in a more consistent performance. Also changed to the call to StatsD to use ast_statsd_log_string() so that a "+" could be sent to StatsD. ASTERISK-25619 #close Change-Id: Iaaeff5c4c6a46535366b4d16ea0ed0ee75ab2ee7	10 years ago
Corey Farrell	ddf4dddf4f	app_meetme: Set default value for audio_buffers. ... The default value was never set for audio_buffers, causing bad audio quality. This ensures the default is always set. ASTERISK-25569 #close Change-Id: I2d2ee3e644120b0f9f6ea6ab9286d7d590942a44	10 years ago
Filip Jenicek	142d4fefb8	chan_sip: Check sip_pvt pointer in ast_channel_get_t38_state(c) ... Asterisk may crash when calling ast_channel_get_t38_state(c) on a locked channel which is being hung up. ASTERISK-25609 #close Change-Id: Ifaa707c04b865a290ffab719bd2e5c48ff667c7b	10 years ago
George Joseph	21962dad93	res_pjsip: Add existence and readablity checks for tls related files ... Both transport and endpoint now check for the existence and readability of tls certificate and key files before passing them on to pjproject. This will cause the object to not load rather than waiting for pjproject to discover that there's a problem when a session is attempted. NOTE: chan_sip also uses ast_rtp_dtls_cfg_parse but it's located in build_peer which is gigantic and I didn't want to disturb it. Error messages will emit but it won't interrupt chan_sip loading. ASTERISK-25618 #close Change-Id: Ie43f2c1d653ac1fda6a6f6faecb7c2ebadaf47c9 Reported-by: George Joseph Tested-by: George Joseph	10 years ago
Eugene Voityuk	28d9243079	chan_sip.c: Start ICE negotiation when response is sent or received. ... The current logic for ICE negotiation starts it when receiving an SDP with ICE candidates. This is incorrect as ICE negotiation can only start when each call party have at least one pair of local and remote candidate. Starting ICE negotiation early would result in negotiation failure and ultimately no audio. This change makes it so ICE negotiation is only started when a response with SDP is received or when a response with SDP is sent. ASTERISK-24146 Change-Id: I55a632bde9e9827871b09141d82747e08379a8ca	10 years ago
Joshua Colp	246e513110	Merge "res_pjsip/config_transport: Prevent async_operations > 1 when protocol = tls" into 13	10 years ago
Joshua Colp	35cc249732	Merge "translate: Avoid a warning message when doing FEC within Opus Codec." into 13	10 years ago
George Joseph	e03582a1c2	res_pjsip/config_transport: Prevent async_operations > 1 when protocol = tls ... See ASTERISK-25615. If the transport protocol is tls and async_operations > 1, pjproject will segfault if more than one operation is attempted on the same socket. Until this is fixed upstream, a check has been added to throw an error if a tls transport config has async_operations set to > 1. ASTERISK-25615 Change-Id: I76b9a5b2a5a0054fe71ca5851e635f2dca7685a6 Reported-by: George Joseph Tested-by: George Joseph	10 years ago
Alexander Traud	876600ce6e	codec_resample: Increase buffer for Opus Codec with FEC. ... ASTERISK-25599 #close Change-Id: Idbd187f711b2ec63dda949ca0f79aa0c1a0a0b6e	10 years ago
Alexander Traud	69e3d40ad7	translate: Avoid a warning message when doing FEC within Opus Codec. ... ASTERISK-25616 #close Change-Id: Ibe729aaf2e6e25506cff247cec5149ec1e589319	10 years ago
Richard Mudgett	2b992014dc	chan_sip: Fix crash involving the bogus peer during sip reload. ... A crash happens sometimes when performing a CLI "sip reload". The bogus peer gets refreshed while it is in use by a new call which can cause the crash. * Protected the global bogus peer object with an ao2 global object container. ASTERISK-25610 #close Change-Id: I5b528c742195681abcf713c6e1011ea65354eeed	10 years ago
Joshua Colp	eb9a353490	Merge "res_pjsip/contacts/statsd: Make contact lifecycle events more consistent" into 13	10 years ago
Matt Jordan	529535f0c2	Revert "bridges/bridge_t38: Add a bridging module for managing T.38 state" ... This reverts commit 6614babea2. Unfortunately, using a bridge to manage T.38 state will cause severe deadlocks in core_unreal/chan_local. Local channels attempt to reach across both their peer and the peer's bridge to inspect T.38 state. Given the propensity of Local channel chains, managing the locking situation in such a scenario is practically infeasible. Change-Id: Ic687397ffea08dfb899345a443bd990ec3d0416a	10 years ago
George Joseph	450579e908	res_pjsip/contacts/statsd: Make contact lifecycle events more consistent ... It will never be perfect or even pretty, mostly because of the differences between static and dynamic contacts. Created: Can't use the contact or contact_status alloc functions because the objects come and go regardless of the actual state. Can't use the contact_apply_handler, ast_sip_location_add_contact or a sorcery created handler because they only get called for dynamic contacts. Similarly, permanent_uri_handler only gets called for static contacts. So, Matt had it right. :) ast_res_pjsip_find_or_create_contact_status is the only place it can go and not have duplicated code. Both permanent_uri_handler and contact_apply_handler call find_or_create. Removed: Can't use the destructors for the same reason as above. The only place to put this is in persistent_endpoint_contact_deleted_observer which I believe is the "correct" place but even that will handle only dynamic contacts. This doesn't called on shutdown however. There is no hook to use for static contacts that may be removed because of a config change while asterisk is in operation. I moved the cleanup of contact_status from ast_sip_location_delete_contact to the handler as well. Status Change and RTT: Although they worked fine where they were (in update_contact_status) I moved them to persistent_endpoint_contact_status_observer to make it more consistent with removed. There was logic there already to detect a state change. Finally, fixed a nit in permanent_uri_handler rmudgett reported eralier. ASTERISK-25608 #close Change-Id: I4b56e7dfc3be3baaaf6f1eac5b2068a0b79e357d Reported-by: George Joseph Tested-by: George Joseph	10 years ago
Matt Jordan	9c0aaf0609	Merge "res_format_attr_vp8: In SDP, forward max-fr and max-fs for video-codec VP8." into 13	10 years ago
Matt Jordan	ffb0643467	Merge "res_format_attr_opus: Update to latest RFC 7587." into 13	10 years ago
Matt Jordan	e8f78f87a8	Merge "bridges/bridge_t38: Add a bridging module for managing T.38 state" into 13	10 years ago
Alexander Traud	5a18193dc0	res_format_attr_vp8: In SDP, forward max-fr and max-fs for video-codec VP8. ... ASTERISK-25584 #close Change-Id: Iae00071b4ff1ae76f24995aeac4d00284fd14f91	10 years ago
Alexander Traud	3e2178c05e	res_format_attr_opus: Update to latest RFC 7587. ... Beside that, the format-attribute module sends only non-default values in the line fmtp, now. This avoids unnecessary overhead in SDP messages. Furthermore, previously the parameter stereo was not parsed when being the first parameter. ASTERISK-25583 #close Change-Id: Iae85ba3e5960bfd5d51cf65bcffad00dd4875a73	10 years ago
Jonathan Rose	072d94183c	Fix crash in audiohook translate to slin ... This patch fixes a crash which would occur when an audiohook was applied to a channel using an audio codec that could not be translated to signed linear (such as when using pass-through codecs like OPUS or when the codec translator module for the format in use is not loaded). ASTERISK-25498 #close Reported by: Ben Langfeld Change-Id: Ib6ea7373fcc22e537cad373996136636201f4384	10 years ago
Joshua Colp	cfb146e055	Merge "res_pjsip: Use a MD5 hash for static Contact IDs" into 13	10 years ago
Joshua Colp	59134eb7cb	Merge "res_pjsip: Update logging to show contact->uri in messages" into 13	10 years ago
Joshua Colp	1d5ddb4b99	Merge "codec_resample: Increase buffer for Opus Codec." into 13	10 years ago
George Joseph	9184fbeb34	res_pjsip: Use a MD5 hash for static Contact IDs ... When 90d9a70789 was merged, it mostly tested dynamic contacts created as a result of registering a PJSIP endpoint. Contacts generated in this fashion typically have a long alphanumeric string as their object identifier, which maps reasonably well for StatsD. Unfortunately, this doesn't work in the general case. StatsD treats both '.' and ':' characters as special characters. In particular, having a ':' appear in the middle of a StatsD metric will result in the metric being rejected. This causes some obvious issues with SIP URIs. The StatsD API should not be responsible for escaping the metric name passed to it. The metric is treated as a single long string, and it would be challenging to know what to escape in the string passed to the function. Likewise, we don't want to escape the metric in PJSIP, as that involves overhead that is wasted when either res_statsd isn't loaded or enabled. This patch takes an alternative approach. The Contact ID has been changed to be "aor@@uri_hash" instead of "aor@@uri". This (a) won't contain any of the aforementioned special characters, (b) can be done on Contact creation, which has minimal impact on run-time performance, and (c) also conforms to an earlier commit that changed the ID for dynamic contacts. The downside of this is that StatsD users will have to map SHA1 hashes back to the Contacts that are emitting the statistics. To that end, the CLI commands have been updated to include the first 10 characters of the MD5 hash, which should be enough to match what is shown in Graphite (or some other StatsD backend). ASTERISK-25595 #close Change-Id: Ic674a3307280365b4a45864a3571c295b48a01e2 Reported-by: Matt Jordan Tested-by: George Joseph	10 years ago
Joshua Colp	8ab0c2107a	Merge "res_sorcery_memory_cache.c: Fix off nominal ref leak." into 13	10 years ago
Joshua Colp	3b1452b542	Merge "sched.c: Make not return a sched id of 0." into 13	10 years ago
Joshua Colp	53d4f77064	Merge topic 'ASTERISK-25476' into 13 ... * changes: Audit improper usage of scheduler exposed by 5c713fdf18. (v13 additions) Audit improper usage of scheduler exposed by 5c713fdf18.	10 years ago