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Author	SHA1	Message	Date
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
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
George Joseph	ed9134282e	res_pjsip: Update logging to show contact->uri in messages ... An earlier commit changed the id of dynamic contacts to contain a hash instead of the uri. This patch updates status change logging to show the aor/uri instead of the id. This required adding the aor id to contact and contact_status and adding uri to contact_status. The aor id gets added to contact and contact_status in their allocators and the uri gets added to contact_status in pjsip_options when the contact_status is created or updated. ASTERISK-25598 #close Reported-by: George Joseph Tested-by: George Joseph Change-Id: I56cbec1d2ddbe8461367dd8b6da8a6f47f6fe511	10 years ago
Jonathan Rose	eadad24b59	Unset BRIDGEPEER when leaving a bridge ... Currently if a channel is transferred out of a bridge, the BRIDGEPEER variable (also BRIDGEPVTCALLID) remain set even once the channel is out of the bridge. This patch removes these variables when leaving the bridge. ASTERISK-25600 #close Reported by: Mark Michelson Change-Id: I753ead2fffbfc65427ed4e9244c7066610e546da	10 years ago
Richard Mudgett	bb0b60619d	res_sorcery_memory_cache.c: Fix off nominal ref leak. ... Change-Id: If83d63cf11cbc6df9b15251848b01feb570ade49	10 years ago
Richard Mudgett	e7c88e11aa	sched.c: Make not return a sched id of 0. ... According to the API doxygen a sched ID of 0 is valid. Unfortunately, 0 was never returned historically and several users incorrectly coded usage of the returned sched ID assuming that 0 was invalid. ASTERISK-25476 Change-Id: Ib19c7ebb44ec9fd393ef6646dea806d4f34e3a20	10 years ago
Richard Mudgett	4aed349a7b	Audit improper usage of scheduler exposed by 5c713fdf18. (v13 additions) ... chan_sip.c: * Initialize mwi subscription scheduler ids earlier because of ASTOBJ to ao2 conversion. * Initialize register scheduler ids earlier because of ASTOBJ to ao2 conversion. chan_skinny.c: * Fix more scheduler usage for the valid 0 id value. ASTERISK-25476 Change-Id: If9f0e5d99638b2f9d102d1ebc9c5a14b2d706e95	10 years ago
Richard Mudgett	6d9156d10f	Audit improper usage of scheduler exposed by 5c713fdf18. ... channels/chan_iax2.c: * Initialize struct chan_iax2_pvt scheduler ids earlier because of iax2_destroy_helper(). channels/chan_sip.c: channels/sip/config_parser.c: * Fix initialization of scheduler id struct members. Some off nominal paths had 0 as a scheduler id to be destroyed when it was never started. chan_skinny.c: * Fix some scheduler id comparisons that excluded the valid 0 id. channel.c: * Fix channel initialization of the video stream scheduler id. pbx_dundi.c: * Fix channel initialization of the packet retransmission scheduler id. ASTERISK-25476 Change-Id: I07a3449f728f671d326a22fcbd071f150ba2e8c8	10 years ago
Alexander Traud	b76c196e13	codec_resample: Increase buffer for Opus Codec. ... ASTERISK-25599 #close Change-Id: I1f88a88c59fb4e1e62bbdbb100c7152d48e73f10	10 years ago
Matt Jordan	6614babea2	bridges/bridge_t38: Add a bridging module for managing T.38 state ... When 4875e5ac32 was merged, it fixed several issues with a direct media bridge transitioning to handling a T.38 fax. However, it uncovered a race condition caused by the bridging core. When a channel involved in a T.38 fax leaves a bridge, the frame queued by the channel driver that should inform the far side that it is no longer in a T.38 fax may not make it across the bridge. The bridging framework is *extremely* aggressive in tearing down the bridge, and control frames that are currently in flight *may* get dropped. This patch adds a new module to the bridging framework, bridge_t38. This module maintains some notion of the T.38 state for the two channels in a bridge. When the bridge detects that it is being torn down or when one of the two channels leaves, it informs the respective channel(s) that they should stop faxing. This ensures that channels switch back to audio if they survive and are ejected out of a bridge while faxing. ASTERISK-25582 Change-Id: If5b0bb478eb01c4607c9f4a7fc17c7957d260ea0	10 years ago
Niklas Larsson	3fcf160fae	CHANGES: Fix a typo ... Change-Id: Iceb3d9bb78140c376174a7bee197dfcf8ef9cda7	10 years ago
Matt Jordan	762dc9c89d	Merge "fastagi: record file closed after sending result" into 13	10 years ago
Matt Jordan	5d80a6e714	Merge "main: Slight refactor of main. Improve color situation." into 13	10 years ago
Kevin Harwell	45efbf8503	fastagi: record file closed after sending result ... The fastagi record-file testsuite test sometimes fails reporting an empty recorded file. This was happening because Asterisk was sending the agi result notification prior to actually closing the file and the data, being buffered, had not been written to the file yet when the test attempts to check the file size. This patch makes it so the record file stream is closed prior to sending the agi result notification. ASTERISK-25593 #close Change-Id: I6b2b3be3ae37f7c7b18e672c419a89b3b8513cde	10 years ago
Walter Doekes	b2787876d6	main: Slight refactor of main. Improve color situation. ... Several issues are addressed here: - main() is large, and half of it is only used if we're not rasterisk; fixed by spliting up the daemon part into a separate function. - Call ast_term_init from rasterisk as well. - Remove duplicate code reading/writing asterisk history file. - Attempt to tackle background color issues and color changes that occur. Tested by starting asterisk -c until the colors stopped changing at odd locations. ASTERISK-25585 #close Change-Id: Ib641a0964c59ef9fe6f59efa8ccb481a9580c52f	10 years ago
Matt Jordan	e96604c902	Merge "Fixed some typos" into 13	10 years ago
David M. Lee	59881fbb99	Fixed some typos ... Fixes some minor typos in the CHANGES file, plus an embarrasing typo in the StatsD API. Change-Id: I9ca4858c64a4a07d2643b81baa64baebb27a4eb7	10 years ago