[Pkg-voip-commits] [janus] annotated tag upstream/0.2.6 created (now be6c743)
Jonas Smedegaard
dr at jones.dk
Wed Dec 20 21:53:51 UTC 2017
This is an automated email from the git hooks/post-receive script.
js pushed a change to annotated tag upstream/0.2.6
in repository janus.
at be6c743 (tag)
tagging de9d953e1807c343691a5c74449fac93761531ea (commit)
replaces upstream/0.2.5
tagged by Jonas Smedegaard
on Wed Dec 20 21:46:35 2017 +0100
- Log -----------------------------------------------------------------
Upstream version 0.2.6
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Adam Duskett (4):
fix libressl call in dtls.c
Fix SSL library detection
Add openssl pre 1.1 api macro
Add openssl pre 1.1 api macro
Alessandro Toppi (31):
Improve iceloop lifecycle management.
Merge branch 'master' into fix-iceloop-quit
Move iceloop running check from janus_ice_webrtc_free to janus_ice_handles_check.
Check iceloop ptr.
Remove sleep in janus_ice_thread after loop unlocking.
Merge branch 'master' into fix-iceloop-quit
Wait for handle icethread end before scheduling handle destruction.
Quit the main loop in janus_ice_handles_check in case the loop is still running.
Use session table in videoroom plugin to avoid using invalid sessions.
Lock sessions in janus streaming plugin.
Check if ice send thread is ended before freeing session. Front push dtls alert message in case of hangup.
Remove send_thread ptr check in janus_ice_webrtc_hangup.
Reject jsep and trickle while cleaning a previous session.
Add error symbol for wrong WebRTC status.
Revert send_thread ptr check in janus_ice_webrtc_hangup.
Cast away const qualifier from a char ptr to avoid compiler warnings.
Notify hangup to users only when janus_ice_webrtc_free is called.
Use janus_process_error in order to avoid (char *) casting.
Check that icectx is not NULL before invoking g_main_context_wakeup.
Merge branch 'master' into iceloop-sessions-mgmt
Merge branch 'master' into iceloop-sessions-mgmt
Merge branch 'master' into iceloop-sessions-mgmt
Merge remote-tracking branch 'origin/master' into iceloop-sessions-mgmt
Use active sessions lookup in every plugin (excluding videocall). Make a synchronous hangup_media in streaming and recordplay.
Merge branch 'master' into iceloop-sessions-mgmt
Remove some debugging logs.
Add missing janus_plugin_session_is_alive checks.
Use active plugin sessions map, in place of old sessions map.
Set lws count_threads to 1 before creating lws context.
Add some extra iceloop checks.
Fix not initialized old_rooms in janus_textroom_destroy. Re-evaluate as needed 'now' timestamp in textroom and streaming watchdog.
Alex (1):
Fixes #1094
Ancor Gonzalez Sosa (2):
Bower: update webrtc-adapter to 5.0.1
Don't use es6 specific code
Bao Nguyen (1):
Streaming plugin UDP multicast socket bind option to `SO_REUSEADDR`
Ivan Demchuk (2):
Close web socket connection when destroying session
Add missing semicolon
Johan Ouwerkerk (1):
Instruct ESLint to ignore Janus JS module contents when linting your project.
Jonas Smedegaard (1):
New upstream version 0.2.6
Jonathan Martin (2):
Add Thread ID callback in lws_protocols[0] to help identifying caller of `lws_callback_on_writable` and as such speed up lws reactivity
brew installation of libcurl is now called just `curl`
Lorenzo Miniero (222):
New Janus plugin, NoSIP, for legacy interop without touching signalling
Several new helper methods for SDP utilities
New Janus plugin, NoSIP, for legacy interop without touching signalling
Removed info string, cleaned up a bit, and refactored code to limit reduncancy
Merge branch 'nosip' of github.com:meetecho/janus-gateway into nosip
Merge branch 'sdputils-pt2' into nosip
Modified NoSIP plugin code to use #796 and #804 (RTP context and SRTP stuff)
Made incoming_rtcp handler more compact too
Merge branch 'sdputils-pt2' into nosip
Integrated updated IP utilities in NoSIP plugin
Deallocate the local_ip when getting rid of the plugin
Merge branch 'master' into nosip
Merge branch 'master' into nosip
First commit of SIPre plugin placeholder (WIP, still broken)
Merge branch 'master' into sipre
Moved IP self-detect of NoSIP plugin outside of the config parse code
Aligned SIPre plugin to IP utils
Use mqueue to make sure libre calls are done on the loop thread
Updated code to reflect latest feedback
Initialize libre in the loop thread
Working REGISTER
Working outgoing INVITE (incoming calls, BYEs, etc. still WIP)
Refactored message queue payload, and (almost) working incoming calls
Merge branch 'master' into nosip
Removed unneeded memset
Merge branch 'master' into nosip
Merge branch 'master' into sipre
Fixed endless retransmissions on incoming calls
Better management of BYEs and call cleanup
Merge branch 'master' into nosip
Merge branch 'master' into sipre
Merge branch 'master' into nosip
Merge branch 'master' into sipre
Merge branch 'master' into nosip
Merge branch 'master' into sipre
Merge branch 'master' into sipre
Integrated fix made for #885 in SIPre plugin as well
Merge branch 'master' into nosip
Aligned to configure-related changes in master
Merge branch 'master' into sipre
Aligned to configure-related changes in master
Merge branch 'master' into nosip
Aligned to changes to janus.js (intrack)
Aligned to changes to janus.js (ontrack)
Added support to custom headers in REGISTER and INVITE in SIPre plugin
Merge branch 'master' into nosip
Changed some TODOs in FIXMEs
Merge branch 'master' into sipre
Merge branch 'master' into nosip
Merge branch 'master' into sipre
Merge branch 'master' into nosip
Merge branch 'master' into sipre
Parse (and notify) display name when receiving incoming INVITEs
Moved stack to per-user property (but still a single thread and queue)
Fixed crash when shutting Janus down
Implemented DTMF via SIP INFO
Implemented hold/unhold (still WIP)
Made expires in REGISTER configurable/overridable
Temporarily disabled HA1 REGISTER option in SIPre demo
First attempts at getting re-INVITES to work in SIPre plugin
Fixed broken re-INVITE support in SIPre plugin
Send incoming/outgoing SIP messages to event handlers when using SIPre plugin, if sip_set_trace is available in libre
Cleanup of log verbosity in SIPre plugin
Fixed 486 response when we're busy in another call (SIPre plugin)
Merge branch 'master' into nosip
Merge branch 'master' into sipre
Merge branch 'master' into nosip
Merge branch 'master' into sipre
Merge branch 'master' into nosip
Merge branch 'master' into sipre
Aligned SIPre plugin to recent SIP plugin features
Merge branch 'master' into sipre
Add a DNS client when allocating a SIPre stack
Added support for outbound proxies to the SIPre plugin
Merge branch 'master' into nosip
Merge branch 'master' into sipre
Merge branch 'master' into sipre
Merge branch 'master' into nosip
Merge branch 'master' into nosip
Better management of POLLERR errors in NoSIP plugin
Merge branch 'master' into sipre
Added support for offerless INVITEs to the SIPre plugin
Added support for SIP INFO to the SIPre plugin
Implemented early media (183 Session Progress) in SIPre plugin
Cleaned up some currently unused features
Merge branch 'master' into nosip
Made a few changes to how a SIPre session is destroyed
Merge branch 'master' into sipre
Added support for authenticated INVITEs, in case the user registered as guest
Merge branch 'master' into nosip
Added some checks/fixes already made in master
Merge branch 'master' into sipre
Added some checks/fixes already made in master
Merge branch 'master' into nosip
Merge branch 'master' into sipre
Merge branch 'master' into nosip
Merge branch 'master' into sipre
Merge branch 'master' into nosip
Merge branch 'master' into sipre
Merge branch 'master' into nosip
Merge branch 'master' into sipre
Merge branch 'master' into nosip
Aligned NoSIP branch to latest changes
Aligned SIPre branch to latest changes
Merge branch 'master' into nosip
Merge branch 'master' into sipre
Merge branch 'master' into nosip
Merge branch 'master' into sipre
Merge branch 'master' into nosip
Merge branch 'master' into sipre
Merge branch 'master' into nosip
Updated NoSIP plugin to use new rtpsrtp.h header
Merge branch 'master' into sipre
Updated SIPre plugin to use new rtpsrtp.h header
Merge branch 'master' into nosip
Merge branch 'master' into nosip
Added configurable port range for RTP/RTCP ports in the NoSIP plugin
Moved NoSIP links in the demos to highlight the different nature of the plugin
Merge branch 'master' into sipre
Fixed broken enforcement of new RTP/RTCP range in NoSIP plugin
Clarified that the SIPre demo is not currently as stable as the older Sofia one
Added configurable port range for RTP/RTCP ports in the SIPre plugin
Merge branch 'master' into nosip
Merge branch 'master' into sipre
Bumped to version 0.2.6
Merge pull request #799 from meetecho/nosip
Fixed conflicts introduced after merge of NoSIP branch
Merge pull request #823 from meetecho/sipre
Fixed typo in SIPre demo and the navigation bar
Made EchoTest and VideoCall recordings aware of negotiated codecs
Made a few recorder properties atomic
Removed spurious debugging line from EchoTest
Made RTCP BYE management more tolerant, to accomodate older Firefox 52
Merge pull request #1037 from aduskett/master
Fixed some checks in the web demos
Allow multiple codecs in VideoRoom, and publishers to choose which one
Make sure codecs match when switching publishers in a VideoRoom
Added info on emitter (server name) to Janus handlers events
Merge pull request #1040 from meetecho/videoroom-multicodec
Added EINTR checks for all poll() calls (fixes pfunix issue at startup)
Added way to provide custom PeerConnection constraints (see #1028)
Merge pull request #1044 from pitkonst/rabbitmq_transport_exchange
Fixed typo in RabbitMQ transport comment
Merge pull request #1045 from pitkonst/RabbitMQ_events_handler
Improved RTP headers rewriting in case of context switches
Merge pull request #1061 from mquander/rust-plugin-binding-docs
Merge pull request #1060 from mquander/readme-libwebsockets
Updated reference to stable version of libwebsockets in README
Fixed STUN check at startup in IPv6 network (fixes #1053)
Experiments to improve performance of the ICE send thread
Merge pull request #1068 from RouquinBlanc/fix-mac-brew-packages
Merge pull request #1066 from sysbot/bao/SO_REUSEADDR
Merge pull request #1074 from ancorgs/bower_adapter_501
Merge pull request #1078 from ancorgs/old_compat
Merge pull request #1071 from aduskett/master
Merge pull request #1073 from aduskett/ssl
Merge pull request #1082 from suranapranay/master
Fixed check of Jansson version in configure (fixes #1077)
Merge branch 'master' of github.com:meetecho/janus-gateway
Added missing transaction ID to error (fixes #1084)
Added missing PLI when restoring subscriber's video with configure (fixes #1087)
Merge pull request #1085 from edvinanet/master
Merge pull request #1067 from RouquinBlanc/fix-websocket-bsd
Added option to override threshold for detecting timestamp resets
Make sure SDP rid attributes are parsed before ssrc (fixes #1072)
Don't use RTCP BYE as DTLS alerts to close PeerConnections
Allow Streaming viewers to temporarily disable/enable audio/video/data
Use GQueue instead of GList, when last item is important
Merge branch 'master' into icesend-perf
Fixed typo in VideoRoom demo (fixes #1088)
Make sure pending messages are sent before closing a Unix Socket for timeout (fixes #1009)
Merge pull request #1062 from meetecho/stun-ipv6
Merge pull request #1092 from mquander/sctp-error-logging
Fixed include paths of TextRoom plugin
Handle MSG_EOR in datachannels
Fixed typo in Record&Play (caused old recordings not to replay)
Merge pull request #1093 from meetecho/sctp-msgeor
Added onended event to track screensharing from UI button in demo
Merge branch 'master' of github.com:meetecho/janus-gateway
Merge pull request #1035 from meetecho/iceloop-sessions-mgmt
Fixed small nits (coding style)
Merge branch 'master' into icesend-perf
Reverted g_async_queue_push_front to g_async_queue_push (which needs glib 2.46 that may be too recent)
Only use g_async_queue_push_front if glib is recent enough
Merge branch 'master' into icesend-perf
Restored 'stopping' event to handlers in Streaming plugin
Removed extra mutex unlock in SIP plugin
Fixed assertion when accessing non-existing GQueue
More compact cleanup of retransmit buffer
Merge pull request #1063 from meetecho/icesend-perf
Make sure an alert trigger is only enqueued if the send thread exists (see #1083)
Added missing properties to permanent save in AudioBridge and VideoRoom
Return more info about mountpoints if the admin secret is provided
Only list RTSP info when querying mountpoint if libcurl is available
Added option to force UDP when registering in the SIP plugin
Added missing attributes when saving permanent streaming mountpoints (fixes #1096)
Fixed typo
Fixed typo (double check)
Reduced verbosity when postprocessing VP8/VP9 mjr files
Merge branch 'master' of github.com:meetecho/janus-gateway
Fixed problem of printing non-terminated string when logging in SCTP code
Added missing return statement in videocall demo
Merge pull request #1103 from cmacq2/eslint-disable
Merge branch 'master' of github.com:meetecho/janus-gateway
Fixed some typos (static analysis)
Make sure the Opus file is flushed when postprocessing
When postprocessing Opus, flush when closinf the file too
Be a bit more tolerant of ICMP errors on RTP in SIP, SIPre and NoSIP plugins (fixes #1095)
Make close_pc and end_session calls truly asynchronous (see #1109)
Fixed nits and typos
Handle RTCP for all remote SSRCs, including video simulcast
Merge branch 'master' into rtcp-ssrc
Several RTCP related changes
Reduced unneeded verbosity when overwriting SSRCs in RTCP
Merge pull request #1110 from meetecho/rtcp-ssrc
Removed unneeded check
Fixed some nots and typos
Merge pull request #1111 from Demivan/master
Got rid of GList traversal to calculate lastsec bytes
Fixed event sent to handlers related to simulcast video streams
Improved management of incoming NACKs
Fixed typos
Marshall Quander (3):
LWS_MAX_SMP=1 in README LWS build instructions
Add Rust plugin wrapper to resources page
Log error codes for SCTP-related errors
Olle E. Johansson (1):
Update README for certificates
Piter Konstantinov (4):
Add optional exchange for RabbitMQ transport
RabbitMQ event handler
Fixes after review
Sample config file fixes
Pranay Surana (1):
Update janus_audiobridge.c
suranapranay (1):
fixing media_hangup deadlock
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