[Pkg-voip-commits] [janus] annotated tag upstream/0.2.3 created (now df843de)

Jonas Smedegaard dr at jones.dk
Mon Jun 12 19:00:06 UTC 2017


This is an automated email from the git hooks/post-receive script.

js pushed a change to annotated tag upstream/0.2.3
in repository janus.

        at  df843de   (tag)
   tagging  713c163f109ff3ac1fa6a6e9a3fd30df0ca1aea9 (commit)
  replaces  upstream/0.2.2+dfsg
 tagged by  Jonas Smedegaard
        on  Mon Jun 12 16:34:28 2017 +0200

- Log -----------------------------------------------------------------
Upstream version 0.2.3
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Alex Smirnov (2):
      fix #871 detaching of already detached plugin
      mode plugin detached status check before request construction

Alexander Clark (8):
      Merge remote-tracking branch 'refs/remotes/meetecho/master'
      set Access-Control-Max-Age header
      Merge remote-tracking branch 'refs/remotes/meetecho/master'
      Merge remote-tracking branch 'refs/remotes/meetecho/master'
      Merge remote-tracking branch 'refs/remotes/meetecho/master'
      Merge remote-tracking branch 'refs/remotes/meetecho/master'
      listDevices support for custom GUM parameters
      prevent unintended recursion

Ancor Gonzalez Sosa (1):
      bower: external webrtc-adapter instead of bundled one

Fabrizio Bertone (5):
      open RTCP port for RTSP streams
      open rtcp port for rtsp streams (change janus_streaming_rtsp_parse_sdp signature)
      bind random ports in rtsp
      RTSP: bind random RTP port, adjacent RTCP port
      fix TextRoom name in launching handler error

Flavio Grossi (1):
      add ssl support for the rabbitmq transport

Giordano Cardillo (2):
      Added audio to screen sharing
      FIX on adding stream. Chat recordings were messed up.

Gábor Tóth (1):
      fix for unititialized event handler close

Johan Ouwerkerk (1):
      Fix: broken check for whether or not data channels were actually requested.

Jonas Smedegaard (1):
      New upstream version 0.2.3

Joshua Dickson (1):
      change packet queueing log level

Lorenzo Miniero (139):
      Several new helper methods for SDP utilities
      Merge branch 'master' into sdputils-pt2
      Merge branch 'master' into sdputils-pt2
      Merge branch 'master' into sdputils-pt2
      Better integration of new IP tools in Janus core and plugins
      Merge branch 'master' into iputils-usage
      Merge branch 'master' into sdputils-pt2
      Changed API of janus_network_detect_local_ip to better fit ip-utils, and added wrapper (integrated in janus.c and janus_sip.c) that returns an allocated string
      Fixed indentation issue
      Further cleanup of ip-utils related code
      Switched to version 0.2.3
      Fixed a couple nits in the README
      Made the DTLS set-timeout feature more evident in both README and configure
      Merge pull request #803 from meetecho/iputils-usage
      Merge branch 'master' into sdputils-pt2
      Add ICE Lite status to the Janus info
      Make sure reply is initialised (TextRoom plugin)
      Make SRTP errors way less spammy (unless debug=7 is used)
      Integrated SDP utils in Record&Play plugin too
      Removed leak in AudioBridge plugin
      Merge pull request #784 from meetecho/sdputils-pt2
      Updates Janus manpage to addres recent cmdline argument addition (fixes #812)
      Install only headers needed by third party components (plugins), and add proper subdirectories for API headers (plugins, transports, events) (fixes #811)
      Make sure apierror.h is installed with the headers too
      Removed unused code
      Merge pull request #810 from neilkinnish/fix/record-rename
      Fixed typo introduced in #810
      Make sure config.h is installed with the headers too
      Stripped ICC profile from arrow image in demos (fixes #815)
      Make sure cur_seq is a valid pointer
      If a Streaming recorder fails, return an error
      Fixed typo when matching user agent (introduced in #808)
      Return a valid event after an AudioBridge leave (fixes #816)
      Fixed typos in docs
      Updated date in docs
      Fixed broken re-INVITE management in SIP plugin
      Fixed compilation warning in ice.c (fixes #818)
      Unlink UnixSockets when shutting down (fixes #819)
      Added Debian repo to resources
      Removed usages of /tmp, and used placeholders where relevant (see #814)
      Moved IP self-detect of SIP plugin outside of the config parse code
      Merge pull request #821 from stormbkk87/mqtt-auth-fix
      Fixed crash when disabling non-RTP mountpoints
      Don't print errors if transports are simply disabled by configuration
      Fixed broken link in EchoTest demo (fixes #827)
      Fixed error in HTTP module (reported on group)
      Clarify whether a room (AudioBridge, VideoRoom, TextRoom) is PIN-protected when answering a list request (fixes #826)
      Check PIN in TextRoom, if available (fixes #830)
      Option to add temporary extension while recording
      Make sure RTCP buffers are reset before they're written to (fixes #833)
      Make sure new_head is not NULL before accessing it
      Merge pull request #834 from flaviogrossi/add_rabbit_tls_support
      Reset participation type after a leave in the VideoRoom
      Better management of VideoRoom kick
      Merge pull request #841 from fbertone/open-rtcp-port
      Made janus_streaming_rtsp_parse_sdp static
      Updated configure.ac requirements: g_clear_pointer was added 2.34 and not 2.32
      Added git commit + compile time information to the Janus logs
      Merge pull request #843 from Sean-Der/master
      Merge pull request #844 from fbertone/random-rtsp-ports
      Fixed broken section titles in README
      Merge pull request #850 from Sean-Der/master
      Fixed fmtp parsing in Streaming plugin for RTSP
      Support for on-hold in SIP plugin
      Merge branch 'master' into on-hold
      Optional forcing of private IDs for subscriptions for better kick in VideoRoom
      Fixed occasional deadlock when kicking
      Fixed (optional) rabbitmq-c compilation steps in README (see #847)
      Modified rabbitmq-c instructions to better adhere to official guidelines
      Reverted instructions (hopefully for the last time)
      Merge pull request #760 from mirkobrankovic/videoroom-audiolevel-event
      Small post-merge fixes to AudioBridge and VideoRoom code
      Attempt to fix race condition when kicking publishers and their subscriptions
      Fixed bug where session_timeout=0 config setting was now honored
      Cleaned up warnings from #760
      Merge pull request #846 from meetecho/commit-date
      Merge pull request #854 from meetecho/on-hold
      Reconnect RTSP stream if it goes down (Streaming plugin)
      Aligned with recent changes in master
      Fixed generation of version.c when not on a git repo (fixes #860)
      Log when we managed to reconnect
      Don't do anything until the RTSP stream is reconnected
      Fixed deadlock when connecting to an RTSP server at creation time fails
      Fixed broken AudioBridge/VideoRoom PeerConnections from Firefox Nightly, due to new checks on extmap direction
      Merge pull request #856 from meetecho/requite-pvtid
      Merge pull request #861 from meetecho/rtsp-reconnect
      Added versioning details to /info results
      Merge pull request #859 from Sean-Der/master
      Fixed broken indentation (spaces)
      Use MHD_USE_AUTO if libmicrohttpd is recent enough
      Specify MHD_USE_AUTO_INTERNAL_THREAD or MHD_USE_POLL_INTERNALLY explicitly, when doing MHD_USE_THREAD_PER_CONNECTION
      Added elixir-janus to the Resources documentation page
      Reverted MHD_USE_TLS back to MHD_USE_SSL
      Merge pull request #870 from meetecho/mhd-fix
      Added audio output device selection, if available, to devicetest demo (fixes #869)
      Don't detach handles when you're destroying a session in janus.[nojquery].js
      Merge pull request #873 from hijaq/fix-detached-plugin
      Fixed no_media_timer log line
      Make sure s-values in SDP are always simple (fixes #874)
      Detect RTCP BYE messages and hangup in case
      Attempt to provide an issue template as a guideline for issue reports
      Moved CONTRIBUTING.md to subfolder
      Fixed typo when talking about Opus complexity in AudioBridge
      Fixed typo in HTTP plugin (fixes #882)
      Merge pull request #880 from cmacq2/fix-broken-data-sctp-check
      Merge pull request #883 from fbertone/patch-1
      Added new project to the resources documentation page
      Made username mandatory when registering, guest or not (fixes #885)
      New configure options to disable all plugins
      Fix on nasty race condition (SRTP contexts for outgoing traffic)
      Merge branch 'master' into sendthread-fix
      Removed broken warning on lack of VP9 support in postprocessor (see #878)
      Merge pull request #872 from meetecho/output-device
      Merge pull request #893 from joshdickson40/log-fix
      Merge pull request #886 from meetecho/disable-all
      Merge branch 'master' into sendthread-fix
      Make sure the agent for a handle is not created twice
      Merge pull request #894 from joshdickson40/ontrack-fix
      Allow custom headers in REGISTER too, in SIP plugin
      Reverted ontrack/onaddstream replacement, as it broke Firefox (see #894)
      Merge branch 'master' into sendthread-fix
      Merge pull request #887 from meetecho/sendthread-fix
      Small RTCP fixes
      Merge branch 'master' of github.com:meetecho/janus-gateway
      Merge pull request #903 from agclark81/master
      Merge pull request #904 from chunmeng/master
      Better management of hangup conditions
      Fixed extra spaces
      Send an event back when a DTMF has been sent via SIP INFO as requested
      Send SR way less frequently
      Talking / stopped talking events, instead of talking-only repeats
      Allow handles to force BUNDLE/rtcp-mux via API without waiting for negotiation
      Merge pull request #917 from ancorgs/external_adapter
      Merge pull request #910 from meetecho/talking
      Merge pull request #911 from tgabi333/patch-1
      Merge pull request #912 from meetecho/capabilities
      Fixed check in janus.js that was breaking the new Safari WebRTC support
      Merge pull request #915 from giordanocardillo/audio-in-screenshare
      Unmute screensharing video, when acting as a viewer (see #915)

Mirko Brankovic (6):
      Fixed indentation and changed number of packets to 150 (3s) and acumulated dBov 3000 (~20 average)
      Fixed indentation and changed number of packets to 150 (3s) and acumulated dBov 3000 (~20 average)
      * Proper use of level var of dBov
      Merge branch 'master' into videoroom-audiolevel-event
      Merge pull request #1 from mirkobrankovic/videoroom-audiolevel-event
      Merge branch 'master' into videoroom-audiolevel-event

Neil Kinnish (2):
      Fixes missing directory (full path) when renaming from temp extension
      Fixed, issues around my original commit

Sean DuBois (3):
      Use g_file_get_contents instead of fopen/fseek/fread, using the former causes AddressSanitizer to report a heap-buffer-overflow
      Fix leak in janus_sctp_incoming_data, callback used by usrsctp_socket didn't free passed data
      Allow video/audio port of 0 for RTP streaming, also add 'audio_port' and 'video_port' to find the randomly generated port

StormBkk87 (3):
      Merge remote-tracking branch 'meetecho/master'
      Username/Password fix
      Fixed memory leak username/password

agclark81 (1):
      updating indents to use tabs instead of spaces

chunmeng (1):
      Fix sdp parsing for audio in streaming plugin

joshdickson40 (7):
      Merge remote-tracking branch 'meetecho/master'
      Merge remote-tracking branch 'meetecho/master'
      Merge remote-tracking branch 'meetecho/master'
      Merge remote-tracking branch 'meetecho/master'
      Merge remote-tracking branch 'meetecho/master' into ontrack-fix
      update webrtc-adapter 3.1.5 -> 3.4.3
      use ontrack instead of onaddstream

mirkobrankovic (15):
      * removed reference to old method
      added event notification
      Merge remote-tracking branch 'upstream/master'
      FastFWD + fixed noted from elminiero
      Fixed removed data channel code
      Configurable parameters for audio level in room
      Only in case of audio packets
      Merge branch 'master' of https://github.com/meetecho/janus-gateway into videoroom-audiolevel-event
      Audio room event
      Switched from User->Display to user_id
      * config file samples removed
      Covered > 0 cases for int values
      * moved example in sample file to proper place
      Added check if this is not video packet
      Changed the shape of event

root (5):
      Added aditional function to get dBov audio level so that average can be calculated for X amount of packets.
      Merge branch 'videoroom-audiolevel-event'
      Event is working
      Added aditional function to get dBov audio level so that average can be calculated for X amount of packets.
      Event is working

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