[Pkg-voip-commits] [janus] annotated tag upstream/0.2.2 created (now cce1222)
Jonas Smedegaard
dr at jones.dk
Tue Mar 14 10:41:56 UTC 2017
This is an automated email from the git hooks/post-receive script.
js pushed a change to annotated tag upstream/0.2.2
in repository janus.
at cce1222 (tag)
tagging 5c3c02950a11aca5b7fd18fb38004c8d7960a894 (commit)
replaces upstream/0.2.1+dfsg
tagged by Jonas Smedegaard
on Sun Mar 12 14:12:57 2017 +0100
- Log -----------------------------------------------------------------
Upstream version 0.2.2
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Aaron Hamilton (1):
Remove ini_config from configure.ac, since it's not required.
Akagi201 (6):
fix compile failed on Mac
add compile on macOS to README
update README
update README
update README
support for old macOS
Alessandro Toppi (1):
Sequential loading of required JS scripts
Alex Smirnov (5):
Fix getting min/max values in janus_rtp_header_extenstion_parse_playout_delay
Check participant->room before using it at janus_videoroom_leave_or_unpublish
Check for videoroom listener at janus_videoroom_incoming_rtcp
Fix setting a XHR.status property that has only a getter
Fix missed jquery $.ajax to nojquery Janus.ajax
Alexander Clark (1):
support for setting an iceTransportPolicy
Ancor Gonzalez Sosa (4):
Added data channels support to videoroom plugin (MCU)
Prevent bower to use a too recent adapter.js
Set the limit of open files in systemd unit example
Use the bundled adapter.js instead of an external dependency
Andreas Girgensohn (17):
emacs.el to set the Janus coding style in Emacs
new JANUS_VALIDATE_JSON_OBJECT macros
new JANUS_CHECK_SECRET() and JANUS_CHECK_PIN() helper macros for plugins
use JANUS_VALIDATE_JSON_OBJECT() and related helpers in all plugins
Check out_stats.video_packets when dealing with video.
In videoroom, protect recorders with a mutex to avoid race conditions.
In SIP, protect recorders with a mutex to avoid race conditions.
Don't warn in response to a "detached" event because that situation happens when detaching from JavaScript.
Add calls to janus_videoroom_message_free
Reduce code duplication in videoroom plugin with several new functions.
New function janus_videoroom_recorder_create. Set the rejected mline at the end.
Validate request parameters in janus.c with new macro
janus_videoroom_access_room returns error_cause. New functions janus_videoroom_sdp_a_format, janus_videoroom_sdp_v_format.
Combine log messages for codec mismatch.
Handle LWS_CALLBACK_WSI_DESTROY
Assign new value before freeing old value to avoid state with freed value.
Remove code duplication between regular and admin web sockets.
Andrei Nesterov (1):
Added support of MQTT transport
Benjamin Trent (1):
session pointer not set to NULL after free in videoroom session free function, corrected it
Bojacob (4):
Ability to configure virtual host, username, and password for RabbitMQ
free allocated memory and move up credentials to be used by either admin or janus api
bloody semicolon
fix indents
Chad Furman (2):
configurable screensharing framerate
Update janus.js
Chad Furman (2016-2020) (3):
Revert "Update janus.js"
made same changes to nojquery
tabs not spaces in nojquery
Chad Phillips (9):
Fix typo in textroom plugin log message for list command
add custom libnice install instructions to README
Allow updating display value via configure command
include display in parameter validation, get rid of extra if statement
remove recommendation to install newer version of libnice
enhancements to BoringSSL handling in autoconf
fix path typo in README
free old display, make setting new display more compact
add janus-event-server to resources page
Computician (10):
fixed session cleanup to remove sessions from the hash table, fixed mutex locking in room destroy message case
Adding api request response for listing videorooms and determining if a videoroom exists or not
Fixing typo
Trying to correct bug where when a room is destroyed and participants try to leave the room at almost the exact same time, there are seg faults
still having overruns, trying to add a room mutex so that rooms are safe from being destroyed while people are accessing them...may need to only protect certain room elements and not the whole shebang
making change so that room status is checked on each iteration, and also so that room participants are protected
room insertion was in the wrong place in create...moved it up so that the updated list contains the newly created room
rtp_listener feature added for videoroom plugin
corrected indention, moved to create sockaddr_in structures for individual streams, both media types are now not mandatory, and changed to rtp_forward
Forgot to change OPUS back to actually being opus
Damon Oehlman (1):
Added gstreamer 1.0 command variant
David Rajchenbach-Teller (1):
Resolves #569 On MacOS X, libraries can be in /opt/local/lib
Davide Bertola (14):
recordplay: avoid stopping if already stopped
recordplay: fix wrong error message
recordplay: allow client to specify filename (optional)
better use g_snprintf
recordplay: send rtcp rembs every second
recordplay: uniform session variable names
recordplay: send rtcp pli on packet loss
recordplay: also send rtcp fir on packet loss
recordplay: make remb ramp-up faster
recordplay: add call to set video bitrate cap
recordplay: change js bitrate value to bits/s
recordplay: add plugin api to set keyframe interval
recordplay: implement “configure” api
recordplay: handle ‘slow_link’ event on the client
Dustin Oprea (3):
Aptitude packages missing libopus-dev.
Naming fix in help output (GGO and README).
Removed and ignored auto-gen'd cmdline files.
Eduardo Barbosa (1):
Added supervisor sample to the documentation
Emmanuel Riou (2):
fix MACOS endianness issue (due to lack of standart environment variables) + make janus-pp-rec compile on MACOS
Merge remote-tracking branch 'upstream/master' into pullreq
Evan Coury (1):
Fix Fedora package name for pkgconfig
Fabrizio Bertone (1):
Update README.md
Ferdinand Full (3):
Add package.json
Add files array to package.json to only install client side scripts
Add janus.nojquery.js to files array at the package.json
Filip Jenicek (3):
Send DTMF tones using SIP INFO messages
Send DTMF tones using SIP INFO messages - configurable duration
Send DTMF tones using SIP INFO messages - use inband in the demo
Florian P. Nierhaus (6):
fix read cert_pem for REST https
off by one buffer overflow
protect access to freed janus_websockets_client with old_wss_mutex
Merge branch 'modular-transports' into fpn_double_free_websocket
fixup patch according to janus coding style
msg->handle->plugin_handle may not exist when message is handled
Giacomo Vacca (2):
autogen.sh requires autoconf package
Update README.md
Graeme (1):
Update Ubuntu/Debian .deb install
Hubert Figuière (1):
Fix test pipeline for the streams plugin
Jack Leigh (15):
Support older libavcodec versions
Another libavcodec version #if
We only send the new publisher here not all publishers
ice: Give 'container' meaningful names
postprocessing: Use top-level debug header
postprocessing: Fix old-style function definitions
postprocessing: Const correctness
postprocessing: Return type fix
Only access the global stop variable atomically
Change HAVE_WS to HAVE_WEBSOCKETS
Call handle handle and plugin_session plugin_session
Staticise plugin globals
Autotoolize build system
Convert sessions watchdog to use a glib mainloop
Install certs
Jay Ridgeway (7):
init buffered logging
formatting
tabs are from the devil
remove timed wait, reduce locking, tabs
remove more glib
free buffers and synchronization fixes
ditch vasprintf from glib printf routines
Johan Ouwerkerk (3):
Add utility functions to map a network device name or IP address to a network interface.
Permit user configurable network device selection for listening to multicast RTP and RTSP streams in Janus.
Support the datasctpnetwork configuration option for RTP streams.
Jonas Smedegaard (1):
New upstream version 0.2.2
Kishan Lachhani (1):
fix typo
Leon Klingele (2):
Fix Janus.isWebrtcSupported
Remove redundant whitespaces
Lets_Vape (1):
Fix videoremote id for spinner
Lorenzo Miniero (1044):
Fixed install.sh script for Ubuntu
Merge pull request #19 from dsoprea/master
Merge pull request #62 from leighman/cleanup
Merge pull request #63 from leighman/master
Merge pull request #64 from mrauhu/fix-screensharing-chrome-34+
Merge pull request #68 from leighman/misc
Fixed wring ifdef in janus.h that caused broken compile with rabbitmq disabled
Added missing ifdef for optional rabbitmq-related code
Merge pull request #79 from nowylie/master
Merge pull request #82 from ancorgs/data_in_videoroom
Merge pull request #86 from giavac/master
Merge pull request #85 from Computician/master
Merge pull request #92 from ploxiln/minor_fixes
Merge pull request #87 from Computician/master
Merge pull request #94 from megawac/deps
Merge pull request #99 from ploxiln/rammitmq
Merge pull request #103 from giavac/master
Merge pull request #102 from meetecho/nack-timer
Merge pull request #104 from ploxiln/more_gitignore
Merge pull request #105 from gatecrasher777/master
Merge pull request #106 from Computician/master
Merge pull request #111 from Computician/master
Merge pull request #113 from mporrato/wip
Merge pull request #116 from ploxiln/nice_debug
Merge pull request #119 from ploxiln/echotest_button_ff
Merge pull request #120 from ploxiln/fix_file_copy_120
Merge pull request #122 from mingewang/sip
Merge pull request #123 from Computician/master
Merge pull request #114 from leonuh/master
Fixed padding (added in #124 related commit)
Merge pull request #142 from nowylie/master
Merge pull request #139 from scottmas/master
Merge pull request #125 from yultide/master
Merge pull request #146 from nowylie/master
Merge pull request #152 from ploxiln/firefox_size_constraints
Merge pull request #155 from davibe/master
Merge pull request #158 from uxmaster/master
Merge pull request #157 from davibe/master
Merge pull request #163 from EvanDotPro/patch-1
Merge pull request #164 from saghul/sip_from
Merge pull request #168 from saghul/sip_fixes
Merge pull request #169 from saghul/sip_fixes1
Merge pull request #170 from saghul/sip_fixes2
Merge pull request #171 from ploxiln/admin_memleak
Merge pull request #172 from LetsVape/spinnerfix
Merge pull request #176 from saghul/sip_fixes3
Merge pull request #177 from saghul/sip_fixes4
Merge pull request #178 from saghul/sip_features1
Merge pull request #179 from ploxiln/plugin_message_leak
Merge pull request #174 from davibe/recordplay_bitrate
Merge pull request #180 from ploxiln/no_allocate_keepalive
Merge pull request #181 from saghul/sip_fixes5
Merge pull request #182 from saghul/sip_features2
Merge pull request #183 from ploxiln/streaming_resp_restore
Merge pull request #184 from saghul/sip_features3
Merge pull request #187 from saghul/sip_fixes6
Merge pull request #186 from saghul/fix_getifaddrs
Merge pull request #198 from mpromonet/master
Merge pull request #199 from ploxiln/handle_sigterm
Merge pull request #200 from xenyou/master
Merge pull request #216 from saghul/sip_fixes7
Merge pull request #204 from ploxiln/janus_log_flatten
Merge pull request #220 from ploxiln/double_trinary
Merge pull request #218 from ploxiln/trickle_error_newline
Merge pull request #219 from ploxiln/http_logging
Merge pull request #221 from ploxiln/ice_logging_tinyfix
Merge pull request #224 from ploxiln/session_cleanup_quiet
Merge pull request #225 from ploxiln/waiting_log_once
Merge pull request #226 from ploxiln/feature_logging_consolidation
Merge pull request #227 from ploxiln/logging_ice_added_looping
Merge pull request #228 from ploxiln/nack_logging
Merge pull request #229 from ploxiln/loglevel_followups
Merge pull request #217 from mpromonet/master
Merge pull request #232 from ploxiln/still_cleaning_log
Merge pull request #230 from ploxiln/slowlink_count_period
Merge pull request #222 from ploxiln/remote_candidate_logging
Merge pull request #234 from ploxiln/slowlink_retransmits
Merge pull request #235 from ploxiln/http_shutdown
Merge pull request #238 from ploxiln/redo_nack_gen
Merge pull request #248 from saghul/sip_dovideo
Merge pull request #253 from meetecho/certs-1024
Merge pull request #255 from Computician/master
Merge pull request #262 from khejing/samplerate
Merge pull request #264 from saghul/ws_subprotocol
Merge pull request #265 from saghul/echo_error
Merge pull request #270 from saghul/sip_decline_fixes
Merge pull request #271 from saghul/sip-ha1-secret
Merge pull request #276 from saghul/sip_no_register
Merge pull request #282 from ancorgs/fix_bower
Merge pull request #275 from saghul/sip-demo-ha1
Merge pull request #296 from saghul/registration-failed-reason
Merge pull request #295 from saghul/sip-remote-identity
Merge pull request #301 from saghul/usrsctp-repo
Merge pull request #303 from saghul/extra_clean
Merge pull request #305 from saghul/no-build-static
Fixed issue in janus_dtls_bio_filter_ctrl (issue #308)
Merge branch 'master' into modular-transports
Fix in management of HTTP URL splitting (issue #309)
Fix in management of HTTP URL splitting (issue #309)
Fixed wrong verbosity level added in previous commit
First take at a daemon/service documentation page (see #306)
Added option to disable colors in logging (issue #304)
Fix management of new UDP/TLS/RTP/SAVPF rewriting in SIP plugin
Fixed issue of sending busy that also hanged up the current call in SIP plugin (see issue #312)
Better management of issue #312, new missed_call event in SIP plugin, and fixed missing registration_failed event handler in SIP demo
Added upstart sample to the documentation
Documentation on how to effectively debug Janus
Parse SSRC used for retransmissions by Chrome
Merge branch 'master' into modular-transports
Fixed issue when destroying streaming mountpoints
Merge branch 'master' into modular-transports
Merge pull request #313 from saghul/init-docs
Changed default value of hangingup when creating plugin sessions to false
Merge branch 'master' into modular-transports
Merge pull request #314 from ancorgs/limit_no_file
Fixed a couple of data channels potential leaks, and addressed potential overflow when forwarding data channel messages in plugins (see issue #302)
Merge branch 'master' into modular-transports
First attempt at getting rid of the increasing delay in audiobridge rooms when network is shaky for a few users
Merge pull request #266 from mpromonet/master
Merge pull request #318 from saghul/apisecret-typo
Added configuration files to .gitignore
Attempt to fix occasional issue with websockets and session timeouts (see issue #307)
Merge pull request #317 from meetecho/audiobridge-delay
Changed recordings header to contain more info (as of now, mostly codecs and created/first written times), using a JSON format so that it can be extended in the future (old recordings can still be read/played)
Only unlock the audiobridge peek buffer after mixing has been done (may help issue #319)
Merge branch 'modular-transports' of github.com:meetecho/janus-gateway into modular-transports
Merge branch 'master' into modular-transports
Fixed error in updating configuration file (was replaced by HTML)
Merge branch 'master' into modular-transports
Fixed occasional multiple events in reply to the same request
Added flags to check whether offer and/or answer have been received
Added new token based authentication mechanism for the Janus API
Integrated token and apisecret in janus.js
Fixed typo in audiobridge plugin (issue #324)
Merge branch 'master' into modular-transports
Added a way for plugins to validate API secret and tokens through the core, when needed (e.g., HTTP long polls)
Brief example on how to use API secret and tokens in janus.js
Make core more conservative when checking plugin sessions
Updated year in demos
Added a new Resources page to the documentation
Merge pull request #322 from meetecho/got-answer
Merge pull request #326 from ploxiln/log_color_select_refactor
Added some timing related details to the handle info in the admin API
Merge branch 'master' into modular-transports
Asynchronous trickle request management
Fixed detection of private address at startup (issue #331)
Merge pull request #332 from MichaelB76/master
Merge pull request #334 from saghul/buffer-size
Merge pull request #330 from meetecho/pending-trickles
Don't allocate fake attribute for sendrecv hack every time
New destroyOnUnload parameter in janus.js to override onbeforeunload behaviour
Merge pull request #336 from saghul/fix-compilation
Merge branch 'master' into modular-transports
Fixed missing mutex unlock
Merge branch 'master' into modular-transports
Fixed link to libsrtp in both README and docs
Merge branch 'master' into modular-transports
Fixed recording of SIP calls when filename is provided
Merge branch 'master' into modular-transports
Merge pull request #339 from saghul/fix-ice-lite
Merge branch 'master' into modular-transports
Added new ICE 'enforce' list, to specify the only interfaces to use for gathering candidates
Updated janus.cfg sample to address the new ICE enforce list
Removed unused public_ip setting from janus.cfg sample
Allow IP addresses to be passes to the ICE enforce list
Merge pull request #311 from MathRobin/master
Use different handlers for ws and sws (issue #340)
Use different handlers for ws and sws (issue #340)
Don't add prflx candidates to the SDP offer/answer
Merge branch 'master' into ice-enforce-list
Restored the old public_ip setting as a new nat_1_1_mapping setting (-1 on the command line), to clarify what it is for and when it should be used
Add optional BoringSSL support via configure
Fixed occasional inability to remove RTP forwarders in videoroom plugins
Require a valid certificate key when staring Janus
Merge pull request #341 from meetecho/ice-enforce-list
Merge branch 'master' into modular-transports
Use 'checkout' instead of 'fetch origin' for BoringSSL
Switched inet_ntoa to inet_ntop (new resolving method in utils)
Changed debugging for skipped candidates from warning to verbose
Merge branch 'master' into modular-transports
Free addrinfo after it's been used
Merge branch 'master' into modular-transports
Merge branch 'master' into boringssl-support
Fixed echo test data channels forwarding (last character cut away)
Merge pull request #342 from bebo/fpn-modular-transports
Converted memory allocations to GLib ones, and fixed a couple of leaks
Converted memory allocations to GLib ones, and fixed a couple of leaks
Merge branch 'modular-transports' of github.com:/meetecho/janus-gateway into modular-transports
Merge branch 'master' into modular-transports
Added the possibility to specify an optional PIN to access streaming mountpoints and audiobridge/videoroom conference rooms
Merge branch 'master' into modular-transports
If both API secret and token auth mechanism are enabled at the same time, either one that is provided and valid is fine
If both API secret and token auth mechanism are enabled at the same time, either one that is provided and valid is fine
Merge branch 'master' into modular-transports
Merge pull request #338 from MichaelB76/master
Merge branch 'master' into modular-transports
Merge pull request #344 from saghul/sip-reinvite-missed-call
Merge branch 'master' into modular-transports
Added method to save a configuration object to file
Merge branch 'master' into modular-transports
Merge pull request #345 from saghul/sip-manual-reinvite
Merge branch 'master' into modular-transports
Merge pull request #346 from saghul/editorconfig
Merge branch 'master' into modular-transports
Fixed a couple of compilation warnings
Added a comment header with time for saved configuration files
Use minimum FPS as the info to put in WebM header when postprocessing
Merge branch 'master' into modular-transports
Add info on when the handle was created to the admin API
Merge branch 'master' into modular-transports
Merge pull request #347 from saghul/fix-sip-handle-reset
Merge branch 'master' into modular-transports
Merge pull request #348 from saghul/sip-sdp-leak
Merge branch 'master' into modular-transports
Merge pull request #349 from bebo/fpn_fixes
Further check before pushing plugin session event
Merge branch 'master' into modular-transports
Fixed typo in writing recording header
Merge branch 'master' into modular-transports
Merge branch 'master' into boringssl-support
Added possibility to limit scope of auth tokens to specific plugins
Merge branch 'master' into boringssl-support
Fixed token/plugin check when API secret is involved
Merge branch 'master' into modular-transports
Merge branch 'master' into modular-transports
Merge pull request #343 from meetecho/boringssl-support
Merge branch 'master' into modular-transports
Merge pull request #350 from bebo/fpn_double_free_websocket
Add a new helper method to get the system real time, besides the monotonic one
Use janus_get_real_time instead of janus_get_monotonic_time for a few things
Merge branch 'master' into modular-transports
Added admin API methods to dynamically toggle log colors and timestamps
Added admin API methods to dynamically toggle log colors and timestamps
Merge branch 'master' into modular-transports
Return whether API secret and token mechanism are enabled in the server info
Return whether API secret and token mechanism are enabled in the server info
Merge branch 'master' into modular-transports
New UI and features for the admin API web demo
Merge branch 'master' into modular-transports
Add transports to the new admin API web demo
Show docs creation/update time in html pages
Merge branch 'master' into modular-transports
Allow enter to be used in admin web UI for new tokens
Merge branch 'master' into modular-transports
Decreased verbosity for some lines (info to verb), and added call to nice_agent_remove_stream when enforcing bundle/rtcp-mux (see #154)
Merge branch 'master' into modular-transports
Merge pull request #354 from tuijldert/master
Merge branch 'master' into modular-transports
Merge pull request #356 from tuijldert/master
Allow applications to provide their own MediaStream to janus.js
Merge branch 'master' into modular-transports
Updated documentation
Merge branch 'master' into modular-transports
Skip packets that are too large to be RTP in the post processor
Fix postprocessing when last packet is broken
Merge pull request #359 from amnonbb/rtp-forward
Merge branch 'master' into modular-transports
Make sure rec_dir is honored even when providing a filename in a videoroom configure request (issue #357)
Merge branch 'master' into modular-transports
Fixed missing CR in SDP generation
Merge branch 'master' into modular-transports
Added new console wrappers to janus.js, and bound them to debug level in init (see #292)
Merge branch 'master' into modular-transports
Fill gaps in audio recordings with silence, when postprocessing
Merge branch 'master' into modular-transports
Fixed detection of Opus and VP8 payload types in some cases
Merge branch 'master' into modular-transports
Fixed detection of Opus and VP8 payload types in some cases
Merge branch 'master' into modular-transports
Removed unneeded extra debugging
Merge branch 'master' into modular-transports
First attempt at getting Edge and Janus to talk to each other
Use code 480 in case a SIP decline is caused by a denied permission on WebRTC
Don't start data thread until ICE connectivity has been established
Merge branch 'master' into modular-transports
Merge branch 'master' into janus-edge
Prettier admin UI for handle info
Merge branch 'master' into modular-transports
Merge branch 'master' into janus-edge
List discovered (prflx) remote candidates when querying the admin API
Merge branch 'master' into modular-transports
Merge branch 'master' into janus-edge
Make sure trickle candidates are not passed to the stack until we have both offer and answer ready
Use MediaStreamTrack.stop() (see #363)
Merge branch 'master' into janus-edge
Merge branch 'master' into modular-transports
Merge branch 'master' into spinning-threads
Fixed occasional failure to start ICE when answering from a plugin
Verbosity change for trickle queueing message
Updated references to videoroom in the demos, and clarified it's an SFU and not MCU
Merge branch 'master' into modular-transports
Merge branch 'master' into janus-edge
Merge pull request #360 from meetecho/janus-edge
Added autorefresh checkbox for handle info in admin API web demo
Fixed parsing of fingerprints so that they can be different per each stream
Merge branch 'master' into modular-transports
Merge branch 'master' into spinning-threads
Fixed missing stream/component IDs in janus_ice_component
Merge branch 'master' into spinning-threads
Merge branch 'master' into modular-transports
Force dummy candidate for unneeded RTCP components when rtcp-mux has been negotiated
Changed IP for dummy candidate to 127.0.0.1
Added an UDP server (random port) to act as blackhole for keepalives from unneeded RTCP components
Merge pull request #368 from ploxiln/nat_1_1_and_stun
Merge branch 'master' into modular-transports
Merge branch 'master' into spinning-threads
Added note about better logging when launching Janus via systemd
Fixed typo in docs
Merge branch 'master' into spinning-threads
Fixed blackhold fd initialization
Merge pull request #362 from meetecho/spinning-threads
Merge branch 'master' into modular-transports
More conservative suggestions for systemd based logging
Fixed access to invalid component when forcing rtcp-mux (issue #370)
Merge pull request #369 from jing3018/postprocessing
Fixed silence packet size written when postprocessing audio
Removed usage of SO_REUSEADDR for UDP sockets
Merge pull request #367 from bebo/fpn_issue_366
Added fix from #366 and #367 to other plugins as well
Merge branch 'master' into modular-transports
Removed dependency from libini_config, changed the way categories are accessed, and added permanent save of configurations in some plugins
Increase plugin API version, although it's the INI stuff that changed
Merge pull request #371 from meetecho/config-save
Merge branch 'master' into modular-transports
Merge pull request #281 from meetecho/modular-transports
Update janus.cfg by removing now useless transport related settings
New methods to mute/unmute audio and video in janus.js
First code to allow Janus to run as a daemon (no logging yet)
Updated version in configure.ac
Fixed typo when handling plugin-originated answer
First attempt at using conditions (wait/signal) instead of sleeps for some of the workers we have (at the moment, echotest plugin only for testing)
Fixed problem of VideoCall plugin not working anymore due to always failing check
Fixed problem of SIP calls not getting working RTP after the first time
Merge pull request #380 from zazabe/fix-js-websocket-listeners
Use g_async_queue_pop to implement conditions automatically
Optional SIPS when registering
Use TAG_IF for NUTAG_SIPS_URL
Merge pull request #386 from meetecho/optional-sips
Don't gather TCP candidates if ICE-TCP support is disabled
Merge branch 'master' into conditions
Use g_async_queue_pop for handler threads in other plugins as well
Use static exit_message for plugin handler threads
Use new audio mute functions in videoroom demo
Use single GAsyncQueue for incoming/outgoing dat channel messages
Reverted unsafe usage of condition in signal handler
Allow admin UI to show either raw or prettified handle info
Removed frequent sleeps in HTTP transport module
Use g_async_queue_pop instead of g_async_queue_try_pop in RabbitMQ transport
Set got_response when mutex is locked
Removed accidentally added video file
Merge pull request #389 from jayridge/bufferedlogging
Allow for console and/or logfile output (to hook to config/cmd line)
Update janus_log_console when initializing
Merge pull request #392 from ploxiln/error_string_leakfix
Configurable logging and daemonization
Merge branch 'master' into conditions
Fixed docs typo
Make sure we don't free the static exit message
Restored sleep-based approach for HTTP transport, and added some fixes as to RabbitMQ
Fix message response condition wait in HTTP transport
Make sure the session is valid and not being destroyed when notifying events (issue #378)
More details when something in OpenSSL fails
Attempt to fix the infamous DTLS decrypt alert error (issue #316)
Merge pull request #396 from xorgy/remove-ini-config-from-configure
Merge pull request #394 from meetecho/dtls-alert-fix
Only modify the ice-udp and ice-tcp libnice attributes if the library supports them
Added option to create/destroy/check PID file
Doxygen documentation for new utils methods
Set default logging level to info
Merge branch 'master' into conditions
Merge branch 'master' into pidfile
Merge pull request #400 from saghul/cfg-fixes
Merge branch 'master' into pidfile
Merge branch 'master' into conditions
Use atexit to always remove the PID file (if any) before leaving
Moved janus_log_destroy to the atexit function
Merge pull request #393 from ploxiln/simplify_log_thread
Fixed message response condition wait in HTTP transport for admin too
Merge branch 'master' into conditions
Merge branch 'master' into pidfile
Merge pull request #399 from meetecho/pidfile
Merge pull request #402 from saghul/sip-missed-call-display-name
Check the result of fscanf wne reading a PID file
Merge pull request #405 from saghul/clean-cfgs
Redirect stdin/stdout/stderr to /dev/null (#407), move the related code to log.c (otherwise log init errors when daemonizing may be lost) and don't enable libnice debugging unless explicitly stated (not even if debug level is 7)
Don't close standard file descriptors, let freopen do that
Merge branch 'master' into conditions
Updated (and prettified) resources page in documentation
Merge pull request #411 from ploxiln/configure_lib_version_checks
Merge branch 'master' into conditions
Merge pull request #413 from ancorgs/update_bower
Added LWS_WITH_OLD_API_WRAPPERS=1 in README for building libwebsockets, to account for the change in their API (issue #410)
Merge pull request #415 from ploxiln/fix_dup_typedef
Destroy libwebsockets contexts at shutdown
Merge branch 'master' into conditions
Don't free the static exit_message message when shutting down plugins
Merge pull request #384 from meetecho/conditions
Merge pull request #417 from ploxiln/config_comment_obo
Added third-party PHP stack to the resources page in the docs
Fixed a couple of memory leaks in the SIP plugin
Merge pull request #418 from sgotti/bundle_correctly_skip_candidates
Initialize timeout value before calling DTLSv1_get_timeout (issue #419)
Make DTLS alert and related events more asynchronous
Fixed typo
Add support for partial writes in websockets transport
Merge pull request #426 from ploxiln/timeout_source_unref
Merge pull request #422 from mtdxc/master
Merge pull request #427 from ploxiln/nat_1_1_stun_warning
Merge pull request #420 from meetecho/async-dtls-alert
Merge branch 'master' into ws-partial-writes
Merge pull request #429 from ploxiln/update_valgrind_supp
Renamed valgrind suppression file (see #429)
Use a shared buffer for outgoing websockets messages
Fixed size of data to write when offset is set
Set the right amount of outgoing data to resume after a partial write (websockets)
Always free original response in websockets module
Fixed broken RabbitMQ transport queues (issue #435)
Merge pull request #430 from ploxiln/ice_send_thread_loop
Merge pull request #424 from meetecho/ws-partial-writes
Updated README to use the right tagged version of libwebsockets
Fixed check of when to load adapter.js (issue with Firefox 46)
Fixed outdated demo description
Fix check when hanging up WebRTC peerconnection
Have the parent wait for an exit code from the child during startup, when daemonizing
Wrap write in a do/while to catch EINTR
Shorter do/while code for EINTR management
Further recommendations on AWS deployment in janus.cfg
Merge pull request #442 from hasbean/master
Fixed a couple of indent typos, and added info for new RabbitMQ config values
Check SIP stack before using it in Sofia callback (issue #447)
Use authuser, when provided, for REGISTER as well and not only for INVITE
Merge pull request #449 from MagicIndustries/master
Added note on upstart in documentation (see issue #455)
Merge branch 'master' of github.com:meetecho/janus-gateway
New transport module (Unix Sockets)
Addressed review by @saghul, and added call to transport_gone on disconnection which was missing
Added check for SOCK_SEQPACKET in configure.ac
Addressed further feedback
Define UNIX_MAX_PATH if undefined, and helper method for creating socket
Fixed portable definition of UNIX_PATH_MAX
Use recvmsg() for incoming messages, and check MSG_TRUNC
Check EAGAIN as well when reading
Merge pull request #453 from phillcz/dtmf_sip_info
Fixed a couple of nits after merging #453
Handle 'unpublished' event even in case no DTLS alert was received
Reset the hangingup flag in plugin when a new negotiation occurs (to account for cases when hangup_media arrives without a prior setup_media)
Better management of poll in streaming plugin
Removed unneeded double check
Check POLLERR and POLLHUP when waiting for child to start (daemon mode)
Merge pull request #443 from meetecho/daemon-pipe
Don't fail if libmicrohttpd is not found and --disable-rest was provided (issue #461)
Avoid ambiguity on number of params for send in janus.js (it's always one, an object)
Avoid ambiguity on number of params for send in janus.js (it's always one, an object)
Better management of poll in SIP plugin too, and fixed default values for sockets
Make fd check more explicit
Simplified and clarified poll checks
Handle POLLERR and POLLHUP in Unix Sockets poll
Reset socketpair after a POLLERR
Added SOCK_DGRAM support to the Unix Sockets transport module
Clarified in the README that Janus will require some configuration files, and that make configs installs a default set of them
Fix use of jQuery method before jQuery is loaded (selective logging)
Merge pull request #463 from meetecho/sequential-js-loading
Merge pull request #460 from meetecho/streaming-pollerr
Autodetect libwebsockets version and use the right API
Move initial declaration outside of the loop
Move initial declaration outside of the loop
First take at RTCP SR/RR in core
Buffer the latest received keyframe in streaming plugin for new viewers
Fixed indentation
Fix to race conditions when shuttind down SIP stack
Restored missing su_home_init
Fixed a couple of memory leaks
Fixed video recording for remote packets in SIP plugin
Merge branch 'master' into sip-shutdown
Merge pull request #469 from meetecho/sip-shutdown
Added optional SDES-SRTP support to SIP plugin
Added alternative version of janus.js without jQuery dependency (see #464)
Allow for optional/mandatory SDES support in SIP plugin
Fixed incoming SIP calls with mandatory SDES, and better SDP generation
Merge pull request #468 from meetecho/streaming-bufferkf
Fix for ID parsing precision in several plugins
Merge branch 'master' into libwebsockets-newapi
Fixed compilation errors when detected version of libwebsockets is >= 1.6
Try handling more than one timestamp reset when post-processing recordings
New hangup request in core, and updated docs
Make getUserMedia errors more explicit (due to JSON.stringify failures)
Removed exceedingly verbose debug line
Removed extra unneeded file
Optional docs and updated README
Merge branch 'master' into libwebsockets-newapi
Fixed typo in janus.js when using API secret and WebSockets
Fixed typo (wrong prefix in 1.6 branch)
Merge branch 'master' into libwebsockets-newapi
Fix check for 1.7 version of libwebsockets
Restore, although commented, the README line on the libwebsockets 1.5 stable branch
Merge pull request #466 from meetecho/libwebsockets-newapi
Merge branch 'master' into pf-unix
Merge branch 'master' into postproc-resets
Merge branch 'master' into rtcp-rr
Fixed typos
Merge pull request #470 from meetecho/sip-srtp
Merge branch 'master' into pf-unix
Merge branch 'master' into rtcp-rr
Merge branch 'master' into postproc-resets
Fixed configure.ac check of websockets
Merge branch 'master' into pf-unix
Merge branch 'master' into postproc-resets
Merge branch 'master' into rtcp-rr
Merge pull request #472 from meetecho/postproc-resets
Documented additional modes of janus-pp-rec
Add number of packets sent/received per medium to Admin API
Merge branch 'master' into rtcp-rr
Fix EchoTest demo for Chrome 50
Device selection in janus.js and new demo
Fixed broken screensharing
Removed unneeded verbosity in listDevices
Use right RTP profile when answering
Merge branch 'master' into rtcp-rr
Fixed management of incoming fragmented WebSockets messages
Reduce unneeded verbosity from latest fix
Fixed broken support for non-trickling endpoints
Added RR/SR termination, and filtering of outgoing packets (REMB generation)
Add an inactive SDP attribute for rejected/inactive media streams
Configurable video codec to force in VideoRoom plugin
Merge branch 'master' into videoroom-codecs
Send BYE after a POLLERR on RTP file descriptors in SIP plugin
Merge pull request #482 from zalmoxisus/patch-1
Merge pull request #478 from meetecho/videoroom-codecs
Removed verbosity of line in SIP plugin
Allow users to provide custom headers to add to a SIP INVITE
Added atomic check to avoid creating ICE thread twice (see #481)
Use json_object_iter instead of json_object_foreach (for older jansson versions)
Merge pull request #486 from marchaase/streaming-rtp-stats
Minor fix for coding convention
Added request to get info on a specific mountpoint
Merge branch 'master' into pf-unix
Merge branch 'master' into rtcp-rr
Merge pull request #488 from pallab-gain/master
Minor fixes for coding convention
Added missing doc info
Fix Sofia SIP when both Record-Route and Contact are there
Better management of missing capture devices (see #489)
Fixed check in latest commit (see #489)
Fix broken VideoCall plugin for recent Chrome versions
Handle padding in RTP when postprocessing
Fixed typo
Fixed typo
Fixed typo
Integrated capture devices fix in janus.nojquery.js as well
Handle media event in janus.js
Documented new media event handler in janus.js
Support for other codecs and formats in recorder and post-processor
Fixed typo (extra debug line causing wrong return)
Fixed typo
Pass right codec information to the recorder in the SIP plugin
Reduced verbosity introduced in latest commit
Bump plugin version to force developers to be aware of API changes
Handle rec_dir even if record is false in VideoRoom plugin
Adjustments to postprocessor logging
Merge branch 'master' into recording-codecs
Adjustments to postprocessor logging
Clarified that the license for the janus.js and janus.nojquery.js JavaScript libraries is MIT and not GPLv3
Some more examples in the deploy documentation
Moved some WS stuff in the deploy documentation
Merge pull request #497 from ploxiln/pp_rec_log_destroy
Merge branch 'master' into recording-codecs
Reject datachannels in AudioBridge plugin, if offered (see #501)
Merge pull request #502 from andreasg123/emacs
Fixed segfault when processing recordings with old header
Fixed VP8 post processing
Merge pull request #490 from meetecho/sip-recordroute
Fixed nits from code review
Merge pull request #467 from meetecho/rtcp-rr
Merge branch 'master' into device-selection
Merge pull request #504 from saghul/rtld_local
Better error notification in case of screensharing errors
Merge pull request #476 from meetecho/device-selection
Merge pull request #506 from ploxiln/trickle_parse_error_null
Merge pull request #507 from mabu-github/buildfix
Added node-janus project to the resources in the docs
Merge pull request #510 from leonklingele/fix-webrtc-supported-check
Merge pull request #511 from leonklingele/remove-redundant-whitespaces
Differentiate screen and window sharing in Firefox
Merge pull request #518 from meetecho/screen-window-share
Merge pull request #516 from ploxiln/videoroom_mutex_unluck
Refactored web pages and demos
Allow configuration of a name for the server instance
New webrtcState event in JS API to be notified when PC goes up/down (and a few updated demos to use this)
Merge pull request #519 from stormbkk87/sip-response-codes
Fixed typos
Merge pull request #458 from meetecho/pf-unix
Updated docs (Unix Sockets and Transport API in doxygen)
Clarified Unix Sockets support in docs
Merge pull request #517 from ploxiln/validate_json_helpers
Further cleanup of SDP when stripping for plugin usage (should fix issue #509)
Merge pull request #515 from jing3018/master
Merge pull request #521 from ploxiln/json_valid_helpers_plugins
Fixed typo in streaming API validation
Merge pull request #528 from andreasg123/ice-video-packets-fix
Allow configuration of HTTP method to use to contact TURN REST API, if enabled
Merge pull request #525 from amnonbb/master
Merge branch 'master' into recording-codecs
Fixed other typo in streaming API validation
Fixed incorrect casting in listforwarders
Merge pull request #535 from saghul/sip-display-name
Event handler plugins, first draft
Merge branch 'master' into event-handlers
Only forward events a handler is subscribed to
Autodetect media from payload type if SSRC wasn't advertized ('Not audio and not video' warning)
Fixed typos
Allow websocket transports to only bind to a single interface and not all
Optimization of core-to-plugin communication
Merge pull request #543 from andreasg123/detach-warn
Make sure the result content is a JSON object
Combined result content check
Merge pull request #545 from jswirl/master
Merge pull request #547 from andreasg123/videoroom-message-free
Merge pull request #531 from andreasg123/videoroom-close-recorder
Merge pull request #533 from andreasg123/sip-close-recorder
Merge branch 'master' into recording-codecs
Increase version to 0.1.1, due to recorder changes
Increase version to 0.1.1, due to recorder changes
Merge pull request #492 from meetecho/recording-codecs
Allow for the events to be disabled completely (broadcast=no in [events] of janus.cfg)
Merge branch 'master' into event-handlers
Max number queue in seconds instead of packets, plus some other RTCP related tweaks
Helper method to create MHD daemon in HTTP transport
Allow HTTP transports to only bind to a single interface and not all
New mutexes to protect recorders in plugins from race conditions (see #531 and #533)
Changed granularity of new Max NACK queue to milliseconds instead of seconds (min is 200ms)
Don't buffer packets if max_nack_queue is 0
Fixed typo
Don't notify about a new publisher until its WebRTC setup has been completed
Fixed postprocessor compile error when FFmpeg version doesn't support VP9
Fixed old NACK check time
Merge pull request #548 from meetecho/unmute-delay
Merge pull request #553 from medialwerk/master
Fixed typo in docs
Merge branch 'master' into plugins-json
Use json_true() and json_false() where we used 0/1 integers or true/false strings
Made media event use boolean as well
Merge pull request #538 from andreasg123/janus-validation
Merge branch 'master' into plugins-json
More events, in particular from other plugins than the EchoTest, and added examples to the sample handler plugin
Merge branch 'master' into event-handlers
Added incoming SIP messages to the events (still missing outgoing)
Remove AudioBridge rooms lazily in the watchdog, to avoid race conditions after a destroy
Added outgoing SIP messages to events (to improve/fix)
Better management of incoming RR
Make naming of new attributes in Admin API less ambiguous
Merge branch 'master' into event-handlers
Added queue and thread for actually broadcasting events to handler plugins
Added some RTCP and media related statistics to the events, triggered each second
Disable event handlers by default; added command line flag to enable them
Fixed broken SS/RR/NACK transmission, due to incorrect filtering
Merge branch 'master' into plugins-json
Merge branch 'master' into event-handlers
Support for recording data channel text messages, and post process them to .srt files
Improved locking in AudioBridge rooms and participants management
Added new plugin (and demo) for datachannel based text broadcasting
Fixed broken automatic REMB in VideoRoom
Allow binding HTTP transports to a specific IP
Make HTTP trasports dual stack, if no interface/IP is specifiec
Allow binding WebSockets transports to a specific IP, and fixed some typos
Handle larger buffers of text when post-processing
Reduce verbosity of processing
Merge branch 'master' into datachan-record
Merge branch 'master' into chat-plugin
Merge branch 'master' into event-handlers
Merge branch 'master' into plugins-json
Use janus_process_error_string when error is a complete string
Merge pull request #561 from saghul/sip-ua
Merge pull request #523 from andreasg123/videoroom-duplicate-code
Initialize variables
Merge pull request #563 from saghul/dtls
Clarified in the docs that the Admin API over WebSockets needs a different subprotocol
Started v0.1.2
Merge pull request #558 from meetecho/datachan-record
Merge pull request #560 from meetecho/chat-plugin
Don't ignore alerts for DataChannel only (or non-muxed) components
Added events to new TextRoom plugin and aligned to master
Added new approach to new TextRoom plugin and aligned to master
Added AG Projects' repo to the resources
Add time to outgoing messages in TextRoom plugin
Remove alphanumeric constraint on username for TextRoom
Use MHD_create_response_from_buffer instead of deprecated MHD_create_response_from_data (issue #565)
Fixed ACL for HTTP transport (issue #564)
Fix detection of lost incoming packets
Add display name to joined event in VideoRoom
Fixed creation of live/ondemand file-based streams
Only validate RTSP parameters if libcurl is available
Added 'autoack' parameter to 'call' in SIP plugin to drive NUTAG_AUTOACK
Added optional admin key to selected plugins to protect 'create' methods
Merge branch 'master' into plugins-json
Merge branch 'master' into event-handlers
Merge pull request #567 from jswirl/master
Merge branch 'master' into plugins-json
Merge branch 'master' into event-handlers
Conditional support of DTLSv1_set_initial_timeout_duration
Fixed check of updated BoringSSL
Don't add ongoing recordings to the list
Fixed typo
Fixed duplicate pcma in VideoRoom
Merge pull request #570 from Yoric/master
Moved NACKs counters/timers to janus_ice_stats (before there was ambiguity on direction), and added new core-level 'slowlink' event
Fix new check and local variable setup
Fix new check and local variable setup
Merge pull request #571 from hasbean/master
Fixes for 64-bit identifiers
Fix for issues #509 and #574
Merge pull request #575 from foxxyz/master
Merge branch 'master' into event-handlers
Merge branch 'master' into plugins-json
Avoid shadowing dup
Merge branch 'master' into fix64
Fixed leaks and typos in Record&Play plugin
Merge pull request #573 from meetecho/fix64
Merge branch 'master' into event-handlers
Merge branch 'master' into plugins-json
Removed references to deprecated lws_get_internal_extensions()
Avoid warning when libcurl is not available
Don't ignore return value of read
Fixed sscanf and format related warning
Don't ignore return value of fread
Removed unneeded extra verbosity for candidated in janus.js
Allow for the configurable recording of the contribution of a single participant (AudioBridge)
Explicitly detach libnice data notifiers when hanging up
Fixed a couple of potential leaks in SIP plugin
Removed extra/unneeded calls to json_decref
Merge branch 'master' into event-handlers
Merge branch 'master' into plugins-json
Fixed typo (wrong check in admin API)
Fixed typo in HTTP transport plugin
New SDP utilities to replace Sofia SIP SDP stack
Made Sofia SIP a dependency for only the SIP plugin, cleaned up configure.ac and Makefile.am, added enumeration for media direction, and used new SDP utils in VideoRoom plugin too
Added 'exists' request to the textroom plugin
Return reason for SDP parsing errors
Merge pull request #584 from joshdickson40/master
Changed some comments to #584, and fixed leak in TextRoom plugin
Merge branch 'master' into event-handlers
Merge branch 'master' into plugins-json
Merge branch 'master' into sdp-home
Merge pull request #577 from tuijldert/master
Fixed typo introduced in #577
Added plugin configuration for whether or not to shoot plugin-specific events (even when global configuration is yes)
Reject attempts to start SIP calls with datachannels (fixes #581)
Added helper method to remove payload types from SDP
Helper method to free an SDP attribute
Support session level connection data
Converted SIP plugin to use the new SDP utils
First take at supporting re-invites/updates in SIP plugin with new SDP utils
Aligned with master (fixed conflicts)
Removed unneeded sdp_parser property
Made Record&Play more tolerant with playout (broken files just skipped)
Revert "First take at supporting re-invites/updates in SIP plugin with new SDP utils"
First take at supporting re-invites/updates in SIP plugin (uses #578)
Merge branch 'master' into event-handlers
Documented use of deviceId (fixes #591)
Changed naming of threads, fixed wav header in audiobridge recording, anticipated sessions stuff in Janus startup (to avoid issues when some of the transport plugins drag and requests start arriving)
Removed extra verbose line
Fixed count of packets in large files when postprocessing
Fix for post-processing (timestamp resets + retransmissions)
Merge pull request #593 from jswirl/master
Parse end-of-candidates
Merge pull request #594 from rpadovani/fixTypo
Merge branch 'master' into plugins-json
Merge branch 'master' into event-handlers
Merge pull request #596 from MotorolaSolutions/master
Merge pull request #592 from meetecho/pp-reorder
Made plugin response more concise (code suggested by @andreasg123)
Fixed VideoRoom publish when datachannels are negotiated but not supported
Added configuration options to transport plugins to control how JSON output is serialized (default=indented, plain, compact)
Set pointers to NULL after a g_list_free
Merge branch 'sdp-home' into sip-updates
Increased size of pollfd array to account for pipe file descriptor
Merge branch 'master' into event-handlers
Don't parse attributes for an m-line not associated to any stream
Merge pull request #544 from meetecho/plugins-json
Fixed indentation
Increase plugin API version
Aligned with new v0.2.0
Merge branch 'master' into event-handlers
Increase plugin API version
Aligned with new v0.2.0
Merge branch 'sdp-home' into sip-updates
Fixed merge introduced error
Added JSON serialization options to TextRoom plugin as well (for data channels)
Changes to DTLS BIO filter for OpenSSL 1.1.0
Fixed memory leak
Merge branch 'sdp-home' into sip-updates
Fixed a couple of leaks/checks
Removed some unneeded extra verbosity
Add size of queued packets queue to Admin API info
Fixed typo
Fixed VideoCall media setup
Initialize BIO filter at startup
Merge pull request #612 from matthewmgamble/master
Fixed warnings
Merge pull request #614 from saghul/sdp-empty-attrs
Merge branch 'master' into sdp-home
Merge branch 'sdp-home' into sip-updates
Merge pull request #606 from meetecho/openssl-1.1.0
Merge pull request #618 from saghul/sdp_fixes
Merge branch 'sdp-home' into sip-updates
Updates to janus.js/adapter.js
Removed unneeded pragma
Removed unneeded checks before g_free
Larger buffer when parsing crypto
Added JANUS_SDP_DEFAULT (=JANUS_SDP_SENDRECV)
Don't write direction attribute if it's JANUS_SDP_DEFAULT
Added fmts list, and fixed datachannels negotiation
Merge branch 'sdp-home' into sip-updates
Fixed typo in configuration example (streaming plugin)
Fixed typos in VideoRoom plugin
Implemented RTP forwarding for AudioBridge's mix
Merge branch 'master' into event-handlers
Conditional check of PIX_FMT_YUV420P availability in pprec (issue #622)
Notify on log when we skip a JS dependency because it's already loaded
Invoke previous onbeforeunload callbacks at page close, if set before ours
Allow port re-use in Streaming mountpoints if it's for multicast (issue #617)
Allow AudioBridge RTP forwarder to relay a mix even when the room is empty
Have VideoCall plugin close PeerConnections on hangup (issue #616)
Show if RTP streaming mountpoint is recording in info request
Fix sequence numbers when media is resumed after a configure/false (issue #620)
Reply with sendonly if AudioBrdge peer is recvonly (fixes #629)
Use urls instead of url in iceServers
Make notification on dropped packets less frequent in AudioBridge (see #626)
Fixed typo and period check in AudioBridge
Better check for SPS in H.264 post-processor, and NAL parse debugging
Parse STAP-A packets when processing H.264 recordings (fixes #630)
Merge branch 'master' into event-handlers
Clarified that the sample plugin needs libvurl in the README
Added display to all participant-related events
Implemented event grouping and HTTP auth+timeout in sample event handler
Return an event to publishers leaving
Fixed typo (see #620)
Made basic authentication the only supported method, for now
Added new category of events (core)
Added Raspberry Pi resources (UV4L) to the docs
Fixed typo
Allow plugins to send out-of-context events (no associated session/handle) to event handlers
Use g_ascii_strtoull instead of atol where applicable
Merge pull request #647 from andreasg123/master-websockets-destroy
Fixed indentation
Merge branch 'master' into event-handlers
Merge branch 'master' into sdp-home
Use g_ascii_strtoull instead of atol where applicable (pt.2)
Merge branch 'sdp-home' into sip-updates
Merge pull request #619 from meetecho/janusjs-adapter
Bump version number
Merge pull request #652 from andreasg123/ice-fix-heap-use-after-free
Unload and skip plugin if init failed (see discussion in #645)
Merge pull request #645 from manifest/feature/janus-mqtt-transport
Fixed indentation of MQTT cfg file
Fixed wrong casts when closing plugins
Fixed typo in documentation
Merge branch 'master' into event-handlers
Merge branch 'master' into sdp-home
Merge branch 'sdp-home' into sip-updates
Merge pull request #653 from andreasg123/websockets-code-duplication
Merge pull request #589 from meetecho/sip-updates
Merge pull request #578 from meetecho/sdp-home
Aligned with new v0.2.1
Fixed small typos in documentation
Fixed small typos in documentation
Fixes to get Janus working with Edge again (see #651)
Don't drop video support on Edge, leave it to the application
Fixed a few leaks
Merge pull request #662 from uxmaster/master
Changed the verbosity of some log messages in WebSockets plugin
Merge pull request #670 from seb3s/seb3s-patch-1
Updated instructions for building libsrtp (1.5.4)
Make sure there's always an event to return in HTTP long poll
Fix crash in SIP plugin when no remote IP is found for RTP in the SDP
Added simple retransmission mechanism to the sample event handler plugin
Merge branch 'master' into event-handlers
Reset retransmission counter after a success
Fixed checks for adapter.js (were broken in Chrome 56)
Fix check for auth token support in admin.html/js
Added setting to modify own volume (percent) in audiobridge (see #668)
Fix SSRCs in RTCP before encrypting and not after, in SIP plugin
Updated resources list
New debugging level in janus.js (vdebug), a few changes in JS logging, and new slowLink event handler (example in echotest.js), plus updates to documentation
Use free instead of g_free for strings allocated by json_dumps (fixes #679)
Merge branch 'master' into event-handlers
Merge pull request #690 from fbertone/patch-1
Merge pull request #678 from eduardomb/master
Removed extra unlock (see #694)
Mention libcurl as optional dependency in the documentation (see #691)
Add optional authentication support to RTSP streaming (see issue #692)
Merge pull request #666 from akfork/master
Edited README guidelines for MacOS (see #666)
Fixed error when compiling Streaming plugin without libcurl
Made configure smarter (see issue #689)
Merge pull request #695 from agclark81/master
Lock forwarder mutex before using forwarder hash table (pull #686)
Merge branch 'master' into event-handlers
Added support for (some) RTP extensions
Added playout-delay to the RTP extensions
Automatically try using SIP INFO for DTMF in SIP demo when not on Chrome
Fixed typo
Implemented timeout/GET_PARAMETER support for RTSP in Streaming plugin
Make negotiation of audio-level RTP ext in AudioBridge configurable
Make negotiation of new RTP extensions in VideoRoom configurable
Merge pull request #697 from meetecho/extmap
Fixed --disable-unix-sockets check in configure.ac (fixes #701)
Merge branch 'master' into event-handlers
Bumbed version number and small fixes to the docs
Autodetect libsrtp version (1.5.x vs 2.0.x)
Updated code to reflect API changes in case libsrtp2 is detected
Shimmed libsrtp2 API
Merge pull request #706 from Sean-Der/videoroom-listparticipants-ssrc
Merge pull request #707 from dazzl-tv/pullreq
Fixed async/sync AJAX request for detach/destroy (fixes #704)
Reduced verbosity of a couple of transport related messages
Modified RTCP code to recognize XR packets
Merge pull request #702 from meetecho/libsrtp2
Added failIfNoAudio/failIfNoVideo capture-related flags to janus.js, both default to false (fixes #705)
Merge pull request #536 from meetecho/event-handlers
Fixed leak when reporting media-type events to handlers
Fixed typo when skipping bytes in post-processing
Added support for libsrtp2 to SIP plugin too (fixes #709)
Removed leftover linking reference in Makefile.am (see #709)
Fixed uncaught typeError for slowLink in janus.ks (fixes issue #710)
Handled case of Aggregate Control containing the URL already (RTSP)
Added check on target extension when post-processing .mjr files
Updated the obsoleted FAQ items in the documentation
Added license exception to explicitly allow linking to OpenSSL (fixes #713)
Use real time instead of monotonic time for events in event handlers
Merge pull request #714 from hijaq/fix/rtpminmax
Fixed truncated error messages in textroom (fixes #720)
Merge pull request #722 from linuxmaniac/vseva/fix_typo
Added manpages for janus and janus-pp-rec (addresses #723)
Merge pull request #726 from meetecho/manpages
Added events_folder property to janus.cfg (fixes #728)
Fixed libcurl-related headers leak (sample event handler, textroom)
Fixed events-related leak when handlers are enabled but none's available (should fix #727)
Fixed outdated line in documentation (fixes #730)
Merge pull request #731 from hfiguiere/patch-1
Merge pull request #734 from hijaq/fix/xhr-status
Fixed duplicate assignment (fixes #735)
Merge pull request #715 from hijaq/fix/videoroomleave
Merge pull request #716 from hijaq/fix/videoroomincrtcp
Fixed exception in videoroom demo JS code
Fixed leftover g_free in a couple of transport plugins (should have been json_decref)
Merge pull request #739 from hijaq/fix/janusjs-nojquery
Added optional identifier to match VideoRoom subscribers to a participant
ACL and kick support in AudioBridge, VideoRoom and TextRoom
Removed unused commented lines in janus.js
Merge pull request #741 from meetecho/publisher-viewer-mapping
Opaque identifier to contextualise handles
Added opaque ID to documentation
Added another paper (Jattack) to publications page
Merge pull request #748 from meetecho/handle-mapping
Deallocate opaque ID when destroying handle
Merge branch 'master' into plugin-tokens
Transport-related events
Unref events queue when shutting down
Added opaqueId to all demos to demonstrate intra-session handles correlation
Merge branch 'master' into transport-events
Removed unused property from AudioBridge
Merge pull request #745 from meetecho/plugin-tokens
Merge branch 'master' into transport-events
Return permament/volatile status as a response to create rooms/mountpoints
Merge pull request #750 from meetecho/transport-events
Removed redundant attribute in Streaming plugin event
Merge pull request #754 from chornyitaras/media_change
Fixed #754, and added error message in case of missing/invalid IP
Fixed crashes in VideoCall when event andlers are enabled (fixes #749)
Increase lifetime of remote candidates before they're enforced (fixes #738)
Allow configuring SSRC when creating RTP forwarders (AudioBridge, VideoRoom)
Fixed typo, and clarified doc for AudioBridge
Make sure private IDs in VideoRoom are unique (fixes #755)
Allow Streaming plugin to relay datachannels, and VideoRoom to forward them
Removed verbose debugging text
Merge pull request #757 from meetecho/rtp-forwarders-pt-ssrc
Merge branch 'master' into streaming-forwarders-datachan
Merge pull request #758 from meetecho/streaming-forwarders-datachan
Updated date in footer
Added accept/reject buttons to VideoCall demo
Added FOSDEM2017 presentation on Event Handlers to video resources in FAQ
Added DevDay Napoli presentation to video resources in FAQ
Add LWS_SERVER_OPTION_DO_SSL_GLOBAL_INIT for secure websockets if supported (fixes #768)
Merge pull request #767 from thehunmonkgroup/fix-textoom-list-log-message
Allow some TextRoom commands to be sent via Janus API
Added withCredentials support to XHR requests in janus[.nojquery].js (fixes #742)
Merge branch 'master' of github.com:meetecho/janus-gateway
Updated janus.js documentation
Reply with created/destroyed when requests come from Janus API (fixes #765)
Changed default MAX nack queue to 300ms instead of 1 second
Configurable timeout for the 'not receiving audio/video' events
Merge branch 'master' into media-timeouts
Updated admin.js and documentation
Configurable session timeout value
Option to add temporary extension while recording
Merge pull request #769 from meetecho/textroom-crud
Merge branch 'master' into media-timeouts
Merge branch 'master' into mjr-tempname
Merge branch 'master' into session-timeout
Don't use mountpoint property of session directly (see #777)
Reference third party js/css files externally (see #778)
Added license header to adapter.js (fixes #781)
Fixed broken link to css
Merge pull request #773 from meetecho/media-timeouts
Merge pull request #774 from meetecho/session-timeout
Merge branch 'master' into mjr-tempname
Fixed relative paths to navbar.html and footer.html in docs placeholder
Externalized adapter.js and removed automatic loading of jquery/adapter from janus.js
Allow VideoRoom publishers to force the plugin to drop their data messages
Merge pull request #775 from meetecho/mjr-tempname
Allow websockets server to bind to IP instead of certificate name (fixes #772)
Print when we're using BoringSSL (and turned some related warnings in infos)
Merge pull request #787 from thehunmonkgroup/libnice-custom-install-instructions
Cleaned up the log notification about the crypto lib in use
Merge branch 'master' of github.com:meetecho/janus-gateway
Merge pull request #789 from thehunmonkgroup/boringssl-configure-enhancements
Changed a few warnings to debug messages in janus[.nojquery].js (fixes #791)
Merge branch 'master' into web-refs
Merge pull request #794 from thehunmonkgroup/resource-janus-event-server
New iceState event in janus[.nojquery].js, and enriched webrtcState event
Merge pull request #776 from cmacq2/multicast-multiple-nics
Small changes to #776, and added related doc info to conf file
Merge pull request #786 from thehunmonkgroup/update-display-value-via-configure
Small fixes to #786, and updated example in AudioBridge docs
Merge branch 'master' into web-refs
Merge pull request #780 from meetecho/web-refs
Made RTP context and rewriting part of the core, rather than plugins
Merge branch 'master' into ssrc-changes
Make sure the PeerConnection is valid before invoking the iceState callback
Merge branch 'master' of github.com:meetecho/janus-gateway
Added 'retransmissions' counter to DTLS contexts, available in Admin API and event handlers
Moved most of SRTP-related stuff to rtp.h/.c (cleans dtls and janus_sip)
Require libsrtp >= 1.5 (1.4 will be rejected)
Reduced verbosity of a couple of debug lines
Merge pull request #796 from meetecho/ssrc-changes
Make sure media is only updated after a re-INVITE
Merge pull request #804 from meetecho/srtp-cleanup
Fixed typos in configure.ac
Merge pull request #802 from chadfurman/patch-1
Documented new media.screenshareFrameRate property (see #802)
Merge pull request #808 from oscarvadillog/safari-mobile
Merge pull request #809 from klachhani/patch-1
try/catch JSON.parse in janus.nojquery.js (see #807)
Merge branch 'master' of github.com:meetecho/janus-gateway
Marc Haase (4):
Added last_received timestamps to rtp streams and provide info in 'list' message
make last_received_* rtp members part of struct janus_streaming_rtp_source
clean up rtp list message response to show age in ms and get rid of 'now'
only output video or audio stats if enabled, initialize last_received_* with current monotonic time
Marcin Sielski (5):
Fix for crashes during shutting down
Fix: mountpoints_mutex should be locked
Sync the port with the demos
Fixes after review
Remove condition check
Mathias Burger (1):
fix janus build on mac os x, add openssl CFLAGS
Mathieu Duponchelle (1):
janus-pp-rec: Fix remuxing of opus streams.
Mathieu ROBIN (2):
Check if adapter is already loaded
Fix the duplicate call to the server
Matthew Gamble (2):
Found a bug in janus_sip.c when the sip stack receives an INVITE without SDP after the inital invite. In this call flow, Janus was assuming the invite would always have an SDP and would segfault when receving an invite without one.
Changing log setting on invite without SIP to LOG_WARN
Maurizio Porrato (1):
Fix log typo
Meetecho (1):
Merge pull request #3 from DamonOehlman/gstreamer-1.0-command
MichaelB76 (6):
Fixed a couple of memory leaks regarding sdp_parser usage. sdp_parser_free() was not being called if the call to sdp_session() failed or if a SIP re-invite was received. In the latter case a significant amount of memory was being leaked when using a SIP client that sends periodic re-invites. The memory was cleaned up when the SIP session was terminated, only really being a problem for connections that stay up for a long time, such as a SIP trunk.
A couple more memory leak fixes concerning SIP re-invites. Also fixed some per-session leaks where memory associated with the janus_sip_session structure was not begin freed.
Merge remote-tracking branch 'upstream/master'
Merge remote-tracking branch 'upstream/master'
A couple more memory leak fixes concerning SIP re-invites. Also fixed some per-session leaks where memory associated with the janus_sip_session structure was not begin freed.
Fixed crash caused by extra g_free on session->stack->session.
Michel Meyer (4):
Properly remove WebSocket event listeners
Fix wsHandlers misspelling
Handle websocket error during session destruction
Clear keepalive timeout at session destruction
Michel Promonet (24):
streaming: allow to receive RTP multicast streams
streaming: allow to receive RTP multicast streams
streaming: allow to receive RTP multicast streams
streaming: allow to receive RTP multicast streams
streaming: allow to receive RTP multicast streams
streaming: allow to receive RTP multicast streams
streaming: allow to receive RTP multicast streams
streaming: allow to receive RTP multicast streams : return an error if IP_ADD_MEMBERSHIP fails
streaming: allow to receive RTP multicast streams : comment multicast sample configuration
streaming : rtsp
rtsp : enable rtsp only if libcurl is available
rtsp: rename method
rtsp : fix memory leak + useless duplicate line
plugins rtsp streaming : manage multicast stream
plugins rtsp streaming : fix multicast checking (wrong byte order using IN_MULTICAST macro)
streaming plugins rtsp : fix double free + add timeout for RTSP requests
streaming plugins : initialize ip_mreq
streaming plugins : rtsp : send TEARDOWN before closing connexion and send multicast transport when SDP signal a multicast stream
rtsp streaming plugins : check RTSP DESCRIBE return code and enable cURL output depending on log level
fix usage of audio_port instead of video_port
Merge remote-tracking branch 'upstream/master'
Merge remote-tracking branch 'upstream/master'
fix compilation due to renaming log_level into janus_log_level
Merge remote-tracking branch 'upstream/master'
Mihail Diordiev (1):
Fix typo in voice mail demo
Min Wang (2):
Add support for sip proxy-auth (407)
disable 100rel as it causes segfault/asserts in sofia-sip
Mrau Hu (2):
Fixed: screen sharing, used code from https://github.com/henrikjoreteg/getscreenmedia
Added Google Chrome extensions-sample for https://*/*
Nicholas Wylie (4):
Fixed Screen Sharing Demo
Moved some includes for easier plugin building
Modify build to output header files
Fixed configure flags when libs available
Philip Withnall (34):
config: Remove unreachable memory error handling paths
debug: Fix string literal formatting in JANUS_LOG
build: Factor common build rules into common.make
build: Use CFLAGS and LDFLAGS
build: Update .gitignore
config: Remove an unnecessary destructor call
config: Use the correct destructor for iterators
config: Use the correct destructor for calloc()-allocated memory
sdp: Clear a global variable on deinit
janus_streaming: Memory leak fixes in janus_streaming
janus_videoroom: Memory leak fixes in janus_videoroom
janus: const-correctness fixes
janus: Mark janus_process_error() as gnu_printf
janus: Don’t pass const strings to variables which are later freed
janus: Don’t call g_type_init() for GLib ≥ 2.36.0
janus: Fix various signed/unsigned integer comparisons
plugins: Fix various no-op if-statements
utils: Simplify janus_string_replace() API
janus_videoroom: Replace string_replace() with janus_string_replace()
plugins: #define packet templates to allow format placeholder checking
build: Enable a whole slew of compiler warnings
core: Fix old-style function definitions
build: Add more compiler warnings
janus_videoroom: Use GAsyncQueue to prevent message race conditions
janus_videoroom: Simplify some g_free() calls
janus_videoroom: Automatically free unhandled messages on shutdown
janus: Simplify HTTP event management a little
janus: Switch janus_session from GQueue to GAsyncQueue
janus: Simplify iteration over the sessions hash table
ice: Add a missing mutex unlock on an error path
janus: Tidy up iteration over ICE handles
ice: Ensure ice_handles is accessed with the lock held
janus: Simplify retrieval of session IDs
build: Add a Valgrind suppressions file
Pierce Lopez (55):
make some janus_recorder_create() args const, remove duplicate condition in plugin loading
logging line spelling error: RammitMQ -> RabbitMQ
more comprehensive and specific gitignore
unref or join some threads
plugins: echotest and streaming: use atomic operations for stopping and initialized flags
fix minor mixup of sws and admin_sws
sctp / dtls threads should deref themselves so they are cleaned up
enable libnice debug messages when debug_level >= 7
echotest page start button needs autocomplete off for button to be re-enabled on reload in firefox
fix accidental copy of videocalltest.html, apply intended changes to original
janus.js: make "lowres" "hires" hints affect capture resolution on firefox 35
fix memory leak when constructing admin response with handle info
fix message memory leak in videoroom plugin
keepalive event payload not expected to be allocated
streaming plugin: fix line accidentally remoted in memory leak cleanup
exit cleanly on SIGTERM
make JANUS_LOG macro less redundant
Trickle error log messages lacked trailing newlines
convert double trinary-operator to single trinary for event->payload
clean up "adding remote candidate" code, mainly logging
fix extra newline when logging ice candidate buffer
fix ice log message spelling "credendials"
webserver request logging quieter
quiet cleaning up session / destroying session log messages
log only when starting to wait for webrtc state to change
combine multiple feature-state logs into one, quiet redundant feature-state logs
quieter logging of final "ice candidate added" message
quiet log "Looping ICE"
log number of recent retransmits once per 5 seconds at INFO level
log retransmitted packets summary at VERB instead of INFO
log "Looping ICE" at DBG instead of HUGE
slow_link callback refactor: count NACKs over full second
only log once when Still cleaning up from previous media session
remove check for g_strdup() failing to allocate memory
count retransmits, instead of received NACKs, for slow_link
avoid starting more requests while janus is stopping
in_stats and out_stats: add total new nacks
re-write NACK generation for missing rtp sequence numbers
refactor logging color output
do not let stun public ip override nat_1_1_mapping ip
janus_process_error(): use buf on stack, avoid leaking allocated error string buf
logging: simplify buffer sizing
configure.ac: ssl_version and glib_version should be shell variables
fix structs janus_request and janus_ice_trickle being typedef'ed twice
config comment stripping was off-by-one, fix and simplify
log msg typo fix "Transpor plugins folder:"
fix leak of component (timeout) source
local_ip private network check: if nat_1_1_mapping set, check it instead
re-do valgrind suppressions file
janus_ice_send_thread(): use g_async_queue_timeout_pop() instead of g_usleep()
janus-pp-rec should always janus_log_destroy() at exit
handle NULL error argument to janus_ice_trickle_parse()
move early janus_mutex_unlock(&rooms_mutex)
two tiny fixes for JANUS_VALIDATE_JSON changes
consolidate JANUS_CHECK_PIN() into JANUS_CHECK_SECRET()
Riccardo Padovani (2):
Use `var` keyword before declaring charSet var.
Fix the same problem in janus.nojquery.js as well
Saúl Ibarra Corretgé (96):
Set SIP From header when sending INVITE requests
Don't unnecessarily duplicate strings when passing values as NUA tags
Remove unneeded call to nua_set_params
Fix using proper To and From headers for 200 OK and BYE
Enable TCP and TLS transports in Sofia-SIP
Add configuration option for SIP keep-alive interval
Disable SIP registration validation
Always use rport when registering over SIP
Simplify setting NUA outbound options
Add option to enable helpers if server is behind NAT
Fix compilation warning
Add option to customize SIP User-Agent string
Move NUA options to nua_create
sip: Remove unneeded check
sip: Save given identity, even in guest mode
sip: Save proxy even when using guest mode
sip: Avoid creating unnecessary NUA handle
sip: Use Sofia-SIP's url module to parse SIP URIs
sip: Simplify code for the 'register' command
sip: Make the SIP proxy optional
sip: Use g_strlcat
sip: Simplify setting Contact header username
sip: Explicitly mention the supported SIP methods
sip: Don't advertise support for Session Timers
sip: Do not create listen sockets
sip: Add register_ttl configuration option
sip: Add ability to specify the local IP address
sip: Bind RTp and RTCP ports to the local IP address
sip: Simplify getting local IP address
sip: Simplify code for connecting RTP and RTCP sockets
ice: Improve gathering of local interfaces
janus: Simplify getting local IP address
sip: Pass destination buffer to IP adutodetect function
sip: Fix fd leak in IP autodetect function
sip: Handle possible getnameinfo errors
sip: Remove unneeded include
ice: Better filter for non-routable IPv6 addresses
sip: Fix checking if we can bind to the local IP address
util: Add function to detect if an IP address is valid
sip: Add ability to listen on IPv6
janus: Add ability to use IPv6 addresses on the SDP
sip: definitively remove TPTAG_SERVER tag
sip: simplify handling of allocation failures
sip: fix potential double-free
sip: fix using the duplicated sdp
demo: Add checkbox for using video in the SIP demo
demo: Simplify checking for checkbox state in SIP demo
janus: reject incoming WS connections if sub-protocol is not set
echo: return error if unrecognizable message is received
sip: add ability to choose the response code for 'decline'
sip: refactor emitting the 'hangup' event
sip-demo: print code and reason for hangup event
sip-demo: allow outgoing calls to be rejected
sip: simplify code for handling SIP authentication
sip: add ability to specify a prehashed secret (ha1)
sip: separate registration and call states
sip: remove redundant check
sip: add ability to skip SIP registration
sip-demo: add ability to use HA1 hashed passwords
sip: fix setting the correct caller for the incomingcall event
sip: send a 'registration_failed' event when SIP registration fails
doc: update usrsctp repository location
build: clean doxygen generated sqlite files
build: don't build static versions of the modules by default
doc: small improvements to the systemd service example
doc: add sysvinit script example
config: fix typo, 'apisecret' -> 'api_secret'
core: rename constant to avoid potential collisions
core: raise default buffer size to 8192
build: fix compilation error
ice: fix enabling ICE Lite mode
sip: fixed reporting re-INVITEs as missed calls
sip: manually handle re-INVITEs and reject them with 488
Add .editorconfig file
sip: fixup style
sip: fix handling subsequent incoming calls
sip: fix SDP parser leak when handling reinvites
Improve sample configuration
sip: add display name to missed_call event
build: clean all generated sample files
core: use RTLD_LOCAL when loading plugins and transports
sip: add ability to customize the display name
sip: add ability to override User Agent per account
sip: style fixes
dtls: simplify key loading code
misc: style fixes (editorconfig)
dtls: refactor loading certificate and key files
dtls: automatically generate a key and cert if they were not specified
doc: remove trailing spaces from README
doc: command line options -c and -k apply to DTLS only
dtls: add warning when autogenerating key/cert
Fix processing SDPs with value-less attributes
Style
Fix compilation
Fix crash if attribute value is empty
sip: reply with 488 if offer doesn't contain audio or video
Scott (1):
Added bower.json file so we can register the front end janus.js library with the bower registry, making it easier for front end developers to pull into their projects
Sean DuBois (1):
Include publisher's internal_audio_ssrc and internal_video_ssrc in plugin_videoroom listparticipants
Simone Gotti (1):
Correctly skip candidates when using bundle.
Sébastien Saint-Sevin (1):
fix documentation
Taras Chornyi (1):
Reconnect sockets to new IP as well
Toby Tremayne (1):
added missing var statements
Victor Seva (1):
fix typo thanks to lintian
Yulius Tjahjadi (1):
Janus build fixes for OSX
amnonbb (6):
Send a FIR to the new RTP forward publisher
Remove the extra space
Send FIR only if forward video
Add new listforwarders request
mutex and name fixes
fix port name
cqm (1):
fix for janus_videoroom_listener leak for janus_videoroom_listener_muxed
foxxyz (1):
RTSP PLAY request URL should not have a slash appended
gatecrasher777 (1):
Update videomcutest.js
hasbean (4):
fix processing vp8 with no extended bit
fix indents
fix indents
fix indents again
janus (3):
Merge remote-tracking branch 'upstream/master'
Merge remote-tracking branch 'upstream/master'
changed name configuration from private to is_private
jing3018 (3):
BUGFIX : opus fill silence packet
fixbug postprocessing for opus using DTX
fixbug postprocessing for opus using DTX
joshdickson40 (8):
add support for 'ack' field in textroom messages
ads 'ack' to message parameters
fix atom tab -> 2 space issue
try char fix
try hard tabs
try hard tabs
reset tabs
make ack comment more clear
jswirl (4):
Include fcntl.h to fix build error on Alpine Linux
Fix VideoRoom SDP compose error
Close socket descriptors on error
Address comments
khejing (1):
fix a problem in wav header
leonuh (2):
Check media resources and handle them (videomcu doesn't work if you have
Fix indents
meetecho (346):
First commit
Renamed README.md
Fixed typo in audiobridgetest.js
REST documentation added
Fixed typo in SIP plugin
Added messages to create rooms in the AudioBridge and VideoMCU plugins
Several changes and improvements
Updated README
New demo (screen sharing) and bugfixes
Version 0.0.2, several fixes and improvements
Added support for rtcp-mux
Fix in potential logging issue
Video MCU segfault fix
Fix to logging problems (undefined symbol) in plugins
Added link to Google Group in the README
Small UI (HTML) cosmetic changes
Exclude list for interfaces, Trickle ICE, fix for Firefox and VideoMCU, etc.
Fixed problem with video MCU that caused screen sharing not to work anymore
Removed unneeded MHD_USE_DEBUG for HTTPS
Added BUNDLE support and fixed Trickle ICE
Updated JavaScript documentation
Several changes in the SIP plugin
Fixed getUserMedia when answering in SIP
Bugfixing in SIP plugin
Fixed race condition between setRemoteDescription and createAnswer
Clarified that libopus may or may not be available in Ubuntu/Debian repositories
Added a make cmdline before the actual make, as otherwise cmdline is not built the first time
Fix to issue #20
Added a FAQ to the documentation
Added support for Data Channels
Removed unneeded echo from README
Removed sctptest binaries (added by mistake)
Changed license from AGPLv3 to GPLv3
Made Data Channels support optional when installing
Added help flag to the install script to show usage
Fixed problem when using the web server root as base path
Updated FAQ to address optional data channels and potential usrsctp compilation errors
Attempt to fix occasional race condition when bundle is involved
Several changes to the core
Better error management in plugins and other changes
Some more fixes for the BUNDLE Case
Fixed segfault when audio is not negotiated
Fixed typo in handling bundled streams
Fixed typo in the VoiceMail plugin
Fixed an issue where, for non-bundled streams including data channels, setup_media would not be called in plugins
Experimental WebSockets support and several other changes
SSRC fixing of RTCP in SIP plugin
Added a basic recording functionality plugins can use
Fix in SDP generated m-lines
Fixed link to libwebsock 1.0.4
Fixed link to libwebsock 1.0.4
Added possibility to specify desired room ID when creating rooms in AudioBridge and VideoMCU plugins
Fix in SIP plugin (issue #35)
Improved hangup of PCs from plugins
Fallback addresses in janus.js
Timeout watchdog for sessions
Timeout watchdog for sessions
Fixed segfault on /info endpoint
Added create/destroy commands to the streaming plugin to dynamically manage streame
First steps in adding support for SSRC multiplexing (Plan B) to the VideoMCU plugin
Several bugfixes
Bug fixing
Fixed debugging typo
Ignore RTCP trickle candidates if rtcp-mux is used
Allow passing a desired ID for a new publisher in the video MCU (issue #56)
Added first version of admin/monitor/overview API (issue #41, disabled by default)
Disable admin/monitor by default
Admin/monitor documentation
A bit of fixes and improvements in the streaming plugin
Simple admin/monitor demo page
Fixed issue with video not working on latest Firefox 34 Nightly
Restored publishers event after merge #62
Some post merge #62 fixes
Fixed occational segfault when participants left the video MCU
Aligned some glib usage to the recent cleanup
Added support for escaped semicolons in configuration files
Added option to provide fmtp codec parameters to RTP-based streams (streaming plugin)
Added way to group trickle candidates in a single request
Added a new joinandconfigure request to the Video MCU to automatically publish when joining as a publisher (needs JSEP offer to be attached to the request)
Updated previous merge to use the new Janus Chrome extension
New synchronous API for plugin messaging and preliminary NACK support
Admin API to change debugging and fixed deadlock on session timeout
Made some requests in the streaming and videoroom plugin synchronous
Fixed leftover in sctp.c (issue #67)
Fixed some missing steps in the new configure/compile/install process (see #68 for details)
Updated configuration file for voicemail plugin
Added experimental support to RabbitMQ as a transport for the Janus API
Fixed linking issue for optional plugins (voicemail, audiobridge), issue #70
Several changes and fixes
Attempt to fix issue #72
Fixed apparent issue with OfferToReceiveAudio/Video when set as false (e.g., MCU for sendonly)
Fixed issue when getting info through websockets/rabbitmq
Couple fixes on ICE, streaming, and admin UI
Fixed typo in README
Fixed issue with WebSockets and missing events (issue #73)
Ad-hoc thread for outgoing media/data
Several fixes
Fixed bad quality (low bitrate) video in MCU on recent Chrome versions
Fixed reception of SCTP label (issue #80)
Fixed reception of SCTP label (issue #80)
Experimental IPv6 support and new Recorder/Playout plugin
Updated list of demos
Fixed segfault for listeners with no publishers (issue #81)
Fixed DTLS/SCTP typo in SDP
Added experimental videoswitching to MCU viewers
Fixed typo (issue #84)
Janus ping/pong message and updated documentation
Merge branch 'master' of github.com:meetecho/janus-gateway
Made max NACK value configurable (command line, configuration file, admin API)
Fixed overflow for RTCP in video mcu (issue #93)
More debug on retransmitted packet (issue #89)
Fixed a typo that excluded last NACK, or only NACK in a list of 1 (issue #89)
Fixed bitrate settings not working in MCU (issue #88)
Fixed dead link to v1.0.4 of libwebsock in README (use git tags)
Fixed dead link to v1.0.4 of libwebsock in README (use git tags)
Fixed dead link to v1.0.4 of libwebsock in README (use git tags)
Further fix on bitrate adaptation in MCU (issue #88)
README clarifications
Changed the way trickle support is detected (issue #83); improved ordering of SDP fields
Always assume trickle is supported (issue #83)
Merge branch 'master' of github.com:meetecho/janus-gateway
New timer for NACKs to avoid retransmitting the same packet over and over
Clarified role of libevent in libwebsock (both optional)
More room for fmtp in streaming plugin
Fixed typo in echo test
Added option to make streaming mountpoints private (won't appear in a list request)
Several changes to the audiobridge plugin
Added support for 'private' rooms in audiobridge and videoroom
Changed json_boolean to json_string for older jansson versions
Merge branch 'thread_stack_leaks' of https://github.com/ploxiln/janus-gateway into ploxiln-thread_stack_leaks
Further changes to the other threads (plugin and core)
Merge branch 'ploxiln-thread_stack_leaks'
Some improvements on the DTLS handshake (and related debugging)
Avoid working on queue if it has been unrefed (issue #96)
Fixed SIP demo page (local stream was not muted on Firefox)
Fix for the recent ICE issues with Firefox stable
Potentially missing mutex unlock when parsing candidate
Configuration and API to enable/disable libnice debugging
Fixed demo pages as per #119
Further fixes on inputs in demo pages as per #119
Fixed typo in agent creation and added some more debugging
New command line flag to enable libnice debugging
Ignore TCP local candidates if libnice is 0.1.8 (TCP still WIP)
Attempt to fix issue #126
Fixed issue #124 (label size for data channels)
Unhide UI box in echotest and videocall if data channels are open
Fixed bug in audiobridge plugin when leaving a room
Removed overly verbose text
Several changes and fixes, mostly to address the new feature added in #114
Some more UI fixes to tackle #114
Fixed console log error in janus.js (issue #128)
Fixed DTLS handshake issue with Firefox Nightly
First draft for some data transfer statistics in the admin API
Some fixes in the AudioBridge join/changeroom behaviour
Plugin API change: compatibility check and admin-related session handle query
Fixed typo
Better DTLS-related debug (handle info)
DTLS fix for issue #132 (and #134 as well?)
Fixed rejoin issue in audiobridge (no audio)
Better management of NACKs and additional statistics in the admin API
Fixed leave/join audio issues in the audiobridge
RTP range (ICE) fix and some debug levels changes
Moved recorder cleanup to hangup_media in videoroom plugin (issue #138)
Fix to autogen.sh after latest pull request #125
Fixed cdone not being reset
Improvements in AudioBridge and VideoRoom plugin
Clarified in docs examples that session_id and handle_id are numeric
Better management of disabled stream; re-added transaction to info reply
Attempt to fix issue #135 (and potentially other similar cases)
First attempt at adding support for ICE-TCP (if libnice >= 0.1.8)
Added switching a-la MCU to the streaming plugin as well (live RTP only)
API notifications ('media') when audio/video is first received/resumed or stopped
Added reason to hangup event (and improved it)
Fixed new 'media' event (missing IDs) and new documentation for it
Added count of sent/received NACKs to the handle info in the admin API
Added number of viewers of a videoroom publisher in admin API
Added support for TURN gathering in Janus and selective enable/disable of ICE-TCP; info added to admin API as well
Fixed a couple of configure checks
Better management of NACKS as per issue #150
Fixed typo (missed in previous commit)
Better management of close_pc; transaction of call in SIP related events
Fixed typo added in #152
Added 16:9 options to the JavaScript library video settings
Fixed typo in configure that mixed data channels and rabbitmq support
Added some doxygen documentation for the plugins APIs as well
Fixed broken NACK behaviour, made shutdown faster and added summary to configure
Fixed typo (too verbose)
Better indentation of #155 and moved check a little earlier
Converted the SIP plugin to use poll instead of select for media relaying
Added AudioBridge API documentation to doxygen
Some cleanups and fixes, especially on session destruction
Fixed deadlock on session timeout after latest cleanup
Added a local mute button to the videoroom demo
Version 0.0.8 of Janus
Fixed very delayed audio in AudioBridge after a destroy and a different join
Converted the streaming plugin to use poll instead of select, and negotiating NACK for RTP streaming too now
Fixed typo in admin API
Limited size of queue for incoming packets (AudioBridge)
Per-participant encoding thread in AudioBridge for better performances
Added option to enable ICE Lite, only way to get ICE-TCP working if it's needed
Command line option description updated
Added way to selectively disable plugins in configuration file (#160)
Added constant time strcmp to the utils (#161)
strlen fix to the constant time strcmp
Added page with paper bibtek
Yet another fix related to issue #161
Added some missing unlocks instream destroy
Fixed usage of new constant time strcmp, which unlike strcmp returns TRUE when strings are equal
Removed extra variable definition (leftover from #168)
Indentation fix for the SIP plugin code
Attempt to fix issue with data channels labels (#165)
Switched from onloadedmetadata to onplaying as per discussion in #172
Attempt to fix data channel issue identified in #165
Made hashtable iteration safer when the hashtable is NULL or empty
Modified slow_link callback to account for uplink and downlink issues, as discussed in #174 and #175
Fixed typo (inverted uplink/downlink behaviour)
Fixed demo pages, page head was not being closed
Some cosmetic changes after merging #174
Changed uplink in slow_link recordplay event to an integer (issue #174)
Added way to use standard lookup for proxies in the SIP demo page, but hidden behind a dialog to avoid confusion (see #178)
Updated memory leaks in all plugins as per #179
Hide warning for rabbitmq-c usage in janus.c
Fixed a couple of leaks
Fixed a couple of leaks
Some more memory leaks in plugins
Some more memory leaks fixes in plugins
Fixes to some memory leaks in the Janus core
Fixed abort after a plugin forces the end of a session (issue #185)
Feedback about slow links to echotest and videocall users
Fixed typo in the streaming plugin
Modified SDP merge in core to use IP6 instead of IP4 in c-lines, when needed
Fixed documentation on the maxed parameter in long polls (see #188)
Removed unneeded unlock/lock when relaying DataChannel data (ref. issue #189)
Fixed potential looping issues at startup in the streaming plugin
Added info on current bitrate to slowlink events in videoroom and record&play
Improved and documented optional debugging of SCTP messaging
Fixed typo (wrong event name)
Changed WebSockets library from libwebsock to libwebsockets
Fix in janus.js for Chrome 43 (broken JSON.stringify for WebRTC objects)
Fix on mid management (Firefox Nightly)
Fixed leftover in SIP plugin (issue #196)
Fixed case of empty s= attribute in SDP (issue #194)
Fixed typo in SDP mid management
Fixed segfault when Record&Play has only audio or video recorded (issue #195)
Attempt to fix websockets-related occasional segfault (issue #193)
Added optional timestamps to logging (issue #191)
Added periodic REMB to videoroom publishers
Fixed wrong settings management introduced with new logging timestamps feature (thanks @ploxiln for spotting that)
Added contributing guidelines
Updated Janus-related publications
Fixed documentation typo
Added support for the TURN REST API (draft-uberti-behave-turn-rest-00) to dynamically get TURN servers and credentials to use within Janus
Configuration template for TURN related settings in Janus
Better management of closed WebSocket sessions (issue #201)
Fixed libcurl/TURN REST API autodetection and disable trigger (issue #207)
Attempt to fix issue #144 (timing related SCTP stack problem)
Fixed incorrect behaviour where WebSockets could not notify connection-related events right away
Provided guidelines for opening issues
Don't try receiving SCTP data unless we sent some (wait for our connect, issue #144)
Fixed deadlock for timed out sessions (issues #210, #211)
Fixed typo
Removed unneeded ready flag in the SCTP stack management (issue #144)
Fixed link to CLA in contributing guidelines
Fixed invalid addresses in Via and Contact headers in SIP plugin (issue #213)
Handle recent change in libwebsockets build that adds a _shared to the so builds
Better management of watchers in case a mountpoint is destroyed (issue #215)
Updated bibtek for Janus performances paper
Updated bibtek for IPTComm 2014 paper on Janus (in proceedings now)
Reduced debug level of REMB transmission in videoroom (VERB, was INFO)
A few changes to pull #217:
Just a couple of cosmetic changes to pull #230 (capitalize first letter of comments)
Fixed indentation (#222)
Added further check to verify validity of SRTP stack
Fixed missing bracket in conditional code in sdp.c
Disabled MHD_quiesce_daemon as per discussion in #235
Cosmetic changes to #238 (comments) and renamed seq_in_range to janus_seq_in_range
Added way for videoroom plugin to just relay FIR/PLI coming from viewers to publishers, for faster video recovery
Fixed missing callback on handle send in janus.js (see #244)
Fixed typo in docs (candidate->candidates
Fixed typo that caused the wrong pointer to be checked (WS/RMQ), see issue #245
Created 1024 bits certificate (see #251), and added small documentation file
Implemented new OpenSSL BIO filter to fix fragmentation issue in DTLS on large certificates (see #252)
Made starting MTU value for the BIO filter configurable
Integrated new OpenSSL BIO filter for DTLS fragmentation
Restored markdown file describing the certificates folder
Added checks to avoid negative integers in API requests (issue #241)
Fixed detection of incoming RTCP packets (audio vs video) when remote SSRC is unknown (issue #258)
Fixed occasional issue when processing video recordings
Remove session from the RabbitMQ manager if it timed out or was destroyed
Added resetdecoder request (synchronous) and queues length in audit to the audiobridge plugin (issue #242)
Added a getVolume() method to janus.js to get the current peer volume, and made both getBitrate() and getVolume() a by request property (don't start timers if they weren't asked for)
Made janus.js getBitrate() work with Firefox too (note: does it break Firefox pre-38?)
Added JANUS_PPREC_DEBUG environment variable to increase debug in post processor
Some more debugging in post processor
Minor nits
Selective listeners of media in videoroom, and related fix in core
Fixed media constraints for Firefox
A few changes and typo fixes; improvements in janus.js
Better checking of invalid configuration object in janus.js
Better handling of invalid handle object in janus.js
Fixed occasional problems with double detaches (as evidenced in #260)
Dropdown menu for registration approach in SIP demo
Simple helper request to verify if Janus can write on the RabbitMQ
Fixed typos in documentation
Fixed a potential problem with incoming RTP streams, and removed a useless parameter in janus_process_success that did nothing (probably a leftover)
First version of Janus with modular/pluggable transports
Removed binaries
Added subscriber configure, to dynamically choose what to receive (issue #277)
Implemented 'transport gone' core callback
Fixed typo in HTTP transport module, and updated documentation
Added Admin API support to WebSockets transport (janus-admin-protocol)
Added Admin API support to RabbitMQ transport (separate queues)
Debugging visibility nits
Changed external int for debugging to avoid clashes with libwebsockets
Additional checks to avoid using old plugin sessions
Changed names of external logging variables to avoid conflicts with libwebsockets
Merged with latest master commits
ICE Lite fix (conflicting roles)
Merge branch 'master' into modular-transports
Fixed a couple of typos in the configuration files, and renamed secure WS stuff to wss in there
Fixed ACL mechanism for HTTP, and implemented ACL mechanism for WebSockets
Merge branch 'master' into modular-transports
Fixed autodetection of libwebsockets shared library name
Fixed ICE not starting when all trickles received before processing remote answer
Merge branch 'master' into modular-transports
Fixed update request in RecordPlay plugin so that deleted recordings are removed from the list (see issue #278)
Merge branch 'master' into modular-transports
Fixed regression in Record&Play demo (issue #278)
Clarified documentation on local, file-based, deployment (issue #291)
Suggest version 1.5 of libsrtp in documentation
Merge branch 'master' into modular-transports
Added AC_CONFIG_AUX_DIR macro to configure.ac (issue #290)
Merge branch 'master' into modular-transports
Reduced verbosity of WebSockets transport plugin ACL
Separate threads for individual WebSockets services
Addressed comments from @ploxiln on #281
Fixed typo in sample configuration file, and updated favico
Print timestamp of first detected keyframe when postprocessing videos
Added alternative git repo for libwebsockets, in case the first one is unreachable
Fixed deadlock in videocall plugin
Merge branch 'master' into modular-transports
Made hangingup checks in plugins atomic (see issue #297)
Merge branch 'master' into modular-transports
Added options to force BUNDLE and/or rtcp-mux (forcing both will always only allocate a single port for media, instead of 2/4)
Merge branch 'master' into modular-transports
Better management of hangingup flag in plugins (issue #297)
mporrato (1):
Fix config files path for staged installations
mpromonet (13):
rtsp : fix rtp port + keep open RTSP connection
Merge remote-tracking branch 'upstream/master'
rtsp: use dynamic port
rtsp : fix crash when media is not supported
remove modification of log
rtsp: manage create message
rtsp update comment
rtsp: fix build without libcurl
rtsp: fix build without libcurl
rtsp: fix build without libcurl
rtsp : merge rtp & rtsp structure to reduce copy of code
rtsp : merge rtp & rtsp structure to reduce copy of code
Merge remote-tracking branch 'upstream/master'
mtdxc (1):
Update janus_videoroom.c
oscarvadillog (3):
Fixed attach and reattach media over iOS devices
Updated browser detection condition. Now, we use adapter.browserDetails
Updated browser detection condition. Now, we use UserAgent if user navigated from mobile Safari
pallab-gain (7):
Configurable audio codecs supports in VideoRoom plugin. We should now be able to decide which audio codec ( OPUS, ISAC 32K, ISAC 16K, PCMU ) to use as publisher during creating a room.
Fixing errors, and suggested improvements by lminiero
PCMA_PT was missing
Fixing indentation bug, and adding missing code convention practise.
Merge remote-tracking branch 'upstream/master'
Fixed indentation bug, and added missing code convention practise, and PCMA audio codec
Fixed indentation bug, and added missing code convention practise
richstorm (3):
Proceeding call state added
Early media for session progress
rolled back changes for early media
tuijldert (8):
Allow for a separate authentication username.
Bug-fix: use the correct 'authuser' fields and some indenting cleanup.
Enhancement: also report display-name of caller when present.
Extra check on "from" field.
Rudimentary handling of SIP session-refresh (keepalive) added.
Merge remote-tracking branch 'upstream/master'
Session-refresh handling (2) - free memory
Merge branch 'master' of https://github.com/tuijldert/janus-gateway
uxmaster (3):
Fixed GLib-CRITICAL after session timeout
typo fix
typo fix
xenyou (1):
update README.md
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