[Pkg-voip-commits] [janus] annotated tag v0.0.9 created (now 93a713e)
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 v0.0.9
in repository janus.
at 93a713e (tag)
tagging 638695a118de5ba7931cae5f0c73bb271455235e (commit)
tagged by Lorenzo Miniero
on Wed Nov 11 11:06:06 2015 +0100
- Log -----------------------------------------------------------------
version 0.0.9 (pre modular-transports)
Ancor Gonzalez Sosa (3):
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
Benjamin Trent (1):
session pointer not set to NULL after free in videoroom session free function, corrected it
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
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.
Evan Coury (1):
Fix Fedora package name for pkgconfig
Giacomo Vacca (2):
autogen.sh requires autoconf package
Update README.md
Graeme (1):
Update Ubuntu/Debian .deb install
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
Lets_Vape (1):
Fix videoremote id for spinner
Lorenzo Miniero (242):
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)
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
Fixed issue when destroying streaming mountpoints
Merge pull request #313 from saghul/init-docs
Changed default value of hangingup when creating plugin sessions to false
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)
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
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)
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)
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
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
Fixed missing mutex unlock
Fixed link to libsrtp in both README and docs
Fixed recording of SIP calls when filename is provided
Merge pull request #339 from saghul/fix-ice-lite
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)
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
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
Free addrinfo after it's been used
Merge branch 'master' into boringssl-support
Fixed echo test data channels forwarding (last character cut away)
Converted memory allocations to GLib ones, and fixed a couple of leaks
Added the possibility to specify an optional PIN to access streaming mountpoints and audiobridge/videoroom conference rooms
If both API secret and token auth mechanism are enabled at the same time, either one that is provided and valid is fine
Merge pull request #338 from MichaelB76/master
Merge pull request #344 from saghul/sip-reinvite-missed-call
Added method to save a configuration object to file
Merge pull request #345 from saghul/sip-manual-reinvite
Merge pull request #346 from saghul/editorconfig
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
Add info on when the handle was created to the admin API
Merge pull request #347 from saghul/fix-sip-handle-reset
Merge pull request #348 from saghul/sip-sdp-leak
Further check before pushing plugin session event
Fixed typo in writing recording header
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 pull request #343 from meetecho/boringssl-support
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
Added admin API methods to dynamically toggle log colors and timestamps
Return whether API secret and token mechanism are enabled in the server info
New UI and features for the admin API web demo
Show docs creation/update time in html pages
Allow enter to be used in admin web UI for new tokens
Decreased verbosity for some lines (info to verb), and added call to nice_agent_remove_stream when enforcing bundle/rtcp-mux (see #154)
Merge pull request #354 from tuijldert/master
Merge pull request #356 from tuijldert/master
Allow applications to provide their own MediaStream to janus.js
Updated documentation
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
Make sure rec_dir is honored even when providing a filename in a videoroom configure request (issue #357)
Fixed missing CR in SDP generation
Added new console wrappers to janus.js, and bound them to debug level in init (see #292)
Fill gaps in audio recordings with silence, when postprocessing
Fixed detection of Opus and VP8 payload types in some cases
Fixed detection of Opus and VP8 payload types in some cases
Removed unneeded extra debugging
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 janus-edge
Prettier admin UI for handle info
Merge branch 'master' into janus-edge
List discovered (prflx) remote candidates when querying the admin API
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 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 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 spinning-threads
Fixed missing stream/component IDs in janus_ice_component
Merge branch 'master' into spinning-threads
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 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
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
Added fix from #366 and #367 to other plugins as well
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
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 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'
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 (40):
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
Saúl Ibarra Corretgé (77):
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
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
Yulius Tjahjadi (1):
Janus build fixes for OSX
amnonbb (3):
Send a FIR to the new RTP forward publisher
Remove the extra space
Send FIR only if forward video
gatecrasher777 (1):
Update videomcutest.js
janus (3):
Merge remote-tracking branch 'upstream/master'
Merge remote-tracking branch 'upstream/master'
changed name configuration from private to is_private
jing3018 (1):
BUGFIX : opus fill silence packet
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 (322):
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)
Added subscriber configure, to dynamically choose what to receive (issue #277)
Additional checks to avoid using old plugin sessions
Changed names of external logging variables to avoid conflicts with libwebsockets
ICE Lite fix (conflicting roles)
Fixed ICE not starting when all trickles received before processing remote answer
Fixed update request in RecordPlay plugin so that deleted recordings are removed from the list (see issue #278)
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
Added AC_CONFIG_AUX_DIR macro to configure.ac (issue #290)
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
Made hangingup checks in plugins atomic (see issue #297)
Added options to force BUNDLE and/or rtcp-mux (forcing both will always only allocate a single port for media, instead of 2/4)
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'
tuijldert (4):
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.
uxmaster (1):
Fixed GLib-CRITICAL after session timeout
xenyou (1):
update README.md
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