Changelog for janus-gateway-0.0.9+git20151026.a61dd85-5.1.x86_64.rpm :
Wed Oct 28 13:00:00 2015 opensuse-packagingAATTopensuse.org
- Update to version 0.0.9+git20151026.a61dd85:
+ Check if adapter is already loaded
+ Fix the duplicate call to the server
+ 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.
+ 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
+ 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.
+ 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
+ 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
+ Fixed crash caused by extra g_free on session->stack->session.
+ fix read cert_pem for REST https
+ Add optional BoringSSL support via configure
+ Fixed occasional inability to remove RTP forwarders in videoroom plugins
+ Require a valid certificate key when staring Janus
+ 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
+ Fixed echo test data channels forwarding (last character cut away)
+ Converted memory allocations to GLib ones, and fixed a couple of leaks
+ 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
+ If both API secret and token auth mechanism are enabled at the same time, either one that is provided and valid is fine
+ sip: fixed reporting re-INVITEs as missed calls
+ Added method to save a configuration object to file
+ sip: manually handle re-INVITEs and reject them with 488
+ Add .editorconfig file
+ sip: fixup style
+ 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
+ sip: fix handling subsequent incoming calls
+ sip: fix SDP parser leak when handling reinvites
+ off by one buffer overflow
+ protect access to freed janus_websockets_client with old_wss_mutex
+ Further check before pushing plugin session event
+ Fixed typo in writing recording header
+ Added possibility to limit scope of auth tokens to specific plugins
+ Fixed token/plugin check when API secret is involved
+ fixup patch according to janus coding style
+ 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
+ Added admin API methods to dynamically toggle log colors and timestamps
+ 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
+ New UI and features for the admin API web demo
+ Add transports to the new admin API web demo
+ Show docs creation/update time in html pages
+ Allow enter to be used in admin web UI for new tokens
+ Allow for a separate authentication username.
+ Decreased verbosity for some lines (info to verb), and added call to nice_agent_remove_stream when enforcing bundle/rtcp-mux (see #154)
+ 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.
+ 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
+ Send a FIR to the new RTP forward publisher
+ Remove the extra space
+ Send FIR only if forward video
+ 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
+ Prettier admin UI for handle info
+ List discovered (prflx) remote candidates when querying the admin API
+ Use MediaStreamTrack.stop() (see #363)
+ Updated references to videoroom in the demos, and clarified it\'s an SFU and not MCU
+ Added autorefresh checkbox for handle info in admin API web demo
+ Fixed parsing of fingerprints so that they can be different per each stream
+ Fixed missing stream/component IDs in janus_ice_component

Tue Sep 22 14:00:00 2015 opensuse-packagingAATTopensuse.org
- Switched to the modular-transports git branch
(version 0.0.9+git20150921.e617df8)

Wed Sep 9 14:00:00 2015 opensuse-packagingAATTopensuse.org
- Update to version 0.0.9+git20150909.f1b79cd:
+ 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
+ fix compilation due to renaming log_level into janus_log_level
+ First attempt at getting rid of the increasing delay in audiobridge rooms when network is shaky for a few users
+ config: fix typo, \'apisecret\' -> \'api_secret\'
+ Attempt to fix occasional issue with websockets and session timeouts (see issue #307)
+ 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) Added recording capability to EchoTest, VideoCall and SIP plugins Fixed a few nits here and there
+ 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

Fri Aug 28 14:00:00 2015 opensuse-packagingAATTopensuse.org
- Update to version 0.0.9+git20150828.1894378:
+ Fixed issue when destroying streaming mountpoints Added missing su_home_unref to SIP plugin
+ doc: small improvements to the systemd service example
+ doc: add sysvinit script example
+ Changed default value of hangingup when creating plugin sessions to false
+ Set the limit of open files in systemd unit example
+ Fixed a couple of data channels potential leaks, and addressed potential overflow when forwarding data channel messages in plugins (see issue #302)

Wed Aug 26 14:00:00 2015 opensuse-packagingAATTopensuse.org
- Update to version 0.0.9+git20150826.3c5413c:
+ Prevent bower to use a too recent adapter.js
+ Additional checks to avoid using old plugin sessions
+ Changed names of external logging variables to avoid conflicts with libwebsockets
+ ICE Lite fix (conflicting roles)
+ sip-demo: add ability to use HA1 hashed passwords
+ 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)
+ sip: fix setting the correct caller for the incomingcall event
+ Fixed typo in sample configuration file, and updated favico
+ sip: send a \'registration_failed\' event when SIP registration fails
+ 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 Fixed hangup_media not being invoked in some plugins when preceded by destroy_session (see issue #297 and #298)
+ 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)
+ doc: update usrsctp repository location
+ build: clean doxygen generated sqlite files
+ build: don\'t build static versions of the modules by default
+ 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

Tue Jul 14 14:00:00 2015 opensuse-packagingAATTopensuse.org
- Update to version 0.0.9+git20150713.bb495e7:
+ echo: return error if unrecognizable message is received
+ 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
+ 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
+ Minor nits
+ Selective listeners of media in videoroom, and related fix in core
+ Fixed media constraints for Firefox
+ 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
+ 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)

Wed Jul 1 14:00:00 2015 cwhAATTsuse.com
- Enable janus_postprocessing for oS newer than 13.2

Thu Jun 18 14:00:00 2015 opensuse-packagingAATTopensuse.org
- Update to version 0.0.9+git20150617.31ac9d1:
+ 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
+ demo: Add checkbox for using video in the SIP demo
+ demo: Simplify checking for checkbox state in SIP demo
+ 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
+ Restored markdown file describing the certificates folder
+ Added checks to avoid negative integers in API requests (issue #241)
+ 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
+ Fixed detection of incoming RTCP packets (audio vs video) when remote SSRC is unknown (issue #258)
+ Fixed occasional issue when processing video recordings
+ fix a problem in wav header
+ janus: reject incoming WS connections if sub-protocol is not set
+ Remove session from the RabbitMQ manager if it timed out or was destroyed

Thu May 21 14:00:00 2015 ancorAATTsuse.com
- added websockets support

Wed May 20 14:00:00 2015 mrueckertAATTsuse.de
- prepared post processing support atm guarded with
janus_postprocessing:
new BR: libavformat libavcodec libavutil via pkgconfig
- added BR for a few more features:
ogg and opus via pkgconfig and curl-devel

Wed May 20 14:00:00 2015 mrueckertAATTsuse.de
- first ronud of clean ups
- guard all systemd related parts in %if %{with systemd}
- create rc symlink in the systemd case
- create user and group and the home directory for the user
- split out header files into a devel package
- disable silent rules to fix gcc post build check
- move the config handling from %post to %install where it
belongs
- mark config files with config(noreplace) and set permissions to
u=rwX,g=rX,o= with root:janus
- delete the sample key and cert
- make unit file start it as janus:janus
- remove the logger call from the unit file. systemd does that
for us already

Wed May 20 14:00:00 2015 opensuse-packagingAATTopensuse.org
- Update to version 0.0.9+git20150520.3f16cb8:
+ Added way for videoroom plugin to just relay FIR/PLI coming from viewers to publishers, for faster video recovery

Tue May 19 14:00:00 2015 opensuse-packagingAATTopensuse.org
- Run the service as root (until we found the culprit of the errors)
- Update to version 0.0.9+git20150519.0ac1398:
+ streaming : rtsp
+ rtsp : enable rtsp only if libcurl is available
+ rtsp : fix rtp port + keep open RTSP connection
+ 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
+ rtsp: rename method
+ rtsp : fix memory leak + useless duplicate line
+ 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
+ Updated bibtek for Janus performances paper
+ 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
+ Updated bibtek for IPTComm 2014 paper on Janus (in proceedings now)
+ Reduced debug level of REMB transmission in videoroom (VERB, was INFO)
+ only log once when Still cleaning up from previous media session
+ remove check for g_strdup() failing to allocate memory
+ A few changes to pull #217: Added related info to the sample config file; Fixed an htons that was using a pointer instead of the short int; A few cosmetic changes to align to the code style
+ Just a couple of cosmetic changes to pull #230 (capitalize first letter of comments)
+ Fixed indentation (#222)
+ count retransmits, instead of received NACKs, for slow_link
+ avoid starting more requests while janus is stopping
+ Added further check to verify validity of SRTP stack
+ Fixed missing bracket in conditional code in sdp.c
+ in_stats and out_stats: add total new nacks
+ Disabled MHD_quiesce_daemon as per discussion in #235
+ re-write NACK generation for missing rtp sequence numbers
+ Cosmetic changes to #238 (comments) and renamed seq_in_range to janus_seq_in_range

Wed May 6 14:00:00 2015 ancorAATTsuse.com
- added systemd service file
- added janus user
- Update to version 0.0.9+git20150501.35a37ba:
+ make JANUS_LOG macro less redundant
+ 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)
+ sip: definitively remove TPTAG_SERVER tag
+ sip: simplify handling of allocation failures
+ sip: fix potential double-free
+ sip: fix using the duplicated sdp

Tue Apr 21 14:00:00 2015 lnusselAATTsuse.de
- switch to disabled service file for factory package
- change versioning format to 0.0.9+git...

Mon Apr 13 14:00:00 2015 ancorAATTsuse.com
- update to 0.0.8 version
- enable data channels

Mon Oct 27 13:00:00 2014 mlinAATTsuse.com
- switch back to upstream master

Thu Oct 23 14:00:00 2014 mlinAATTsuse.com
- switch the source to ancorgs\'s break_mcu tree

Thu Oct 23 14:00:00 2014 mlinAATTsuse.com
- add usrsctp for data channel support

Wed Oct 22 14:00:00 2014 mlinAATTsuse.com
- let the samples being default configuration if no config file found

Wed Oct 22 14:00:00 2014 maxlinAATTlocalhost
- initial import from git