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| Name: janus-gateway-devel | Distribution: openSUSE:Factory:zSystems |
| Version: 1.1.4 | Vendor: openSUSE |
| Release: 1.1 | Build date: Fri Oct 27 21:31:23 2023 |
| Group: Development/Libraries/C and C++ | Build host: s390zl23 |
| Size: 226904 | Source RPM: janus-gateway-1.1.4-1.1.src.rpm |
| Packager: https://bugs.opensuse.org | |
| Url: https://github.com/meetecho/janus-gateway | |
| Summary: Development files for Janus Gateway plugins | |
Janus is a general-purpose WebRTC gateway designed and developed by Meetecho. This package holds the file needed to compile plugins for Janus Gateway.
GPL-3.0-or-later
* Tue Oct 10 2023 chris@computersalat.de
- Update to [v1.1.4] - 2023-05-19
* Moved discussions from Google Group to Discourse
* Fixed typo in command line argument validation
* Refactored RTP forwarder internals as a core feature [PR-3155]
* Refactored SVC processing as a core feature, and removed deprecated VP9/SVC demo [PR-3174]
* Don't create IPv6 sockets if IPv6 is completely disabled [PR-3179]
* Fixed some VideoRoom race conditions [PR-3167]
* Added simulcast/SVC params to switch in VideoRoom (thanks @brave44!) [PR-3197]
* Add support for receiving offers in Streaming plugin (for WHEP) [PR-3199]
* Add newline for SIP headers that are overflown in length (thanks @zayim!) [PR-3184]
* Save SIP reason state on multiple callbacks (thanks @kenangenjac!) [PR-3210]
* Avoid parsing whitespace as invalid JSON when receiving WebSocket messages (thanks @htrendev!) [PR-3208]
* Remove old tracks before adding/replacing new ones in janus.js [PR-3203]
* Tweaks to some janus.js internals (thanks @i8-pi!) [PR-3211]
* Fixed some typos and added some tweaks to Admin API demo
* Refactored npm version of janus.js
* Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)
- update _service, _servicedata file
- fix License to GPL-3.0-or-later
- add janus-gateway.sysusers for group/user creation
- Update dependencies
* build with libpcap
* --enable-plugin-lua
* --enable-librtp2 (swith to libsrtp2)
* Wed Dec 07 2022 michael@stroeder.com
- Update to version 1.1.1:
* Added timing info on ICE starting and gathering done to Admin API
* Fixed rare crash when generating SDP to send [Issue-3081]
* Fixed rare crash when checking payload types (thanks @zevarito!) [PR-3086]
* Fixed double a=ssrc attribute in SDP for inactive m-line
* Replaced non-portable strcasestr() with strncasecmp() (thanks @iskraman!) [PR-3076]
* Fixed parameters not being URL-encoded when using TURN REST API [Issue-3112]
* Fixed renegotiation sent to VideoRoom subscribers when a room is destroyed [Issue-3083]
* Added option to prevent automatic SDP offer updates to VideoRoom subscribers when a publisher leaves
* Fixed "send" property not being automatically reset to "true" in the VideoRoom for new subscriptions
* Fixed small memory leak in AudioBridge (thanks @RSATom!) [PR-3088]
* Minor fixes to the Streaming plugin
* Enforced media direction policies when SIP call is on hold PR-3087]
* Added code to send PLI to SIP peer when recording [PR-3093]
* Fixed renegotiations in VideoCall not updating session properties
* Other smaller fixes and improvements
* Tue Oct 04 2022 michael@stroeder.com
- Update to version 1.1.0:
* Added versioning to .so files [PR-3075]
* Allow plugins to specify msid in SDPs [PR-2998]
* Fixed broken RTCP timestamp on 32bit architectures [Issue-3045]
* Fixed problems compiling against recent versions of libwebsockets [Issue-3039]
* Updated deprecated DTLS functions to OpenSSL v3.0 PR-3048]
* Switched to SHA256 for signing self signed DTLS certificates (thanks @tgabi333!) [PR-3069]
* Started using strnlen to optimize performance of some strlen calls (thanks @tmatth!) [PR-3059]
* Added checks to avoid RTX payload type collisions [PR-3080]
* Added new APIs for cascading VideoRoom publishers [PR-3014]
* Fixed deadlock when using legacy switch in VideoRoom [Issue-3066]
* Fixed disabled property not being advertized to subscribers when VideoRoom publishers removed tracks
* Fixed occasional deadlock when using G.711 in the AudioBridge [Issue-3062]
* Added new way of capturing devices/tracks in janus.js [PR-3003]
* Removed call to .stop() for remote tracks in demos [PR-3056]
* Fixed missing message/info/transfer buttons in SIP demo page
* Fixed postprocessing compilation issue on older FFmpeg versions [PR-3064]
* Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)
* Mon Aug 01 2022 michael@stroeder.com
- Update to version 1.0.4:
* Fixed problem with duplicate ptypes when codecs are added in renegotiations
* Added codec info to event handlers stats
* Allow offers to include other roles besides 'actpass' for DTLS [PR-3020]
* Fixed rare race conditions when attempting to relay packets sent by plugins [PR-3010]
* Fixed unprotected access to medium instances in janus_plugin_handle_sdp
* Set appropriate channel type when sending DATA_CHANNEL_OPEN_REQUEST message (thanks @ktyu!) [PR-3018]
* Fixed rare race condition when handling incoming RTCP feedback in VideoRoom
* Fixed memory leak in VideoRoom when using rid-based simulcast (thanks @OxleyS!) [PR-2995]
* Fixed IPv6 always enabled for VideoRoom RTP forwarders [Issue-3011]
* Start recording VideoRoom publisher on PeerConnection establishment, if needed (thanks @adnanel!) [PR-3013]
* Added an optional ID in subscribe requests to match with subscription events (thanks @JanFellner!) [PR-3027]
* Make Streaming plugin use SDP utils, and codecs instead of rtpmaps [PR-2994]
* Check response codes of RTSP requests in Streaming plugin [Issue-3015]
* Fixed small memory leak in SIP plugin [Issue-3032]
* Fixed broken simulcast support in Lua and Duktape plugins
* Don't use .clone() on tracks to render them in demos [PR-3009]
* Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)
* Mon Jun 20 2022 michael@stroeder.com
- Update to version 1.0.3:
* Keep track of RTP extensions when storing packets for retransmission [PR-2981]
* Fixed negotiation of RTP extensions when direction is involved
* Fixed broken VP8 payload descriptor parsing when 7-bit PictureID are used
* Support for batched configure requests in VideoRoom [PR-2986]
* Added missing info to VideoRoom publisher's info own event [Issue-2988]
* Fixed memory leaks in when upgrading old-style Videoroom requests (thanks @OxleyS!) [PR-3002]
* Fixed memory leak in VideoRoom when updating subscriptions with no changes
* Added 'kick_all' requests and possibility to remove PIN code to both
Audiobridge and Streaming plugins (thanks @mikaelnousiainen!) [PR-2978]
* Added support for notifications in the Streaming plugin when metadata
for a mountpoint is changed (thanks @amoizard!) [PR-3000]
* Fixed missing checks on auth challenges in SIP plugin
* Fixed missing Contact header in SUBSCRIBE requests in SIP plugin [PR-2973]
* Fixed segfault in SIP plugin when freeing a session with a
subscription still active [PR-2974]
* Add new shared JavaScript file for settings in demos [PR-2991]
* Other smaller fixes and improvements
* Mon May 23 2022 michael@stroeder.com
- Update to version 1.0.2:
* Support for dummy publishers in VideoRoom (#2958)
* Fixed RED parsing not returning blocks when only primary data is available
* Link to -lresolv explicitly when building websockets transport
* src/dtls-bio.h: fix build with libressl >= 3.5.0 (#2980)
* Temporarily increase VideoRoom subscriber reference while we're creating it (see #2953)
* Fix address size in Streaming plugin RTCP sendto call (see #2976)
* Make SIP timer T1X64 configurable (#2972)
* Added custom headers for SIP SUBSCRIBE requests (#2971)
* Fixed incorrect removal of owner/subscriptions mapping (fixes #2965)
* Explicitly return list of IDs VideoRoom users are subscribed to for data (fixes #2967)
* Fixed typo in stereo support in EchoTest plugin
* Added configurable property to put a cap to task threads (see #2964)
* Disable IPv6 in WebSockets transport if binding to IPv4 address explicitly (fixes #2969)
* Abort DTLS handshake if DTLSv1_handle_timeout returns an error
* Fixed data port not being returned when creating Streaming mountpoints with the legacy API
* Fixed rtx SSRC incorretly added to SDP when disabled
* Fixed rtx not being offered on Janus originated connections
* Better cleanup when closing PeerConnection in multistream VideoRoom demo (see #2942)
* Tue Apr 26 2022 michael@stroeder.com
- Update to version 1.0.1:
* release v1.0.1
- Removed gengetopt as a dependency, to use Glib's GOptionEntry instead [PR-2898]
- Fixed occasional problem of duplicate mid attribute in Janus SDPs [Issue-2917]
- Fixed receiving=false events not being sent right away for higher simulcast substreams [Issue-2919]
- Fix highest sequence number not being properly initialized in the RTCP context [Issues-2920]
- Reset rids when renegotiating SDPs [PR-2931]2931)]
- Fixed missing PLI when restoring previously paused streams in VideoRoom (thanks @flaviogrossi!) [PR-2922]
- Fixed deadlock when using the moderate API in the VideoRoom [Issue-2956]
- Check if IPv6 is disabled to avoid failure when creating forwarder sockets in AudioBridge and VideoRoom [PR-2916]
- Fixed invalid computation of Streaming mountpoint stream age (thanks @RouquinBlanc!) [PR-2928]
- Also return reason header protocol and cause if present in BYE in the SIP plugin (thanks @ajsa-terko!) [PR-2935]
- Fixed segfault in UNIX transport teardown caused by pathnames of different sizes
- Added new demos on WebAudio and Virtual Backgrounds [PR-2941]
- Fixed potential race conditions in multistream VideoRoom demo [Issue-2929]
- Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)
* release v1.0.0
- Refactored Janus to support multistream PeerConnections [PR-2211]
- Moved all source files under new 'src' folder to unclutter the repo [PR-2885]
- Fixed definition of trylock wrapper when using pthreads [Issue-2894]
- Fixed broken RTP when no extensions are negotiated
- Added checks when inserting RTP extensions to avoid buffer overflows
- Added missing support for disabled rid simulcast substreams in SDP [PR-2888]
- Fixed TWCC feedback when simulcast SSRCs are missing (thanks @OxleyS!) [PR-2908]
- Added support for playout-delay RTP extension [PR-2895]
- Fixed partially broken H.264 support when using Firefox in VideoRoom
- Fixed new VideoRoom rtp_forward API ignoring some properties
- Fixed deadlock and segfault when stopping Streaming mountpoint recordings [Issue-2902]
- Fixed RTSP support in Streaming plugin for cameras that expect path-only DESCRIBE requests (thanks @jp-bennett!) [PR-2909]
- Fixed RTP being relayed incorrectly in Lua and Duktape plugins
- Added Duktape as optional dependency, instead of embedding the engine code [PR-2886]
- Fixed crash at startup when not able to connect to RabbitMQ server
- Improved fuzzing and checks on RTP extensions
- Removed distinction between simulcast and simulcast2 in janus.js [PR-2887]
- Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)
* Fri Feb 11 2022 michael@stroeder.com
- Update to version 0.11.8:
* Added initial (and limited) integration of RED audio (#2685)
* Added support for Two-Byte header RTP extensions (RFC8285) and, partially,
for the new Depencency Descriptor RTP extension (needed for AV1-SVC) (#2741)
* Fixed rare race conditions between sending a packet and closing a connection (#2869)
* Fix last stats before closing PeerConnection not being sent to handlers (#2874)
* Changed automatic allocation on static loops from round robin to least used (#2878)
* Added new API to bulk start/stop MJR-based recordings in AudioBridge (#2862)
* Fixed broken duration in spatial AudioBridge recordings
* Fixed broken G.711 RTP forwarding in AudioBridge (#2875)
* Fixed broken recordings in NoSIP plugin
* Fixed warnings when postprocessing Opus recordings with DTX packets
* Other smaller fixes and improvements
* Mon Jan 24 2022 michael@stroeder.com
- Update to version 0.11.7:
* Added faster strlcat variant that uses memccpy for writing SDPs (#2835)
* Fixed occasional crash when updating WebRTC sessions (#2840)
* Changed SDP syntax for AV1 from "AV1X" to "AV1" (#2844)
* Fixed signed_tokens property not being saved to permanent rooms in VideoRoom (#2843)
* Made record directory changeable via "edit" in both AudioBridge and VideoRoom
* Added configurable expected loss to AudioBridge to actually send FEC (#2802)
* Fixed SIP plugin not working when using Sofia SIP >= 1.13 (#2683)
* Fixed occasional crashes in SIP plugin (#2853)
* Take note of video orientation extension when recording video in SIP plugin (#2836)
* Allow 180 besides 183 to have SDP as well (#2849)
* Fixed post-processor compilation issue with newer versions of FFmpeg (#2833)
* Added option to print extended info on MJR file as JSON in postprocessor (#2858)
* Allow pcap2mjr to autodetect SSRC
* Fixed problems compiling post-processor with older versions of FFmpeg
* Other smaller fixes and improvements
* Mon Dec 13 2021 michael@stroeder.com
- Update to version 0.11.6:
* Fixed CVE-2021-4020 (see also boo#1193156):
Cross-site Scripting (XSS) vulnerability in some of the demos (#2817)
* Added strlcat helper to detect and report truncations (#2792)
* Grow buffer as needed when generating SDPs (#2797)
* Added DTX support to some plugins (#2789)
* Added option to forcibly quit Janus when getting dlopen errors (#2828)
* Fixed broken signed tokens in VideoRoom when using UUIDs (#2812)
* Added option to choose whether signed tokens should be used in the
VideoRoom when enabled in the core (#2825)
* Added configurable expected loss to AudioBridge to actually send FEC (#2802)
* Added MESSAGE authentication and out-of-dialog MESSAGE support to SIP plugin (#2786)
* Fixed potential race conditions in SIP plugin (#2823)
* Added basic history support to TextRoom plugin (#2814)
* Added support for custom datachannel options in janus.js (#2806)
* Other smaller fixes and improvements
* Mon Oct 18 2021 michael@stroeder.com
- dropped obsolete 0001-include-rand-header-file.patch
- Update to version 0.11.5:
* Add API to optionally force Janus to use TURN (#2774)
* Fixed slow path on SDP parsing (#2776)
* Added event handlers option to send stats for a PeerConnection
in a single event, rather than per-media (#2785)
* Fixed occasional deadlocks on malformed requests in VideoRoom (#2780)
* Fixed AudioBridge plain RTP thread sometimes exiting prematurely
* Fixed broken upsampling when using G.711 in AudioBridge
* Add pause/resume recording functionality to Record&Play and SIP plugins (#2724)
* Fixed broken support for Unix Sockets in WebSockets Admin API (#2787)
* Added timing info for video rotation when post-processing recordings
* Added linter checks to janus.js (#2272)
* Other smaller fixes and improvements
* Thu Sep 23 2021 Johannes Segitz <jsegitz@suse.com>
- Added hardening to systemd service(s) (bsc#1181400). Modified:
* janus.service
* Mon Sep 06 2021 Michael Ströder <michael@stroeder.com>
- added janus-gateway-rpmlintrc
- removed systemd-related conditionals to fix obsolete-suse-version-check
- added 0001-include-rand-header-file.patch
- use %fdupes macro
- Update to version 0.11.4:
* Fixed ICE restart issues with recent versions of libnice (#2729)
* Changed randon number generators to use crypto-safe functions (#2738)
* Added support for abs-send-time RTP extension (#2721)
* Added configurable mechanism for manually setting static event loop to use for new handles (#2684)
* Fixed datachannel protocol not being sent to plugins for incoming messages (#2753)
* Added ability to specify recordings folder in AudioBridge (#2707)
* Added support for forwarding groups in AudioBridge (#2653)
* Fixed missing Contact header in SIP plugin when using Sofia >= 1.13 (#2708)
* Better SDES-SRTP negotiation in SIP and NoSIP plugins (#2727)
* Fixed WebSocket transport and event handler lagging 25/30s when shutting down or reconnecting (#2734)
* Fixed incoming_header_prefixes not working for helper sessions in SIP plugin
* Fix partial/broken ACL support in TextRoom plugin (#2763)
* Fixed potential race condition when reclaiming sessions in HTTP transport plugin
* Fixed WebSocket event handler reconnect mechanism (#2736)
* Other smaller fixes and improvements
* Tue Jun 15 2021 michael@stroeder.com
- Update to version 0.11.3:
* Fixed rare crash when detaching handles (#2464)
* Added option to offer IPv6 link-local candidates as well (#2689)
* Added spatial audio support to AudioBridge via stereo mixing (#2446)
* Added support for plain RTP participants to AudioBridge (#2464)
* Added API to start/stop AudioBridge recordings dynamically
(thanks @rajneeshksoni!) (#2674)
* Fixed broken mountpoint switching when using different payload types
in Streaming plugin (#2692)
* Fixed occasional deadlock on Streaming plugin mountpoint destroy
during RTSP reconnects (thanks @lionelnicolas!) (#2700)
* Added "Expires" support to SUBSCRIBE in SIP plugin
(thanks @nicolasduteil!) (#2661)
* Added option to specify Call-ID for SUBSCRIBE dialogs in SIP plugin
(thanks @nicolasduteil!) (#2664)
* Fixed broken simulcast support in VideoCall plugin
(thanks @lucily-star!) (#2671)
* Implemented RabbitMQ reconnection logic, in both transport and event handler
(thanks @chriswiggins!) (#2651)
* Added support for renegotiation of external streams in janus.js
(thanks @kmeyerhofer!) (#2604)
* Added support for HEVC/H.265 aggregation packets (AP) to janus-pp-rec
(thanks @nu774!) (#2662)
* Refactored janus-pp-rec to cleanup the code, and use libavformat for Opus as well
(thanks @lu-zero!) (#2665)
* Added additional target formats for some recorded codecs (#2680)
* Other smaller fixes and improvements
* Mon May 03 2021 michael@stroeder.com
- Update to version 0.11.2:
- Added support for relative paths in config files, currently only in
MQTT event handler (thanks @RSATom!) (#2623)
- Removed support for now deprecated frame-marking RTP extension
(#2640)
- Fixex rare race condition between VideoRoom publisher leaving and
subscriber hanging up (#2637)
- Fixed occasional crash when using announcements in AudioBridge
- Fixed rare crash in Streaming plugin when reconnecting RTSP streams
(thanks @lucylu-star!) (#2542)
- Fixed broken switch in Streaming plugin when using helper threads
- Fixed rare race conditions on socket close in SIP and NoSIP plugins
(#2599)
- Added support for out-of-dialog SIP MESSAGE requests (thanks
@ihusejnovic!) (#2616)
- Fixed memory leak when using helper threads in Streaming plugin
- Added support for datachannel label/protocol to Lua and Duktape
plugins (#2641)
- Added ability to use WebSockets transport over Unix sockets (thanks
@mdevaev!) (#2620)
- Added janus-pp-rec mechanism to correct wrong RTP timestamps in MJR
recordings (thanks @tbence94!) (#2573)
- Other smaller fixes and improvements
* Tue Apr 06 2021 michael@stroeder.com
- Update to version 0.11.1:
* Add new option to configure ICE nomination mode, if libnice is recent enough (#2541)
* Added support for per-session timeout values (thanks @alg!) (#2577)
* Added support for compilation on FreeBSD (thanks @jsm222!) (#2508)
* Fixed occasional auth errors when using both API secret and stored tokens (#2581)
* Added support for stdout logging to daemon-mode as well (#2591)
* Fixed odr-violation issue between Lua and Duktape plugins (#2540)
* Fixed missing simulcast stats in Admin API and Event Handlers when using rid (#2610)
* Fixed VideoRoom recording not stopped for participants entering after global recording was started (#2550)
* Fixed 'audiocodec'/'videocodec' being ignored when joining a VideoRoom via 'joinandconfigure'
* Added content type support to MESSAGE in SIP plugin (#2567)
* Made RTSP timeouts configurable in Streaming plugin (#2598)
* Fixed incorrect parsing of backend URL in WebSockets event handler (#2603)
* Added support for secure connections and lws debugging to WebSockets event handler
* Fixed occasionally broken AV1 recordings post-processing
* Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)
* Mon Feb 08 2021 michael@stroeder.com
- Update to version 0.10.10:
* Reduced verbosity of a few LOG_WARN messages at startup
* Close libnice agent resources asynchronously when hanging up PeerConnections (thanks @fbellet!) [PR-2492]
* Fixed broken parsing of SDP when trying to match specific codec profiles [PR-2549]
* Added muting/moderation API to the VideoRoom plugin [PR-2513]
* Fixed a few race conditions in VideoRoom plugin that could lead to crashes [[PR-2539][#2539)]
* Send 480 instead of BYE when hanging up calls in early dialog in the SIP plugin (thanks @zayim!) [PR-2521]
* Added configurable media direction when putting calls on-hold in the SIP plugin [PR-2525]
* Fixed rare race condition in AudioBridge when using "changeroom" (thanks @JeckLabs!) [[PR-2535][#2535)]
* Fixed broken API secret management in HTTP long polls (thanks @remvst!) [PR-2524]
* Report failure if binding to a socket fails in WebSockets transport plugin (thanks @Symbiatch!) [PR-2534]
* Updated RabbitMQ logic in both transport and event handler (thanks @chriswiggins!) [PR-2430]
* Fixed segfault in WebSocket event handler when backend was unreachable
* Added TLS support to MQTT event handler (thanks @RSATom!) [PR-2517]
* Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)
* Wed Dec 23 2020 michael@stroeder.com
- Update to version 0.10.9:
* Replaced Travis CI with GitHub Actions [[PR-2486](#2486)]
* Fixed data channel messages potentially getting stuck in case of burst transfers (thanks @afshin2003!) [[PR-2427](#2427)]
* Fixed simulcast issues when renegotiating PeerConnections [[Issue-2466](#2466)]
* Added configurable TURN REST API timeout (thanks @evorw!) [[PR-2470](#2470)]
* Added support for recording of binary data channels [[PR-2481](#2481)]
* Fixed occasional SRTP errors when pausing and then resuming Streaming plugin handles after a long time
* Fixed occasional SRTP errors when leaving and joining AudioBridge rooms without a new PeerConnection after a long time
* Added support for playout of data channels in Record&Play plugin and demo (thanks @ricardo-salgado-tekever!) [[PR-2468](#2468)]
* Added option to override connections limit in HTTP transport plugin [[PR-2489](#2489)]
* Added options to enable libmicrohttpd debugging in HTTP transport plugin (thanks @evorw!) [[PR-2471](#2471)]
* Fixed a few compile and runtime issues in WebSocket event handler
* Refactored postprocessing management of timestamps to fix some key issues [[PR-2345](#2345)]
* Fixed postprocessing of audio recordings containing RTP silence suppression packets [[PR-2467](#2467)]
* Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)
* Tue Dec 22 2020 Jan Engelhardt <jengelh@inai.de>
- Do not hard-require systemd. Drop redundant wording from
description. Drop %defattr.
* Mon Dec 21 2020 Michael Ströder <michael@stroeder.com>
- Initial packaging of 0.10.8 for Factory
/usr/include/janus /usr/include/janus/apierror.h /usr/include/janus/config.h /usr/include/janus/debug.h /usr/include/janus/events /usr/include/janus/events/eventhandler.h /usr/include/janus/ip-utils.h /usr/include/janus/log.h /usr/include/janus/loggers /usr/include/janus/loggers/logger.h /usr/include/janus/mutex.h /usr/include/janus/plugins /usr/include/janus/plugins/plugin.h /usr/include/janus/record.h /usr/include/janus/refcount.h /usr/include/janus/rtcp.h /usr/include/janus/rtp.h /usr/include/janus/rtpsrtp.h /usr/include/janus/sdp-utils.h /usr/include/janus/text2pcap.h /usr/include/janus/transports /usr/include/janus/transports/transport.h /usr/include/janus/utils.h
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