Index index by Group index by Distribution index by Vendor index by creation date index by Name Mirrors Help Search

janus-gateway-1.1.4-1.1 RPM for s390x

From OpenSuSE Ports Tumbleweed for s390x

Name: janus-gateway Distribution: openSUSE:Factory:zSystems
Version: 1.1.4 Vendor: openSUSE
Release: 1.1 Build date: Fri Oct 27 21:31:23 2023
Group: Productivity/Networking/Other Build host: s390zl23
Size: 10440674 Source RPM: janus-gateway-1.1.4-1.1.src.rpm
Summary: Janus WebRTC Gateway
Janus is a general-purpose WebRTC gateway designed and developed
by Meetecho.






* Tue Oct 10 2023
  - 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
  - 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
  - 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
  - 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
  - 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
  - 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
  - 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
  - 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
  - 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
  - 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
  - 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 <>
  - Added hardening to systemd service(s) (bsc#1181400). Modified:
    * janus.service
* Mon Sep 06 2021 Michael Ströder <>
  - 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
  - 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
  - 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
    - 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
    - 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
  - 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
  - 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
  - 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 <>
  - Do not hard-require systemd. Drop redundant wording from
    description. Drop %defattr.
* Mon Dec 21 2020 Michael Ströder <>
  - Initial packaging of 0.10.8 for Factory



Generated by rpm2html 1.8.1

Fabrice Bellet, Sat Mar 9 12:50:11 2024