File janus-gateway.changes of Package janus-gateway

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File janus-gateway.changes of Package janus-gateway (Revision df5df2ac1c47844ff24d507064dca82d)

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-------------------------------------------------------------------
Sat Jan 13 22:09:55 UTC 2024 - chris@computersalat.de

- Update to v1.2.1 - 2023-12-06
  * Added support for abs-capture-time RTP extension [PR-3161]
  * Fixed truncated label in datachannels (thanks @veeting!) [PR-3265]
  * Support larger values for SDP attributes (thanks @petarminchev!) [PR-3282]
  * Fixed rare crash in VideoRoom plugin (thanks @tmatth!) [PR-3259]
  * Don't create VideoRoom subscriptions to publisher streams with no associated codecs
  * Added suspend/resume participant API to AudioBridge [PR-3301]
  * Fixed rare crash in AudioBridge
  * Fixed rare crash in Streaming plugin when doing ICE restarts [Issue-3288]
  * Allow SIP and NoSIP plugins to bind media to a specific address (thanks @pawnnail!) [PR-3263]
  * Removed advertised support for SIP UPDATE in SIP plugin
  * Parse RFC2833 DTMF to custom events in SIP plugin (thanks @ywmoyue!) [PR-3280]
  * Fixed missing Contact header in some dialogs in SIP plugin (thanks @ycherniavskyi!) [PR-3286]
  * Properly set mid when notifying about ended tracks in janus.js
  * Fixed broken restamping in janus-pp-rec
  * Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)
- Update to v1.2.0 - 2023-08-09
  * Added support for VP9/AV1 simulcast, and fixed broken AV1/SVC support [PR-3218]
  * Fixed RTCP out quality stats [PR-3228]
  * Default link quality stats to 100
  * Added support for ICE consent freshness [PR-3234]
  * Fixed rare race condition in VideoRoom [PR-3219] [PR-3247]
  * Use speexdsp's jitter buffer in the AudioBridge [PR-3233]
  * Fixed crash in Streaming plugin on mountpoints with too many streams [Issue-3225]
  * Support for batched configure requests in Streaming plugin (thanks @petarminchev!) [PR-3239]
  * Fixed rare deadlock in Streamin plugin [PR-3250]
  * Fix simulated leave message for longer string ID rooms in TextRoom (thanks @adnanel!) PR-3243]
  * Notify about count of sessions, handles and PeerConnections on a regular basis, when event handlers are enabled [PR-3221]
  * Fixed broken Insertable Streams for recvonly streams when answering in janus.js
  * Added background selector and blur support to Virtual Background demo
  * Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)

-------------------------------------------------------------------
Tue Oct 10 13:21:45 UTC 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 17:24:59 UTC 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 14:34:59 UTC 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 09:53:18 UTC 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 11:11:52 UTC 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 10:16:33 UTC 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 20:24:51 UTC 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 22:27:07 UTC 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 14:36:20 UTC 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 12:26:10 UTC 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 16:20:34 UTC 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 05:14:47 UTC 2021 - Johannes Segitz <jsegitz@suse.com>

- Added hardening to systemd service(s) (bsc#1181400). Modified:
  * janus.service

-------------------------------------------------------------------
Mon Sep  6 12:53:58 UTC 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 13:44:32 UTC 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 10:24:07 UTC 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 10:49:55 UTC 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 10:27:21 UTC 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 12:32:09 UTC 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 09:25:17 UTC 2020 - Jan Engelhardt <jengelh@inai.de>

- Do not hard-require systemd. Drop redundant wording from
  description. Drop %defattr.

-------------------------------------------------------------------
Mon Dec 21 19:16:41 UTC 2020 - Michael Ströder <michael@stroeder.com>

- Initial packaging of 0.10.8 for Factory

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