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janus-gateway
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File janus-gateway.changes of Package janus-gateway
------------------------------------------------------------------- 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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