File janus-gateway.changes of Package janus-gateway

-------------------------------------------------------------------
Sun Jun 26 11:13:25 UTC 2022 - ecsos@opensuse.org

- Update to version 0.12.3:
  * Updated Changelog (0.12.3)
  * Use inet_pton instead of inet_net_pton
  * Only reset rid when processing video m-line (fixes #2992)
  * Add new shared JavaScript file for settings in demos (see #2991)
  * Fixed broken VP8 payload descriptor parsing when 7-bit PictureID are used
  * Fixed typo in destroy request of Streaming plugin
  * Fixed exception when adding helper in SIP plugin demo
  * Fixed missing contact header in SUBSCRIBE (#2973) and crash in SIP plugin when freeing a session while a subscription is active (2974)
  * Fixed negotiation of RTP extensions when direction is involved
  * Improved check on when to send playout-delay extension
  * Fixed missing checks on auth challenges in SIP plugin
  * Keep track of extensions when storing packets for retransmission (see #2981)
  * Fixed issues/PRs links in ChangeLog
  * Bumped to version 0.12.3 (legacy)

-------------------------------------------------------------------
Sun Jun 26 11:13:11 UTC 2022 - ecsos@opensuse.org

- Update to version 0.12.2:
  * Updated Changelog (0.12.2)
  * Fixed RED parsing not returning blocks when only primary data is available
  * Link to -lresolv explicitly when building websockets transport
  * Fix build with libressl >= 3.5.0 (see #2980)
  * Fix address size in Streaming plugin RTCP sendto call (#2976)
  * Make SIP timer T1X64 configurable (see #2972)
  *  Added custom headers for SIP SUBSCRIBE requests (see #2971)
  * Fixed spaces instead of tabs
  * Added synchronous request to start/stop recording single participant in VideoRoom
  * Fixed typo in stereo support in EchoTest plugin
  * Added configurable property to put a cap to task threads (see #2964)
  * Return an error when attempting to postprocess a non-MJR file
  * Disable IPv6 in WebSockets transport if binding to IPv4 address explicitly (see #2969)
  * Honor "audio", "data", "video" flag changes in subscriber updates. (#2963)
  * Abort DTLS handshake if DTLSv1_handle_timeout returns an error
  * Fixed error message being displayed incorrectly when creating a mountpoint
  * Bumped to version 0.12.2 (legacy)

-------------------------------------------------------------------
Sun Jun 26 11:12:44 UTC 2022 - ecsos@opensuse.org

- Update to version 0.12.1:
  * Updated Changelog (0.12.1)
  * Fix misaligned access in pp-rec when parsing VP8/VP9 frame resolution
  * Fix error message when creating a session (see #2890)
  * Also return reason header protocol and cause if present in BYE in the SIP plugin (see #2935)
  *  Better error when trying to rejoin on an existing VideoRoom handle
  * Extend H.265 keyframe checks to more NALs (see #2323)
  * videoroom: always default {min,max}_delay to -1 (#2936)
  * Check if IPv6 is disabled to avoid failure when creating forwarder socket (see #2915 and #2916)
  * Reset rids when renegotiating SDPs (see #2927 and #2931)
  * Fix segfault in UNIX transport teardown when pathnames have different sizes.
  * Fix highest sequence number not being properly initialized in the RTCP context (see #2920)
  * Added new request to SIP plugin to reset the establishing flag if it's stuck
  * Fixed typo in config sample (missing quote) (see #2912)
  * Bumped to version 0.12.1 (legacy)

-------------------------------------------------------------------
Sun Jun 26 11:11:25 UTC 2022 - ecsos@opensuse.org

- Update to version 0.12.0:
  * Updated Changelog (0.12.0)
  * Fixed typos (see #2909)
  * Add quirk for RTSP servers (#2909)
  * Fix transport-wide CC feedback when simulcast SSRCs are missing (see #2908)
  * Update version number in npm package (see #2901)
  * Add support for playout-delay RTP extension (see #2895)
  * Fixed leftover variable in janus.js
  * Fix some other build warnings (output should be clean now).
  * Added missing fix related to #2894 (0.x)
  * Fix definition of trylock wrapper when using pthreads (see #2894)
  * Link to Duktape engine as a library (see #2886)
  * Remove distinction between simulcast and simulcast2 in janus.js (see #2887)
  * Added checks when inserting RTP extensions to avoid buffer overflows
  * Fix some build warnings.
  * Keep track of when simulcast substreams are disabled in SDP (see #2888)
  * Fixed broken RTP when no extensions are negotiated (see #2881)
  * Link to CONTRIBUTING file from README instead of ISSUE_TEMPLATE folder
  * Fixed crash at startup when not able to connect to RabbitMQ server
  * Bumped to version 0.12.0 (legacy)
  * Fix links in Changelog
  * Make 0.x more apparent in README
  * Clarify this is the 0.x branch in the README, and update links to demos/docs
  * Preliminary changes for 0.x branch of Janus

-------------------------------------------------------------------
Sun Jun 26 10:56:58 UTC 2022 - ecsos@opensuse.org

- Update to version 0.11.8:
  * Updated Changelog (0.11.8)
  * Change automatic allocation on static loops from round robin to least loaded (#2878)
  * Fixed outdated janus-pp-rec option description
  * Changed default distance in postprocessor to 0, and removed unneeded DTX flag
  * Check continuity of Opus packets when postprocessing (see #2880, for DTX)
  * Reset extensions struct when not used
  * fix PCMA/PCMU RTP forwarding in audiobridge - incorrect RTP header offset (#2875)
  * Fixed runtime error
  * Update RTP extensions for outgoing packets in the PeerConnection loop (fixes #2867) (#2869)
  * Added link to FOSDEM2022 presentation to FAQ
  * Fix last stats before closing PeerConnection not being sent to handlers (replaces #2873) (#2874)
  * janus: add yet another missing NULL check for opusred (#2872)
  * Add another missing NULL check on stream when setting opus RED
  * Modified issues template
  * janus: add missing NULL check on stream (#2865)
  * Fixed broken duration in spatial AudioBridge recordings
  * Fixed ambiguity in AudioBridge documentation (fixes #2863)
  * Add new API to bulk start/stop MJR-based recordings in AudioBridge (#2862)
  * Remove deprecated issue template
  * Update issue templates
  * Fixed missing check after merge
  * Initial support for AV1-SVC Dependency Descriptor (#2741)
  * Initial integration of RED for audio (#2685)
  * Fixed broken recordings in NoSIP plugin
  * Add a couple of checks after static analysis
  * Bumbed to version 0.11.8

-------------------------------------------------------------------
Sun Jun 26 10:56:39 UTC 2022 - ecsos@opensuse.org

- Update to version 0.11.7:
  * Updated Changelog (0.11.7)
  * Fixed problems compiling post-processor with older versions of FFmpeg
  * Added option to print extended header in janus-pp-rec (inspired by #2838) (#2858)
  * Make rtp port range variables static in ice.c to prevent clashes with plugins (#2860)
  * Added number of subscribers in response to listpartipants (fixes #2856)
  * Added more checks before dereferencing
  * Make record directory changeable via edit in AudioBridge and VideoRoom
  * Allow pcap2mjr to autodetect SSRC
  * Validate call_id when handling calls and messages (see #2853)
  * Add strerror to errno-related log lines in SCTP code
  * Protect updates to callid in SIP plugin (see #2853)
  * Fixed broken 'sips' in SIP plugin with Sofia >= 1.13 (see #2683)
  * Disable sips by default in SIP plugin (fixes issue with Sofia >= 1.13, see #2683)
  * fix saving signed_tokens field when room is permanent (#2843)
  * Change SDP syntax for AV1 from "AV1X" to "AV1" (fixes #2844)
  * Handle sdp in 180 sip message same as 183 (#2849)
  * html: update webrtc-adapter to 8.1.1 (#2848)
  * janus.d.ts typing fixes (#2847)
  * Add reason when failing to open DTLS cert/key file (see #2845)
  * Updated year in demos and docs
  * Add NULL checks for stream and component in janus_ice_media_stopped (fixes #2840)
  * janus.d.ts missing typing (#2837)
  * Remove unneeded codec configuration in janus-pp-rec (fixes #2833)
  * Add configurable expected loss to AudioBridge to actually send FEC (#2802)
  * Fixed error in Changelog
  * Small code style tweaks
  * Take note of video orientation extension when recording video in SIP plugin (#2836)
  * Add strlcat variant that uses memccpy for writing SDPs (#2835)
  * Fixed missing XSS mitigation (see #2817)
  * Bumbed to version 0.11.7
  * Fixed warning

-------------------------------------------------------------------
Sun Jun 26 10:56:20 UTC 2022 - ecsos@opensuse.org

- Update to version 0.11.6:
  * Updated Changelog (0.11.6)
  * Reject SDPs with invalid media types, and realloc less often when writing one
  * Moved from linux/ip.h to netinet/ip.h (#2831)
  * Fix syntax error in VideoRoom (#2832)
  * Add transport-cc to video rtcp-fb list too
  * Fixed extra CRLF being appended when adding extensions in offers
  * Added Janode presentation to the docs
  * Allow VideoRoom to choose whether signed tokens should be used (#2825)
  * Add missing error code in SIP plugin (fixes #2830)
  * conf: add exit_on_dl_error option (#2828)
  * Protect removal from hashtables when destroying SIP sessions (fixes #2818) (#2823)
  * Added more videos to the documentation
  * Removed loop initial declarations (fixes errors on some compilers)
  * Send receiving:false notifications right after renegotiating (see #2807 and #2808) (#2824)
  * Added additional check when mixing in AudioBridge
  * Added basic history support to TextRoom plugin (#2814)
  * Fix potential Cross-site Scripting (XSS) exploits in demos (#2817)
  * Fixed typos
  * Create SECURITY.md (fixes #2815)
  * Fix warning and remove provided token from error response
  * add typescript client lib (#2813)
  * Fix signed token auth not work on join to the videoroom #2810 (#2812)
  * Disable slowlink events by default (#2803)
  * plugins/janus_sip.c: MESSAGE Authentication and Deliver Status Report (#2786)
  * Added support for multiopus to janus-pp-rec
  * add support for custom datachannel options in janus.js (#2806)
  * Added janode to the list of resources in the docs
  * html: update webrtc-adapter to 8.1.0 (#2798)
  * Fixed incorrect info in SIP plugin documentation
  * Grow buffer as needed when generating SDPs (fixes #2791, see #2793) (#2797)
  * Added missing unrefs (fixes #2795)
  * utils: add janus_strlcat helper (#2792)
  * Initial support for DTX (EchoTest, VideoRoom) (#2789)
  * Added new presentation video to the documentation
  * Fixed typos in Streaming demo
  * Fix plain HTTP links
  * Bumbed to version 0.11.6

-------------------------------------------------------------------
Sun Jun 26 10:56:02 UTC 2022 - ecsos@opensuse.org

- Update to version 0.11.5:
  * Reverted version bump
  * Bumbed to version 0.11.6
  * Updated Changelog (0.11.5)
  * Small tweaks to #2785
  * Allow media statistic events to get dispatched as one event per stream instead of dedicated ones per media (#2785)
  * Fix WebSockets admin unix sockets, and streamlined parsing code (#2787)
  * Avoid ICE local setup if handle has been destroyed (fix some memory leaks)
  * Print time of when video is rotated, if available
  * Add info on incoming REMB values to Admin API and Event handlers
  * Add the token used to join the room to the info sent to event handlers (#2773)
  * Ensure we access room uniformly and safely across all the VideoRoom code (#2782)
  * Added some IDE notes to the docs (see #2784)
  * Fixed systemd sample in documentation (#2783)
  * Add pause/resume recording functionality to Record&Play and SIP plugins (#2724)
  * Insert private ID mapping only when participant is added to the list (fixes #2781)
  * Fix missing unlock/unref in case of publisher errors (fixes #2780).
  * Make janus.js pass linter (#2772)
  * Add API to force Janus to use TURN (#2774)
  * Use prepend with reverse list in the sdp parsing (Credit to OSS-Fuzz) (#2776)
  * handle host being in ignore list and enforced list (#2768)
  * Fixed broken upsampling in AudioBridge
  * Add query string to force codec when joining AudioBridge demo
  * Moved pragma precondition from the middle of an if (fixes #2766)
  * Fixed AudioBridge plain RTP thread sometimes exiting prematurely
  * Clarify in documentation that the mobile SDK is abandoned
  * Fix compilation warning on openSUSE
  * Bumbed to version 0.11.5

-------------------------------------------------------------------
Sun Jun 26 10:55:46 UTC 2022 - ecsos@opensuse.org

- Update to version 0.11.4:
  * Updated Changelog (0.11.4)
  * Doxygen tweaks
  * Fixed compilation warning
  * Fix streaming mountpoint name generation when starting with 0 (#2764)
  * Added new project to resources in the docs
  * Added plain RTP participants documentation to AudioBridge
  * Clearer indication of payload types when using plain RTP participants in AudioBridge
  * Fixed optional arguments in Duktape datachannel relay methods
  * Fixed datachannel protocol not being sent to plugins for incoming messages (fixes #2753)
  * Fix partial/broken ACL support in TextRoom plugin (#2763)
  * Configurable mechanism for manually setting static event loop to use for new handles (see #2450) (#2684)
  * Use crypto safe random numbers (#2738)
  * Adds reconnect() to the types definition (#2745)
  * Added support for forwarding groups in AudioBridge (#2653)
  * Fixed README (see #2744)
  * Support for abs-send-time RTP extension (#2721)
  * Fix sample event handler behaviour (#2743)
  * Fixed link to usrsctp in documentation
  * Fixed indentation in README
  * Update README.md (#2744)
  * Fixed broken links in Changelog
  * Fixed broken links in Changelog
  * Fixed typo in comment
  * Fixed incoming_header_prefixes not working for helper sessions in SIP plugin
  * Fixed incoming_header_prefixes not working for helper sessions in SIP plugin
  * Videoroom API "list" endpoint: update docs, add is_private (#2715)
  * Improve websocket event handler reconnection handling on fail
  * Log connection error reason, in WS event handler
  * Set new remote credentials after restarting ICE (fixes #2672) (#2729)
  * Rename macro constant in ws event handler.
  * Improve ws event handler reconnection and set an upper bound for the backoff (fixes #2734).
  * Better SRTP-SDES negotiation in SIP/NoSIP plugins (fixes #2726) (#2727)
  * Wake up the lws loop when destroying the WebSocket transport.
  * Extend clang matching rule (let autoconf match afl-clang-fast in oss-fuzz).
  * Revert "Remove unneeded definition of RAND_bytes from the RTP fuzzer."
  * Remove unneeded definition of RAND_bytes from the RTP fuzzer.
  * [janus-pp-rec] activate keyframe logic for VP9 (#2730)
  * Fix SIP plugin doxygen command for section reference (#2732)
  * Manually add SIP Contact header for calls when using Sofia >= 1.13 (fixes #2439, replaces #2597) (#2708)
  * Use PKG_CONFIG env variable instead of calling pkg-config directly (fixes #2713)
  * Fix sending BYE (#2709)
  * demos: update webrtc adapter 8.0.0 (#2702)
  * Fixed some warnings when compining SIP plugin
  * Add ability to specify recordings folder in AudioBridge (#2707)
  * Fix potential race condition when reclaiming sessions in HTTP plugin
  * Small fixes after static analysis
  * Bumbed to version 0.11.4

-------------------------------------------------------------------
Sun Jun 26 10:55:27 UTC 2022 - ecsos@opensuse.org

- Update to version 0.11.3:
  * Updated Changelog (0.11.3)
  * Fix missing secret check on room enable_recording (#2706)
  * Fixed AudioBridge recording stop/start not working properly on empty rooms (see #2674)
  * Addess notes to #2674 (missing validation, extra timestamp in static filename)
  * New enable_recording API to dynamically start/stop AudioBridge recordings (#2674)
  * Fix deadlock on mountpoint destroy during RTSP reconnect (#2700)
  * Fix SIP plugin missed_call event call_id field name typo (#2703)
  * Fix memory leak on aborted RTSP connection (#2699)
  * Fixed broken switching when using different payload types in Streaming plugin (#2692)
  * Fixes on JavaScript code snippets (#2695)
  * Fixed typo in docs
  * Added missing semicolon (#2693)
  * Optionally allow IPv6 link-local addresses to be gathered too (#2689)
  * Additional target formats for some recorded codecs (fixes #2658) (#2680)
  * Fix streaming plugin mutex unlock when disabling mountpoint (#2690)
  * Fix SIP plugin unhold request docs typo (#2688)
  * minor adjustment to the audiobridge docs (#2687)
  * fix: [janus_sip] Fix "call_id" property in "missed_call" events (#2679)
  * Fix status vector parsing for incoming twcc feedbacks (resolves #2677).
  * Fixed race condition in VideoRoom
  * Fixes variable name.
  * Clarify that libnice 0.1.18 is recommended
  * Spatial audio support in AudioBridge via stereo mixing (#2446)
  * Cleanup avformat-based preprocessors (#2665)
  * Fixed broken simulcast support in VideoCall plugin (#2671)
  * feat: support for custom call-id in subscribe request + add 'call_id' property to subscribe & notify related events (#2664)
  * Fixed missing macro when using pthread mutexes (fixes #2666)
  * Fixed warning
  * Remove duplicated flag for fuzzing coverage.
  * janus-pp-rec: support HEVC AP(aggregation packet) (#2662)
  * Fixed out of bounds array access
  * feat: support for SUBSCRIBE expiry (Expires header) in sip plugin (#2661)
  * Fixed types
  * RabbitMQ Transport Reconnect Logic (#2651)
  * Add per-participant recording options in AudioBridge to join API as well
  * Small simulcast-related demo tweaks
  * Make sure the source exists, before scheduling nice_agent_close_async (fixes #2655)
  * janus.js - renegotiate with external stream (#2604)
  * createOffer crashes in Firefox (on Mac) when re-negotiating p2p (eg. mute audio) (#2656)
  * Fixed broken g_strsplit limit in VideoRoom when parsing supported codecs (fixes #2657)
  * Fixed missing properties in permanent AudioBridge config saves
  * Add support for plain RTP participants in AudioBridge (#2464)
  * Bugfix: make sure that every destroyed plugin handle is 'detached' (#2652)
  * Bumbed to version 0.11.3

-------------------------------------------------------------------
Sun Jun 26 10:55:10 UTC 2022 - ecsos@opensuse.org

- Update to version 0.11.2:
  * Updated Changelog (0.11.2)
  * Added reference to AudioBridge announcements before using them in the mix
  * Added support for datachannel label/protocol to Lua and Duktape plugins (#2641)
  * Remove support for framemarking RTP extension (#2640)
  * Prevent race conditions on socket close in SIP and NoSIP plugins (#2599)
  * Fixed overflow runtime error
  * Fix for race condition between VideoRoom publisher leaving and subscriber hanging up (fixes #2582) (#2637)
  * Send PLI when starting a paused stream (#2645)
  * Prevent too high shift exponent
  * Fixed type of seq/ts in file-based Streaming mountpoint threads
  * Fix missing g_thread_unref when a streaming helper thread quits.
  * Added custom headers for SIP INFO request (#2644)
  * Free participant->user_id_str in case of opus enc/decoder error.
  * Reject flexfec when offered, as still unsupported (see #2639)
  * Don't add rtx ssrc if m-line is recvonly/inactive (see #2639)
  * Added NULL checks for json_dumps (see #2629)
  * Parse custom headers, if required, in successful REGISTER response (fixes #2636)
  * Timestamp correction for janus-pp-rec (#2573)
  * Don't chain error handler to success handler in Janus.httpAPICall (#2569)
  * Add missing library link for WS event handler (fixes #2628)
  * Fixed broken switch in Streaming plugin when using helper threads
  * Resolves meetecho/janus-gateway#2624 jansson double referencing (#2634)
  * Unlock mountpoints mutex after the spawning of helper threads.
  * Don't fail on duplicate b= lines in the SDP (see #2558)
  * Added ability to use websockets over unix sockets (#2620)
  * Fix typo in getJanusToken function (#2631)
  * Reference new mountpoint when switching, and check it has been destroyed
  * Fix streaming plugin RTSP sample in config template (#2627)
  * Removed unneeded brackets
  * fix moutpoint_mutex deadlock when rtsp reconnect fail (#2542)
  * Add info on moderation to list of publishers, if enabled
  * mqttevh: added ability use relative to config paths in config (#2623)
  * SIP plugin: SIP MESSAGE out of dialog (#2616)
  * Added missing mutex initialization (fixes #2622)
  * Bumbed to version 0.11.2

-------------------------------------------------------------------
Sun Jun 26 10:54:37 UTC 2022 - ecsos@opensuse.org

- Update to version 0.11.1:
  * Fixed typo in CHANGELOG
  * Updated Changelog (0.11.0)
  * Added new --log-stdout flag that enabled stdout logging even when daemonizing the process (#2591)
  * Added extra reference to VideoRoom
  * Add substream to audio/video receiving events (fixes #2615)
  * Fixed typo
  * Added new video (SIP/Janus workshop) to the documentation
  * Added session timeout value to Admin API info
  * Initialize simulcast RTCP contexts even if SSRCs are missing (fixes #2610)
  * audiobridge: GList leak fixed (#2611)
  * Fixed sending responses from Janus for incoming SIP MESSAGE/SIP INFO (#2609)
  * Added wss and debugging support to WebSocket event handler
  * Fixed broken path parsing in WebSocket event handler (fixes #2603)
  * fix memory leak in turnrest (#2606)
  * Making the timeout parameters for streaming plugin for RTSP play out configurable (#2598)
  * FreeBSD support (#2508)
  * Added support for admin-protected custom session timeouts (#2577)
  * Document 'timeout' and 'detached' events (fixes #2576)
  * Added more videos to the list of presentations in the FAQ
  * Fix warning about comparison of integers of different signedness in rtcp.
  * Added check on participant destroyed flag before sending events
  * Fixed typos in Changelog
  * Add some checks to publisher destroyed flag.
  * errors: replace strerror with locale-safe and threadsafe g_strerror (#2565)
  * Fix auth when both (token, secret) modes are enabled (#2581)
  * Changing default ICE nomination mode to 'aggressive' (see #2541)
  * fix rstp instead of rtsp typo (#2590)
  * Fix broken calculation of out-link-quality when NACKS exceed number of sent packets (fixes #2579)
  * Fixed memory leak
  * Make sure the publisher hasn't been destroyed, before trying to relay RTCP
  * Added Content type to SIP message (#2567)
  * clang/ubsan fixes (#2556)
  * add call_id in received sip message (#2563)
  * Fixed missing mutexes around VideoRoom ACL management
  * ice: fix conncheck typo (#2560)
  * feat: add "call_id" to "calling", "declining", "updatingcall" & "incomingcall" events (#2557)
  * Video moderation always returns unmuted (#2559)
  * Fixed typo in keepalive-conncheck usage
  * Add reference to publisher when using RTCP in forwarder
  * Set specific versions for Python 3 and meson in janus-ci yml.
  * Added audiocodec/videocodec supporto to 'joinandconfigure' in VideoRoom API
  * Add new option to configure ICE nomination mode, if libnice is recent enough (#2541)
  * if inviting on destroy, send BYE instead of 480 response (#2554)
  * Fix typo in videoroom docs.
  * Fixed small leak in VideoRoom
  * Initialize packet.is_rtp to false.
  * Add resolution and bitrate to Record&Play playback
  * Update janus.d.ts (#2553)
  * Allow up to 5 (rather than 3) audio/video codecs in the same VideoRoom
  * Allow forcing audio/video codec for VideoRoom publishers via query string
  * Initialize VideoRoom participant recording state when room recording is active (fixes #2550)
  * Fixed broken AV1 post-processing
  * Renamed extern janus_callbacks variables in Lua and Duktape plugins (#2540)
  * Bumped to version 0.11.1

-------------------------------------------------------------------
Sun Jun 26 10:22:20 UTC 2022 - ecsos@opensuse.org

- Update to version 0.10.10:
  * Updated Changelog (0.10.10)
  * Videoroom race condition fixes (see #2509) (#2539)
  * Fix parsing of SDP to find payload type matching profiles (fixes #2544) (#2549)
  * janus.js (#2548)
  * Make compiler fail if implicit-function-declaration is encountered.
  * Fixed non-portable call to strlcpy, and comment styles, in RabbitMQ code (see #2430)
  * Fixed VideoRoom docs on ICE Restarts for subscribers (fixes #2537)
  * Allow marking of RTP extensions in MJR recordings (#2527)
  * Moderator based muting/unmuting of VideoRoom streams (#2513)
  * Reject a=extmap-allow-mixed in SDP, when offered
  * Fix code style comments, also enable routing for direct exchanges
  * Configurable media direction when putting calls on-hold (SIP plugin) (#2525)
  * Added starting DTLS MTU to info returned by Janus API
  * Report fail if binding to a socket fails in websockets (#2534)
  * fix race condition in audiobridge plugin changeroom request (#2535)
  * Janus npm types upgrade (#2528)
  * set webrtc-adapter verstion to 7.4.0 (#2531)
  * Reduced verbosity of a few LOG_WARN messages at startup
  * Feature/enhance typings (#2518)
  * Fixed secret authentication on GET requests (#2524)
  * Dont send bye on early dialog (#2521)
  * Update Webpack instruction after webrtc-adapter dependency update (#2519)
  * Close nice agent resources asynchronously (#2492)
  * mqttevh: tls support implementation finished (#2517)
  * Fixed broken webrtc-adapter links (see #2515)
  * html: update webrtc-adapter to 7.7.0 (#2515)
  * Updated year in demos and docs
  * Fixed crash in WS event handler when backend is unreachable
  * Bumped to version 0.10.10
  * Adds back in default outgoing queue behaviour. Adds support for auto-generated queue_names
  * Adds RabbitMQ options for queues, durable, exclusive and autodelete
  * Check RabbitMQ admin topic in a better way
  * Increase RabbitMQ logging on publish
  * Fix queue_name_admin in rabbitmq transport
  * Update rabbitmq logging information
  * Updates RabbitMQ logic

-------------------------------------------------------------------
Sun Jun 26 10:21:25 UTC 2022 - ecsos@opensuse.org

- Update to version 0.10.9:
  * Updated Changelog (0.10.9)
  * Fixed memory leak when using announcements in AudioBridge (see #2504)
  * devicetest: unused var removed (#2502)
  * Increase participant's ref while handling kick requeest.
  * Fixed typo (see #2501)
  * Fixed a few compile and runtime issues in WebSocket event handler
  * Fix RTP headers when leaving/joining AudioBridge rooms on same PeerConnection
  * Fix occasional missing "left" event in audiobridge (see #2499).
  * Added note on Chrome bugs that prevent multiopus demo from working
  * Fixed occasional SRTP errors when pausing and then resuming Streaming plugin handles after a long time
  * Replace Travis CI with GitHub Actions. (#2486)
  * [janus-pp-rec] Fix potential crash when using skew compensation.
  * Fix regression on RTCP for sendonly video connections (fixes #2496).
  * videoroom: log invalid request name (#2495)
  * Set libmicrohttpd connections limit in http transport configuration. (#2489)
  * Fixed create/attach management with null optional args (fixes #2490)
  * Improve detection of maximum resolution in mjr file (postprocessor) (#2487)
  * Added support for binary data recordings (#2481)
  * Fixed inconsistency of mySdp in janus.js (fixes #2379)
  * Updated instrunctions to build libwebsockets (see #2476)
  * Reset simulcast context for a videoroom publisher also when renegotiating (fixes #2466).
  * Skip SDP munging in janus.js if SIM attribute is already present in the offer (see #2466).
  * Do not require a cacertfile, pass null to openssl (#2485)
  * Increased size of buffer used to render new prflx candidates (fixes #2480)
  * Replay data channel recordings (#2468)
  * Fixed warning
  * [janus-pp-rec] Drop audio RTP silence suppression packets. (#2467)
  * Corrected janus pp rec (#2472)
  * Add an option to enable libmicrohttpd error logs (#2471)
  * Fix types (#2475)
  * Check simulcast layer index also when stream is video only.
  * Fixed incomplete recordings after SSRC change (e.g. hold/unhold) in SIP and NoSIP plugins
  * Make TURN REST API timeout configurable in janus.jcfg (#2470)
  * Added custom headers to 'decline' request (#2465)
  * Adding janus-angular to the Resources page of the documentation (#2459)
  * Fix autoplay policy issues on Safari (see #2455).
  * Added user gesture for Safari in screensharing demo (see #2455)
  * [janus-pp-rec] Enhance timestamp assignment and add payload-type option to CLI. (#2345)
  * Add Custom Headers Hold Event (#2454)
  * fix: disable auto ack when answering incoming call (#2447)
  * Fix incompatible-pointer-types compiler warnings (#2444)
  * fixing a couple of bugs in burst transfers (#2427)
  * Bumped to version 0.10.9

-------------------------------------------------------------------
Sun Jun 26 10:20:58 UTC 2022 - ecsos@opensuse.org

- Update to version 0.10.8:
  * Updated Changelog (0.10.8)
  * fix: disable responses to NOTIFY requests in janus_sip plugin (response is already handled in sofia-sip) (#2441)
  * Add LIBSRTP_CFLAGS to compiler flags of plugins that require srtp headers (#2442)
  * [janus-pp-rec] Do not overwrite original RTP header data when attempting audio skew compensation.
  * [janus-pp-rec] Use 64-bit timestamps for audio skew compensation.
  * Let SIP users cancel pending transactions without waiting for a provisional response. (#2434)
  * janus_streaming: fix warnings if missing libogg (#2438)
  * Added new video to FAQ
  * Warn if sofia-sip logs are redirected from stdout.
  * Small tweaks to AudioBridge prebuffering

-------------------------------------------------------------------
Sun Nov 15 12:07:43 UTC 2020 - ecsos@opensuse.org

- Update to version 0.10.7:
  * Updated Changelog (0.10.7)
  * More aggressive PLI at startup when using simulcast in VideoRoom plugin
  * Configured janus-pp-rec to skip packets with unknown payload types when static payload types are expected (G.711, G.722)
  * Fixed missing initialization of AVPacket that could cause crashes when postprocessing G.722 recordings
  * Modified demos to remove hardcoded 320x240 video element slot
  * Fix RTP header buffer read (#2411)
  * Add missing unref to mDNS resolver gobject. (#2399)
  * Use PKG_CONFIG_PATH as configured for nice version (#2405)
  * Replace rand() with janus_random_uint32() (fixes #2404)
  * Fixed occasional memory leak at shutdown when frequently using timed callbacks in Lua/Duktape plugins
  * Updated insertable streams code in janus.js (and e2ee demo)
  * janus.d.ts: correct mediaState definition (#2396)
  * Minor typo fix (#2393)
  * Fixed broken rid-based simulcast for substreams<3
  * Fixed typo in AudioBridge docs (see #2391)
  * janus.js: allow configuring simulcast send encoding parameters (#2392)
  * Fixed broken indentation
  * Refresh lws 4.x connection validity for any ws incoming message.
  * fix warning about being deployed on private IP (#2386)
  * Implement mutex on rabbitmq-event to control connection (#2380)
  * Do not handle session stack mutex if helper has not been created. (#2387)
  * Fix SDP negotiation when client uses max-bundles (fixes #2390)
  * Keep extra whitespace in legacy simulcast rid SDP line
  * Removed extra whitespace in simulcast recv SDP line
  * Add JSEP flag to invert processing order of rid in SDP (#2385)
  * Fixed compilation error when using libwebsockets < 3
  * Allow AudioBridge to originate SDP offers (#2366)
  * Bumped to version 0.10.7
- Drop libwebsockets-3.patch, because now in upstream.

-------------------------------------------------------------------
Sun Nov 15 12:03:53 UTC 2020 - ecsos@opensuse.org

- Update to version 0.10.6:
  * New mechanism to tweak/query transport plugins via Admin API [PR-2354]
  * Fixed occasional segfault when using event handlers and VideoRoom [Issue-2352]
  * Fixed occasional "Unsupported codec 'none'" log errors (thanks @neilyoung!) [PR-2357]
  * Fixed broken AudioBridge RTP forwarding when using G711 [Issue-2375]
  * Added helper threads support to RTSP mountpoints as well [PR-2361]
  * Fixed data channels not working as expected in Streaming plugin when using helper threads
  * Fixed simulcast occasionally not working in Streaming plugin until manual PLI trigger
  * Added proper fragmentation in WebSockets transport plugin [PR-2355]
  * Fixed timing resolution issue in MQTT transport (thanks @feymartynov!)) [PR-2358]
  * Fixed MQTT transport issue when trying to shutdown gracefully (thanks @feymartynov!)) [PR-2374]
  * Fixed broken configuration of Nanomsg Admin API (thanks @sdamodharan!)) [PR-2372]
  * Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)

-------------------------------------------------------------------
Wed Sep 23 15:07:05 UTC 2020 - ecsos@opensuse.org

- Update to version 0.10.5:
  * Bugfix: prevent borked generated audio file if meetecho header is present with no RTP data next (#2356)
  * Don't print SDP errors if rtx is being negotiated for audio
  * Fixed deprecated lws semantics in WS event handler too
  * Remove deprecated libwebsockets semantics in WS transport (see #2349)
  * Added presentation on Insertable Streams to docs
  * Add missing documentation for janus-pp-rec.
  * Various minor typo fixes (#2313)
  * Fixed typo (see #2341)
  * Easy support for one-to-many scenarios in videoroomtest (#2341)
  * Kick (#2332)
  * Clear publisher codecs in videoroom hangup media.
  * Send a PLI (if supported) to the Streaming mountpoint source when switching (fixes #2333)
  * Added missing token in handle-related event (fixes #2312)
  * Fixed documentation in plugin videoroom (close #2301).
  * Remove unneeded mutex unlock that was causing a crash in the videoroom plugin (fixes #2318).
  * Bugfix: make audio/video recording in videocall working again
  * Bumped to version 0.10.5
  * Updated Changelog (0.10.4)
  * Added Janus workshop made at ClueCon 2020 to list of videos in the docs
  * Fixed definition of variable in for loop
  * Use unique IDs and internal hashtable to map SCTP associations with usrsctp (#2302)
  * Only use CURLOPT_HTTP09_ALLOWED if libcurl is >= 7.66.0 (fixes #2307)
  * Fix occasional curl hiccups with RTSP with some cameras
  * Fixed typo
  * Have VideoCall sessions reference each other, when in a call (see #2300)
  * Add more checks on peer when hanging up VideoCall session
  * Fix minor memory leak in participant inbuf of audiobridge plugin (#2298)
  * Allow specifying multiple IP addresses for 1-1 NAT. (#2279)
  * Fix candidates memory leaks (#2288)
  * Pass MQTT buffer settings to Paho (#2286)
  * Check websocket readystate on destroy (#2276)
  * Increase reference before sending data via SCTP (fixes #2271)
  * Add MQTT v5 properties support (#2273)
  * Fixed crash in VideoRoom plugin when failing to setup subscriber (fixes #2277)
  * Fix broken EchoTest demo for Firefox if datachannels are not supported (#2281)
  * RabbitMQ-Event - Add heartbeat option and create logic to reconnect to rabbitmq (#2267)
  * Added CommCon 2020 talk to the videos in the docs
  * Fix a deadlock in audiobridge changeroom action on "User ID already taken" error (#2280)
  * Improve building with BoringSSL (#2278)
  * Bumped to version 0.10.4
  * Updated Changelog (0.10.3)
  * Added support for 'info' request to janus.js
  * add bitrate_cap to documentation (#2266)
  * add default values to videoroom documentation (#2265)
  * Updated issue template
  * Updated issue template
  * Made documentation of RTP forwarding + simulcast in VideoRoom clearer
  * Set app_handle ptr to NULL when freeing a plugin session.
  * Early add a reference to a subscriber in videoroom handler. (#2253)
  * New demo to use canvas element with EchoTest plugin (#2261)
  * Added missing SRTP support to AudioBridge RTP forwarders (see #2258)
  * Fixed typo (SSRC outbound for RTP forwarders)
  * Fixed typo preventing SRTP support in static AudioBridge RTP forwarders (fixes #2258)
  * fix documentation for mute_room/unmute_room (#2257)
  * Fix opus silence potential to generate huge files (#2250)
  * Added timeout to connections in HTTP transport (120s)
  * Add more checks on validity of NUA before using it in SIP plugin (#2247)
  * Set last sending timestamp for the first packet sent (avoid #2217 overflow issue).
  * Fix occasional recording issues in Lua and Duktape plugins
  * Added NULL check before strstr in Lua and Duktape plugins
  * fix redundant condition (#2240)
  * Update Webpack exports-loader module config example (#2235)
  * plugins/janus_audiobridge.c: fix build without libogg (#2238)
  * Increase travis git depth to 10.
  * Move Travis badge url from .org to .com
  * Refactor videoroom hangup media internal. (#2236)
  * Bumped to version 0.10.3
  * Updated Changelog (0.10.2)
  * There are many places where callbacks.error return `string` and many where return `Error`. And this is the only place, where callbacks.error return `string, Error`. That's why it needed to write redundancy conditions for right handling errors from createOffer(). But we can return only `Error` object like for getUserMedia() to avoid this (#2230)
  * RTCRtpSender unavailable on old browsers (#2206)
  * Check extensions after renegotiations (see #2223, fixes #2192)
  * Fixed typo (Duktape plugin always relaying binary data, even for text)
  * Update Duktape to v2.5.0 (#2233)
  * Removed unused variable
  * Fixed broken simulcast behaviour (#2231)
  * Change session 'started' property in VideoRoom to atomic
  * Fix sscanf-related security issues (#2229)
  * Allow simulcast ports to be picked randomly in Streaming mountpoints (#2225)
  * Removed extra space in janus.js
  * stop all tracks when streamsDone fails (#2134)
  * Increase reference to session when handling SIP calls (see #2188) (#2216)
  * (fixed) Check destroyed flag when handling a subscriber participant.
  * Revert "Check destroyed flag when handling a subscriber participant."
  * Check destroyed flag when handling a subscriber participant.
  * Take simulcast/svc into account for switch request (fixes #2219)
  * Bumped to version 0.10.2
  * Updated Changelog (0.10.1)
  * Fixed typo
  * Send username when using TURN REST API (fixes #2199) (#2201)
  * H264 profile fix (#2212)
  * Fixed silly typo
  * Update janus.d.ts (#2215)
  * Allow empty metadata strings to be passed in Streaming edit (fixes #2208)
  * Added check on libavcodec version for AV1 postprocessing
  * Security fixes in SDP code (#2214)
  * Fix if not building from the top directory (example : yocto) (#2187)
  * Update vp9svctest.js (#2213)
  * add enabled field to stream list (#2210)
  * Update janus.d.ts (#2202)
  * add metadata field to list reposnse, as documented (#2205)
  * fix muted timeout race condition (#2203)
  * Set subscriber's session type before unlocking sessions mutex.
  * Fixed broken link to libnice project in docs (see #2198)
  * Update libnice link (#2198)
  * NoSIP plugin: Fixed SRTP-SDES for "process" request and session update (#2196)
  * Don't keep session in paused when switching mountpoints (#2197)
  * make libcurl follow RTSP 302 redirections (#2195)
  * Fix RTSP parsing (#2190)
  * docs fix (#2194)
  * Don't put session to stopping in watch (#2189)
  * Initial support for end-to-end encryption via Insertable Streams (#2074)
  * Bumped to version 0.10.1
  * Allow negotiation of AV1 and H265 (#2120)
  * Updated Changelog (0.10.0)
  * Add some missing videoroom unref in case of errors. (#2186)
  * Small fixes after static analysis
  * Small fixes after static analysis
  * Removed unneeded check
  * Protect session callee accesses through session mutex in SIP plugin (#2184)
  * Fixed srtp on update request (#2173)
  * Add experimental feature videobufferkf support to RTSP mountpoints (#2180)
  * Added dereference check (fixes #2178)
  * for loop compilation fail (#2183)
  * janus_videoroom: fix bad copy paste of codec name (#2176)
  * Fix streaming plugin demo page when string_ids is true (#2175)
  * Small improvements in the documentation of videoroom (#2171)
  * Fixed many Doxygen warnings
  * Notify speaker about talk events in AudioBridge too (see #2172)
  * Moved comment on talking events
  * Update to notify speaking participant (#2172)
  * Updated resources page in the docs
  * Fixed experimental feature videobufferkf (#2170)
  * Small tweaks to GELF event handler (see #1788)
  * Compile the GELF handler unless it's disabled (no deps)
  * Gelf event handler (#1788)
  * Removed extra empty lines
  * SIP plugin: add audio/video stream with an update request(or reinvite) (#2164)
  * Update deps for web demos
  * Add a secret to all sample mountpoints, in the configuration file
  * Ensure an address family is assigned by the streaming plugin. (#2167)
  * Fixed checks on result of new thread
  * Added missing g_error_free calls when threads can't be created
  * Streamlined code of the demos
  * Some small tweaks to the README
  * Some small tweaks to the README
  * Added note on those creepy .exe builds that apparently are still around
  * Updated README
  * Fixed compilation error with libwebsockets 4 (see #2162)
  * Added support (untested) for libwebsockets 4.x's ping/pong mechanism (see #2162)
  * Updated README to suggest libwebsockets 3.2-stable, for now
  * Disable (for now) ping/pong mechanism if libwebsockets >= 4.x (see #2162)
  * Fixed typo in echotest.lua
  * Update README.md with new libnice instructions.
  * Travis meson libnice (#2163)
  * Added changes from #2161 to AudioBridge, Streaming and TextRoom plugins too
  * List private rooms if valid admin_key was provided. (#2161)
  * Fixed several code style issues (and incorrect log levels) introduced in #2158
  * User talking (#2158)
  * Apply again the changes in 43ddcd2870012e382fecaaf123457000e4d74901.
  * Add missing unref in videoroom.
  * Added support for data channel subprotocol (#2157)
  * Updated README
  * Fix menus in html documentation when using Doxygen > 1.8.14 (#2155)
  * Send a PLI for new viewers, if the Streaming mountpoint has RTCP (fixes #2156)
  * Started adding links to issues/PRs to changelog (0.9.5 only, for now)
  * Add a reference to the subscriber while joining.
  * New plugin callback to know when datachannel is writable (#2060)
  * Add support for VP9 and H.264 profile negotiation (#2080)
  * Bumped to version 0.10.0
  * Updated Changelog (0.9.5)
  * Added VideoRoom option to only allow admins to change the recording state (see #2137)
  * Enable / disable recording while conference is in progress (#2137)
  * Added logging of errno when getifaddrs fails
  * Added token to 'attached' event (handlers) and to Admin API (handle_info)
  * Don't join mixer thread when destroying AudioBridge room
  * Added support for RTP extensions to NoSIP plugin (fixes #2152)
  * Fixed code style
  * Added option to keep candidates with private hosts when using nat-1-1, and advertize them too instead of just replacing them
  * Only process mute events if a timer fired to avoid video flashing. (#2147)
  * Added DSCP support for RTP to NoSIP plugin too (see #2150)
  * Add DSCP on RTP audio packets in SIP plugin (#2150)
  * Added support for multichannel Opus audio (surround) (#2059)
  * Fixed typo in new publication
  * Add a reference to citeus.html
  * Execute `janus_check_sessions` if at least one of (`session_timeout`, `reclaim_session_timeout`) is set (#2143)
  * small HTML fixes (#2136)
  * Reduced verbosity of some AudioBridge messages
  * Fixed typo in VideoRoom error response
  * Fixed typo in AudioBridge error response
  * Fixed typo
  * Added new tool to convert .pcap captures to .mjr recording (#2144)
  * Fix to rare deadlock in Streaming plugin (see #2115) (#2141)
  * Adding support for cipher suite selection in websockets transport (#2135)
  * Added request to globally mute/unmute an AudioBridge room
  * Fixed AudioBridge announcement not waking up sleeping forwarder
  * Remove extra unref when destroying NACK cleanup timeout source.
  * Added API to check if a specific file is playing in the AudioBridge
  * Fix post-processor RTP extensions parsing.
  * Bumped to version 0.9.5
  * Updated Changelog (0.9.4)
  * Fixed duplicate subscriptions in Streaming plugin (fixes #2129)
  * Updated info in Streaming plugin to return count of viewers (if secret is provided)
  * Fix websocket transport disconnected occasionally #2081 (#2107)
  * Fixed incorrect DSCP value being set (see #2055)
  * + Start message processing after requesting candidate gathering (#2121)
  * Make sure the ICE agent still exists, when we try to gather candidates
  * Update transports docs by removing an old sentence about WebSockets not being stable
  * Align Admin API unsupported method error to Janus API
  * New session mutex in Streaming plugin (see #2106) (#2115)
  * Fixed a couple of typos and compilation warnings
  * Don't respond to HTTP requests when still parsing headers (fixes #2118)
  * Fixed .opus file last chunk playback (#2114)
  * Stop using legacy datachannel negotiation in Streaming and TextRoom (fixes 2112)
  * Use a mutex around janus_videoroom_hangup_subscriber and subscriber list. (#2102)
  * rabbitmq exchange type as config value (#2104)
  * Add missing decref in janus_http_timeout. Replace free with g_free in janus_http_return_success.
  * Notify AudioBridge playback start/stop via event handlers
  * Clarified in docs that HMAC-Signed tokens are only supported by VideoRoom
  * Bugfix/cpu usage based on v0.8.2 (#2101)
  * Add some missing static declarations to HTTP and WS transports.
  * Don't wait forever for candidates when half-trickling
  * Updated AudioBridge documentation with new playback feature
  * Added new docker image to the resources in the docs
  * More checks when hanging up VideoRoom subscriber (see #2087) (#2093)
  * Fixed returned address when adding multicast Streaming mountpoints
  * Bumped to version 0.9.4
  * Updated Changelog (0.9.3)
  * Add support for playback of audio files in AudioBridge (#2088)
  * Swap RR/SR Report Blocks if the first block contains rtx data. (#2089)
  * Return mountpoint IP addresses, if a bind interface/IP was provided
  * Added project to resources in the docs
  * Fix libasan use after free in janus_videoroom_handler when events are enabled (#2091)
  * Fix copy-paste error in Streaming plugin docs
  * Fixed a few typos in AudioBridge errors
  * Fixed AudioBridge create API not working properly when using string IDs
  * Define the libnice version string as extern in version.h (fixes gcc10 error)
  * Use custom GSource to handle HTTP request timeouts (see #2062 and #2066) (#2075)
  * Add missing info to videoroom "list" response (#2068)
  * Made libnice warning clearer, and upped suggested version (fixes #2069)
  * Don't show warnings for rtx RTCP packets
  * Reverted isTrickleEnabled check in janus.js (fixes #2064)
  * Added option to configure time needed to detect a missing simulcast substream (#2063)
  * Reference subscriber when handling related messages (see #2045) (#2061)
  * refactoring-clean up (const-var, semicolons, ===, etc.) (#2044)
  * Support for additional constraints on screenshare media (#2043)
  * Fixed syntax error in sample Streaming plugin configuration file
  * Fixed outdated info in VideoRoom docs
  * Fixed typo
  * Added option to disable building AES-GCM support (see #2024 and #2054)
  * Use refcount for Streaming plugin helper threads (#2039)
  * Fixed Streaming destroy not working when using strings
  * Always add remote candidates from the libnice loop (see #2045) (#2048)
  * Add configurable DSCP ToS for PeerConnections (#2055)
  * Added notes on building libsrtp (see #2024)
  * Fixed printout of metadata in Streaming demo
  * Added support for generic metadata to Streaming mountpoints
  * Added support for static Opus files to Streaming plugin (#2040)
  * Detect libsrtp(2) using pkg-config (fixes #2019) (#2033)
  * Don't set ICE credentials when parsing remote credentials (#2046)
  * plugins: drop tautology (#2041)
  * Fixed av_register_all deprecation check in post-processor
  * Fixed VideoRoom destroy not working when using strings
  * Fixed janus-pp-rec build warnings when using ffmpeg >= 4.x
  * janus_http: return earlier if request is NULL (#2031)
  * Bumped to version 0.9.3 (again)
  * Updated changelog for 0.9.2
  * Bumping back to 0.9.2 to re-tag
  * Fixed missing refcount init for Admin API (fixes #2029)
  * test_aiortc: cleanup (#2027)
  *  Add Python aiortc-based functional testing. (#1971)
  * Bumped to version 0.9.3
  * Updated Changelog (0.9.2)
  * Updates to mutex unlocking in textroom and videoroom plugins (#2026)
  * Reference count janus_request instances (#2020)
  * Resolve mDNS candidates asynchronously with GResolver (see #1998) (#2004)
  * Reverted change on janus.js (see #2018)
  * Fixed typo in janus.js error code (fixes #2018
  * Track pending nack cleanup tasks and cancel them when freeing a stream. (#2014)
  * Prepare RTCP Sender Reports by considering the last RTP timestamp sent. (#2007)
  * Update media direction in SIP plugin if remote address is 0.0.0.0 ('hold' fix) (#2013)
  * http_transport: add NULL checks (#2012)
  * Use user_id_str for kicked, leaving, and unpublished events, if enabled. (#2010)
  * Add repos for openSUSE and SUSE (#2009)
  * Added called URI to 'incomingcall' and 'missed_call' events in SIP plugin
  * Fixed small leak in SIP plugin when holding calls
  * Added link to FOSDEM 2020 talk on RTP forwarders to the docs
  * Support for RTSP 'Content-Base' header in Streaming plugin (#1999)
  * Fixed deadlock when using claim on HTTP transport (fixes #2000)
  * Added option to ignore mDNS candidates (#1998)
  * Fixed typo when renegotiating audio in janus.js (fixes #2002)
  * Added option to enforce validation on DTLS certificates (#1992)
  * Fix occasional deadlock in VideoRoom (2) (credits to @mivuDing, fixes #1982) (#1984)
  * Fix rare race condition when claiming sessions (#1990)
  * Small tweaks to #1997 (renamed, moved and documented RSA property in janus.jcfg)
  * Implement ECDSA Certificate generation (#1997)
  * update dtls ciphers (#1995)
  * Several fixes to session management in VideoCall plugin (#1994)
  * Fixes to leaks and race conditions in VoiceMail plugin (#1993)
  * Make sure the session still has a reference when cleaning up HTTP requests
  * Fixed double unlock when listing private rooms in AudioBridge (#1988)
  * Fixed typo in querylogger_parameters (copy/paste error) (#1989)
  * ice: ensure that stream is non-NULL (#1987)
  * Small fixes for TypeScript declaration file (#1986)
  * Added -f to rm in html Makefile.am (fixes #1985)
  * Converted HTTP transport plugin to single thread (#1173)
  * Added maximum value for AudioBridge prebuffering property
  * Add G.711 support to the AudioBridge plugin (#1979)
  * Make prebuffering in AudioBridge configurable (#1975)
  * Bumped to version 0.9.2
  * Updated Changelog (0.9.1)
  * Fixed typo in SIP demo code
  * Fixed abort at server shutdown after using SIP transfers
  * Several enhancements to SIP demo
  * Added more checks on nice_address_set_from_string (fixes #1973) (#1981)
  * Reply to incoming REFER with 202 right away, not 100, in SIP plugin
  * Fixed occasional missing referred-by info in SIP demo
  * Add UI to SIP demo to remove helpers, when created
  * Fixed broken DTMF in SIP demo
  * Removed wrong comment
  * Always use base SSRC when recording VideoRoom simulcast participant
  * Reduced log level to info when logger and event handlers are not found (#1980)
  * Fixed leak when creating Streaming mountpoint dynamically
  * Hide libcurl from pkg-config when testing travis-ci with LIBCURL = NO.
  * Valgrind fixes for sockaddr structs (#1976)
  * Remove /root from the list of protected folders. Make comment text more clear.
  * Fixed broken method signature in Streaming plugin when not using libcurl
  * Added checks on nice_address_set_from_string (fixes #1973)
  * fix #1967 (#1968)
  * Support for strings as unique mountpoint IDs in Streaming plugin (#1969)
  * Fixed typos in TextRoom
  * Added errno info when socket operations fail in Streaming plugin
  * Make sure a publisher exists when asking for a VideoRoom subscriber renegotiation (fixes #1970)
  * Fixed a couple of JSON attributes in VideoRoom when strings are used (see #1880)
  * Remove duplicated codecs when answering SIP call (#1966)
  * Fixed errors creating VideoRoom when strings are used (see #1880)
  * If glib is too old, generate uuid manually when needed (see #1880)
  * Support for strings as unique IDs in AudioBridge, VideoRoom, TextRoom (#1880)
  * Detect H264 key frames with smaller SPS units (#1965)
  * Small tweaks to demo intro text
  * Added license badge to the README
  * Removed unused variables
  * Added link to new event handlers documentation to the doc main page
  * Subtype for some event, and better docs for event handlers (fixes #1953) (#1957)
  * rtp: drop dead code in rtp_header_update callers (#1964)
  * Remove Sofia reference from the title of the SIP demo
  * janus_sip: add missing check for NULL (#1963)
  * add missing callbacks.error check (#1959)
  * Configurable global prefix for log lines (#1940)
  * Bumped to version 0.9.1
  * Updated Changelog (0.9.0)
  * conf: transports: document events option (#1952)
  * We should allow to have ICE-TCP enabled without ICE Lite. Recent versions of libnice allow this combination and gather tcp passive candidates etc. in this setup. (#1946)
  * Avoid RTP header memory misalignment in rtx packets (#1943)
  * Renamed corpora file
  * Optimized parsing of TWCC RTCP message (Credit to OSS-Fuzz)
  * Update debugging section in Janus documentation.
  * Fixed occasional error messages on console when trying to add RTP extensions
  * Travis libnice clang flags (#1941)
  * Update janus_audiobridge.c (#1938)
  * Fixed regression on video bitrates when using monodirectional PeerConnections
  * Add OSS-Fuzz badge.
  * Fixed occasional segfault when parsing TWCC RTCP message (Credit to OSS-Fuzz)
  * Add travis_retry to git clone commands.
  * Fixed leak when parsing broken TWCC RTCP message (Credit to OSS-Fuzz)
  * Fix volume-related functions in janus.js (#1935)
  * Fixed RTCP parsing issue found by OSS-fuzz
  * Fixed typo when adding audio attribute to SDP
  * Fixed broken RTP fuzzer
  * Dynamically update NACK queue size depending on RTT (#1867)
  * Support for transport-wide CC on outgoing streams (#1889)
  * Refactoring of core-plugin callbacks and RTP extensions termination (#1884)
  * Bumped to version 0.9.0
  * Updated Changelog (0.8.2)
  * Janus Travis CI integration (#1932)
  * Added Coverity badge
  * Small tweaks after static analysis
  * Fixed helpers not being able to send SUBSCRIBE requests in SIP plugin
  * Removed deprecated text from screensharing demo
  * Removed deprecated warning in screensharing demo
  * Fixed broken links in docs (plugins list)
  * typo (#1934)
  * Fix g_async_queue usage (#1929)
  * Remove odd respond to automatically responded OPTIONS request (#1930)
  * Updated man file for janus-pp-rec
  * Add audio skew compensation to janus-pp-rec. (#1870)
  * Add math library when fuzzing locally.
  * Add missing mutex unlocks in videoroom message handler.
  * Fixed undefined reference when building fuzzers
  * Better parsing of RTSP messages (see #1922) (#1925)
  * Fixed undefined reference when building postprocessor utilities
  * Add new configuration property to add protected folders not to save to (#1919)
  * Added missing check on SDP attribute value existence
  * Added check on AudioBridge instance in setup_media (fixes #1923)
  * Fixed reference to deprecated configuration file
  * More verbose output on postprocessing output error
  * Fix a possible race condition when joining as a subscriber and destroying the session. (#1911)
  * Bumped to version 0.8.2
  * Updated Changelog
  * Increase buffer when post-processing VP8/VP9 recordings too (see previous commit)
  * Fixed occasional buffer overflow error when post-processing H.264 recordings
  * Use sendBeacon instead of sync XHR in onbeforeunload (fixes #1902) (#1918)
  * Updated year in demos and docs
  * Don't keep TextRoom plugin loaded if data channels were not compiled
  * Fixed warnings when building DTLS bio code
  * Added reference to Snap repo in resources (docs)
  * Fixed late initialization of janus.js constructor callbacks (fixes #1912)
  * Move loggers cleanup to end of logger thread (fixes #1904)
  * fixed typo (#1916)
  * Only close the event handlers directory if it was opened (see #1903)
  * startup: only close the logger directory if it was opened (#1903)
  * Fix out of bounds array access for last_spatial_layer (#1906)
  * Fixed occasional memory leak in Streaming plugin (fixes #1900)
  * Fixed leak in SIP plugin (fixes #1897)
  * Fixed warnings introduced in #1896
  * he 'referred_by' field currently holds the SIP URI value copied from the (#1896)
  * Add in mountpoint/forwarder create response the allocated RTCP ports.
  * Check if rtcp port is > 0 before creating a RTCP socket, in Videoroom plugin.
  * Revert "Check if rtcp port is > 0 before creating a RTCP socket."
  * Check if rtcp port is > 0 before creating a RTCP socket.
  * Allow RTCP ports to be picked randomly using 0, in Streaming plugin
  * Fixed typo in SIP plugin
  * Binary data support in data channels (#1878)
  * Remove SIPre plugin from the repo (#1894)
  * Bumped to version 0.8.1
  * Updated changelog (v0.8.0)
  * Added fwrite checks in record.c (warnings only)
  * Fixed variable shadowing
  * Make sure the installed libcurl knows about CURL_AT_LEAST_VERSION
  * Fixed obsolete value for TWCC period default in docs/hints
  * Fixed small typos in demos
  * Make sure libcurl is available before using CURL_AT_LEAST_VERSION (fixes #1887)
  * [Suggestion] Started the refactoring of the janus.js (#1830)
  * Added changelog (and info on tagged versions) to documentation
  * Fix RTSP SETUP when url includes query string parameters (fixes #1869) (#1875)
  * Add CHANGELOG.md file into the project (#1885)
  * Added link to new video on Simulcast and SVC to docs
  * Fixed wrong default folder for loggers
  * Fixed exception to GPL code (see #713)
  * Avoid gzip functions when fuzzing in OSS and add zlib dependency when fuzzing locally.
  * Updated documentation to include some info on the new logger modules
  * Remove option to enable rtx (now always supported, when negotiated) (#1877)
  * Fixed linking error for post-rocessing tools after recent changes
  * New category of plugins for modular logging (#1814)
  * Gzip compression utility in the core (and sample event handler) (#1846)
  * Bumped to version 0.8.0
  * SIP plugin: custom (non-standard) headers on incoming events (requests) (#1873)
  * Reduced default twcc_period value from 1s to 200ms
  * Reduced verbosity of some lines in the SIP plugin
  * New functionality to add custom Contact URI params to SIP REGISTER (#1874)
  * Fixes to RTSP latching procedure (fixes #1536, replaces #1851) (#1866)
  * Bumped version in postprocessing tool as well
  * Don't send RTCP SR if outgoing media has been disabled via SDP update
  * Keep track of clock rates associated to payload types, for RTCP
  * Feature/ignore unreachable ice server (#1854)
  * Fixed wrong clock rate being used for RTP header updates when using G.722
  * Don't scan libnice version if it wasn't retrieved (fixes #1858)
  * add missing closing curly bracket (#1859)
  * Fixed rare race condition in HTTP plugin that could cause leak (fixes #1665)
  * Fix RTP fuzzing target according to recent VP9 changes.
  * Fixed regression when setting up DataChannels
  * Fixed broken code in AudioBridge
  * Use strtol more, and add checks when atoi is used (#1852)
  * Fixed SIP hangup not sending CANCEL, when inviting (fixes #1856)
  * VP9 SVC fixes (#1849)
  * fix nullptr dereference in streaming plugin (#1855)
  * Fixed typo
  * Add exception var to catch stmt to fix rollup (#1848)
  * Updated link to project in resources (docs)
  * Split lines on line feed only, and trim carriage feed instead
  * Skip multiple b= line break conditional for b=TIAS (#1832)
  * ice: ignore/enforce only when IP starts with the partial-string from the list (#1840)
  * IPv6 support in Streaming plugin (#1807)
  * Add support for domain names (and IPv6) to RTP forwarders (#1778)
  * Support for SIP transfers (#1815)
  * Support for simultaneous calls in SIP plugin (#1772)
  * Bumped to version 0.7.6
  * Fixed check
  * Update getStats() to use a promise instead of callback (#1823)
  * Improved attach/reattach MediaStream helpers avoiding browser versions (#1828)
  * Updated libnice recommended version into the README and mainpage.dox files (#1835)
  * Detect new streams also when mountpoint is disabled
  * Revert previous commit (causes crashes, to investigate)
  * Split lines on line feed, and trim carriage feed (see #1818)
  * Avoid locking mountpoints when reconnecting to RTSP servers. Use 5 seconds connection timeout in curl requests.
  * Fixed simulcast issue when automaticlly dropping to lower layers
  * Reduced verbosity of some RTCP related messages
  * Added Admin API command to inject strings in Janus logs from outside
  * Fixed broken check (again) on libwebsockets version (see #1812)
  * Fixed a few typos in the documentation
  * Fixed outdated text in documentation
  * Clear publisher's room pointer when leaving and add a reference while executing janus_videoroom_leave_or_unpublish. (#1795)
  * Make sure flags are cleared when getting a close_pc() even when a PeerConnection wasn't created (fixes #1800)
  * Fixed broken check on libwebsockets version (see #1812)
  * Fixed compilation error of WebSocket event handler with older version of libwebsockets (fixes #1812)
  * Added reference to JanusCon to the FAQ for learning material
  * --cwd-path (Current Working Directory) CLI option added (#1804)
  * Added new Nanomsg event handler (#1802)
  * Added new WebSockets event handler (#1799)
  * Hangup Custom Headers (#1809)
  * Wait for keyframe when dropping to lower simulcast layer because of inactivity (fixes #1806)
  * Reduced verbosity of successful mDNS resolves
  * Fix a missing pop on Duktape stack when invoking resumeScheduler.
  * Added warning if libnice version is outdated (at least 0.1.15 recommended)
  * Hook Lua print function(s) to Janus logger (#1782)
  * Write moov atom at the head of the MP4 file (#1791)
  * Support for SIP SUBSCRIBE/NOTIFY in SIP plugin (#1768)
  * sdp-utils: check that janus_sdp_get_codec_rtpmap succeeded (#1785)
  * Add some missing atomic checks in videoroom plugin.
  * Mute participants (#1787)
  * Fixed participant ID being reset in AudioBridge web demo
  * Allow for capturing desktop audio when sharing screen (#1771)
  * Duktape getVersion method added (#1786)
  * SIP plugin: add local interface for SDP binds (#1784)
  * Better async managament of new mountpoints with temp map of IDs (#1732)
  * Fixed potential endless loop in Streaming plugin when binding ports (fixes #1762, replaces #1763)
  * Add command line option to janus-pp-rec to specificy the output format (#1777)
  * Don't remove room for subscriber if not closing PeerConnection (fixes #1761)
  * JavaScript logging improved (#1781)
  * moved destroySession connection condition (#1783)
  * Ignore temporary SSL errors in RabbitMQ transport (see #1769)
  * Fixed typo in SIP demo (DTMF digits message)
  * Typo fix in JANUSSDP.removePayloadType (#1779)
  * Updated version in bower package.json too
  * Fixed broken negotiation in SIP plugin for mandatory SDES-SRTP (fixes #1770)
  * Added option to specify local port when testing STUN server via Admin API
  * Fixed broken responses to incoming SIP INFO and MESSAGE requests
  * Add audio level dBov average to talk events in VideoRoom plugin (#1751)
  * Fixed outdated reference to old configuration files in demos
  * Fixed broken indentation
  * Bumped to version 0.7.5
  * Small tweak to verbose output (see #1740)
  * Fixed handling of offerless reINVITE. (#1740)
  * Hopefully final fix for RTCRtpTransceiver check (see #1759)
  * Improved RTCRtpTransceiver check (see #1759)
  * Fixed RTCRtpTransceiver check for Edge (fixes #1759)
  * Fixed wrong private ID mapping for publishers in VideoRoom (fixes #1760)
  * A couple of fixes after static analysis
  * Fixed warning in SIP plugin (see #1756)
  * Added new announcement request to TextRoom (#1758)
  * Use the correct range for small delta twcc feedbacks (0-255). Handle potential overflows using MAX/MIN short values. (#1757)
  * SIP plugin: Hangup reason_text (#1756)
  * Improve the parsing of "timeout" attribute in RTSP SETUP answer.
  * Adding properties to config must replace old ones (#1753)
  * Fix automatic reconnection in MQTT transport (#1737)
  * Fix again twcc feedback, sticking reference_time to uint64 (see #1733).
  * Tear down PeerConnection if janus_ice_setup_local fails (see #1735)
  * Fixed segfault when closing handles failed due to port exhaustion (see #1735)
  * Added new video to documentation
  * Fixed wrong math for updated BWE reference time (see #1733)
  * add some missing types (#1748)
  * Fixed incorrect conversion of reference time in BWE (thanks @ibc! fixes #1733)
  * Add a stop_recording parameter to mp "disable" request, to let Janus keep recording a disabled mountpoint. (#1749)
  * Update janus.c (#1731)
  * Reset media attributes (#1730)
  * Minor streaming fixes (#1734)
  * Use a mutex around janus_streaming_rtsp_connect_to_server to avoid collisions on used ports.
  * Check room pointer before notifying a join.
  * valgrind: suppress internal openssl warnings (#1739)
  * janus: avoid NULL dereferences of ice_handle->stream (#1742)
  * Fix macro names to not use reserved identifiers (fixes #1725) (#1729)
  * Fixed typo in RTSP configuration
  * Add a reference to any streaming helper pkt queue. (#1686)
  * Added arrival time of packets to .mjr files (backwards compatible) (#1719)
  * Split audio video media addresses (#1727)
  * feat(textroom): listparticipants (#1723)
  * Fixed directives indentation to match code style (see #1709)
  * Add MQTT v5 support (#1709)
  * Send copies of events to handlers when more than one is active
  * Fixed memory leak in MQTT event handler
  * Removed unneeded verbose output when initialising event handlers at startup
  * Fixed a bug where re-INVITE isn't offered to the called party for handling if autoaccept-reinvites=FALSE (#1721)
  * Fixed small leak in SIP plugin
  * Fixed typo
  * Handle plugin message requests asynchronously also when coming from Admin API
  * Tool to convert .mjr files to .pcap (#1718)
  * Fixed wrong timing info in postprocessing summary for audio
  * Fixed broken .wav files when postprocessing G711/G722 recordings (fixes #1716, replaces #1717)
  * Fix typo in textroom_handle_admin_message to stop segfault (#1707)
  * add configurable maxBitrate values for simulcast encodings (#1706)
  * Fixed broken SDP when rejecting audio/video m-line
  * [Fix]: janus.js client bug, 'for in loop on array' is a risk since it's take in account any object defined on array as key, for example, if you has prototyped a array herite it, as Array.prototype.mean = function (){... it will fail =( (#1693)
  * Removed unneeded extra check
  * Allow audio and video to negotiate SRTP separately (SIP plugin) (#1682)
  * Bumped to version 0.7.4
  * Fixed a few typos after static analysis
  * Fixed a few typos after static analysis
  * Fix Janus not sending DATA_CHANNEL_ACK when requested stream id = 0. (#1695)
  * Fix a crash in webm post processing.
  * Fixed broken usage of GSource for RTCP support in RTP forwarders (#1694)
  * Reverted end-of-candidates change done in #1670
  * Added CommCon presentation (multistream support) to list of videos in FAQ
  * Fixed leak in SCTP code (fixes #1687)
  * Fixed broken H.264 simulcast in Streaming plugin
  * Don't print errors on empty candidate strings
  * Fix release sctp resources if the creation of the association fails (#1673)
  * Bump number of SCTP streams to 300, to make Firefox 69 happy (see #1679)
  * Fixed several datachannel issues (fixes #1679)
  * Fixed typo
  * ice: avoid dereferencing component if NULL (#1678)
  * Update bower.json and package.json (#1677)
  * Made AudioBridge create API more consistent with the static config (fixes #1676)
  * Advertize SSRC even when not sending media (fixes #1558)
  * Better check on transceivers support in janus.js
  * Don't set port to 0 when m-line becomes inactive
  * Updated web demos to use the new slowLink info (see #1664)

-------------------------------------------------------------------
Wed Apr 22 15:06:09 UTC 2020 - ancor@suse.com

- Update to version 0.9.3:
  * Change libsrtp detection in the configure script to use pkg-config
  * Fixed compilation error with gcc10
  * Fixed RTCP issue that could occasionally lead to broken retransmissions when using rtx
  * Added option to specify DSCP Type of Service (ToS) for media streams
  * Fixed a couple of race conditions during renegotiations
  * Fixed VideoRoom and Streaming "destroy" not working properly when using string IDs
  * Fix occasional segfault in VideoRoom (thanks @cb22!)
  * Fixed AudioBridge "create" not working properly when using string IDs
  * Added support for playing Opus files in AudioBridge rooms
  * Added support to Opus files for file-based mountpoints in Streaming plugin
  * Added support for generic metadata to Streaming mountpoints
  * Streaming plugin now returns mountpoint IP address(es) in "create" and "info", when binding to specific IP/interface
  * Fixed occasional segfault when using helper threads in Streaming plugin
  * Fixed occasional race conditions in HTTP transport
  * Added support for specifying screensharing framerate in janus.js (thanks @agclark81!)
  * Cleaned up code in janus.js (thanks @alienpavlov!)
  * Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)
  * Converted HTTP transport plugin to single thread (now requires libmicrohttpd >= 0.9.59)
  * Fixed .deb file packaging (thanks @FThrum!)
  * Added foundation for aiortc-based functional testing (python)
  * Fixed occasional audio/video desync
  * Added asynchronous resolution of mDNS candidates, and an option to automatically ignore them entirely
  * Updated default DTLS ciphers (thanks @fippo!)
  * Added option to generate ECDSA certificates at startup, instead of RSA (thanks @Sean-Der!)
  * Fixed rare race condition when claiming sessions
  * Fixed rare crash in ice.c (thanks @tmatth!)
  * Fixed dangerous typo in querylogger_parameters (copy/paste error)
  * Fixed occasional deadlocks in VideoRoom (thanks @mivuDing and @agclark81!)
  * Added support for RTSP Content-Base header to Streaming plugin
  * Fixed double unlock when listing private rooms in AudioBridge
  * Made AudioBridge prebuffering property configurable, both per-room and per-participant
  * Added G.711 support to AudioBridge (both participants and RTP forwarders)
  * Added called URI to 'incomingcall' and 'missed_call' events in SIP plugin (in case the registered user is associated with multiple public URIs)
  * Fixed race conditions and leaks in VideoCall and VoiceMail plugins
  * Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)
  * Added configurable global prefix for log lines
  * Implemented better management of remote candidates with invalid addresses
  * Added subtype property to differentiate some macro-types in event handlers
  * Improved detection of H.264 keyframes (thanks @cameronlucas3!)
  * Added configurable support for strings as unique IDs in AudioBridge, VideoRoom, TextRoom and Streaming plugins
  * Fixed small memory leak when creating Streaming mountpoints dynamically
  * Fixed segfault when trying to start a SIP call with a non-existing refer_id (thanks @tmatth!)
  * Fixed errors negotiating video in SIP plugin when multiple video profiles are provided
  * Updated SIP plugin transfer code to answer with a 202 right away, instead of sending a 100 first (which won't work with proxies)
  * Added several features and fixes several nits in SIP demo UI
  * Fixed janus.js error callback not being invoked when an HTTP error happens trying to attach to a plugin (thanks @hxl-dy!)
  * Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)
  * Refactored core-plugin callbacks
  * Added RTP extensions termination
  * Removed requirement to enable ICE Lite to use ICE-TCP, even though it may cause issues (thanks @sjkummer!)
  * Added support for transport-wide CC on outgoing streams (feedback still unused, though)
  * Dynamically update NACK queue size depending on RTT
  * Fixed risk of RTP header memory misalignment when dealing with rtx packets
  * Users muted in AudioBridge by an admin are now notified as well (thanks @klanjabrik!)
  * Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)

-------------------------------------------------------------------
Thu Feb 13 13:29:48 UTC 2020 - ancor@suse.com

- Update to version 0.8.2:
  * Updated Changelog (0.8.2)
  * Janus Travis CI integration (#1932)
  * Added Coverity badge
  * Small tweaks after static analysis
  * Fixed helpers not being able to send SUBSCRIBE requests in SIP plugin
  * Removed deprecated text from screensharing demo
  * Removed deprecated warning in screensharing demo
  * Fixed broken links in docs (plugins list)
  * typo (#1934)
  * Fix g_async_queue usage (#1929)
  * Remove odd respond to automatically responded OPTIONS request (#1930)
  * Updated man file for janus-pp-rec
  * Add audio skew compensation to janus-pp-rec. (#1870)
  * Add math library when fuzzing locally.
  * Add missing mutex unlocks in videoroom message handler.
  * Fixed undefined reference when building fuzzers
  * Better parsing of RTSP messages (see #1922) (#1925)
  * Fixed undefined reference when building postprocessor utilities
  * Add new configuration property to add protected folders not to save to (#1919)
  * Added missing check on SDP attribute value existence
  * Added check on AudioBridge instance in setup_media (fixes #1923)
  * Fixed reference to deprecated configuration file
  * More verbose output on postprocessing output error
  * Fix a possible race condition when joining as a subscriber and destroying the session. (#1911)
  * Bumped to version 0.8.2
  * Updated Changelog
  * Increase buffer when post-processing VP8/VP9 recordings too (see previous commit)
  * Fixed occasional buffer overflow error when post-processing H.264 recordings
  * Use sendBeacon instead of sync XHR in onbeforeunload (fixes #1902) (#1918)
  * Updated year in demos and docs
  * Don't keep TextRoom plugin loaded if data channels were not compiled
  * Fixed warnings when building DTLS bio code
  * Added reference to Snap repo in resources (docs)
  * Fixed late initialization of janus.js constructor callbacks (fixes #1912)
  * Move loggers cleanup to end of logger thread (fixes #1904)
  * fixed typo (#1916)
  * Only close the event handlers directory if it was opened (see #1903)
  * startup: only close the logger directory if it was opened (#1903)
  * Fix out of bounds array access for last_spatial_layer (#1906)
  * Fixed occasional memory leak in Streaming plugin (fixes #1900)
  * Fixed leak in SIP plugin (fixes #1897)
  * Fixed warnings introduced in #1896
  * he 'referred_by' field currently holds the SIP URI value copied from the (#1896)
  * Add in mountpoint/forwarder create response the allocated RTCP ports.
  * Check if rtcp port is > 0 before creating a RTCP socket, in Videoroom plugin.
  * Revert "Check if rtcp port is > 0 before creating a RTCP socket."
  * Check if rtcp port is > 0 before creating a RTCP socket.
  * Allow RTCP ports to be picked randomly using 0, in Streaming plugin
  * Fixed typo in SIP plugin
  * Binary data support in data channels (#1878)
  * Remove SIPre plugin from the repo (#1894)
  * Bumped to version 0.8.1
  * Updated changelog (v0.8.0)
  * Added fwrite checks in record.c (warnings only)
  * Fixed variable shadowing
  * Make sure the installed libcurl knows about CURL_AT_LEAST_VERSION
  * Fixed obsolete value for TWCC period default in docs/hints
  * Fixed small typos in demos
  * Make sure libcurl is available before using CURL_AT_LEAST_VERSION (fixes #1887)
  * [Suggestion] Started the refactoring of the janus.js (#1830)
  * Added changelog (and info on tagged versions) to documentation
  * Fix RTSP SETUP when url includes query string parameters (fixes #1869) (#1875)
  * Add CHANGELOG.md file into the project (#1885)
  * Added link to new video on Simulcast and SVC to docs
  * Fixed wrong default folder for loggers
  * Fixed exception to GPL code (see #713)
  * Avoid gzip functions when fuzzing in OSS and add zlib dependency when fuzzing locally.
  * Updated documentation to include some info on the new logger modules
  * Remove option to enable rtx (now always supported, when negotiated) (#1877)
  * Fixed linking error for post-rocessing tools after recent changes
  * New category of plugins for modular logging (#1814)
  * Gzip compression utility in the core (and sample event handler) (#1846)
  * Bumped to version 0.8.0
  * SIP plugin: custom (non-standard) headers on incoming events (requests) (#1873)
  * Reduced default twcc_period value from 1s to 200ms
  * Reduced verbosity of some lines in the SIP plugin
  * New functionality to add custom Contact URI params to SIP REGISTER (#1874)
  * Fixes to RTSP latching procedure (fixes #1536, replaces #1851) (#1866)
  * Bumped version in postprocessing tool as well
  * Don't send RTCP SR if outgoing media has been disabled via SDP update
  * Keep track of clock rates associated to payload types, for RTCP
  * Feature/ignore unreachable ice server (#1854)
  * Fixed wrong clock rate being used for RTP header updates when using G.722
  * Don't scan libnice version if it wasn't retrieved (fixes #1858)
  * add missing closing curly bracket (#1859)
  * Fixed rare race condition in HTTP plugin that could cause leak (fixes #1665)
  * Fix RTP fuzzing target according to recent VP9 changes.
  * Fixed regression when setting up DataChannels
  * Fixed broken code in AudioBridge
  * Use strtol more, and add checks when atoi is used (#1852)
  * Fixed SIP hangup not sending CANCEL, when inviting (fixes #1856)
  * VP9 SVC fixes (#1849)
  * fix nullptr dereference in streaming plugin (#1855)
  * Fixed typo
  * Add exception var to catch stmt to fix rollup (#1848)
  * Updated link to project in resources (docs)
  * Split lines on line feed only, and trim carriage feed instead
  * Skip multiple b= line break conditional for b=TIAS (#1832)
  * ice: ignore/enforce only when IP starts with the partial-string from the list (#1840)
  * IPv6 support in Streaming plugin (#1807)
  * Add support for domain names (and IPv6) to RTP forwarders (#1778)
  * Support for SIP transfers (#1815)
  * Support for simultaneous calls in SIP plugin (#1772)
  * Bumped to version 0.7.6
  * Fixed check
  * Update getStats() to use a promise instead of callback (#1823)
  * Improved attach/reattach MediaStream helpers avoiding browser versions (#1828)
  * Updated libnice recommended version into the README and mainpage.dox files (#1835)
  * Detect new streams also when mountpoint is disabled
  * Revert previous commit (causes crashes, to investigate)
  * Split lines on line feed, and trim carriage feed (see #1818)
  * Avoid locking mountpoints when reconnecting to RTSP servers. Use 5 seconds connection timeout in curl requests.
  * Fixed simulcast issue when automaticlly dropping to lower layers
  * Reduced verbosity of some RTCP related messages
  * Added Admin API command to inject strings in Janus logs from outside
  * Fixed broken check (again) on libwebsockets version (see #1812)
  * Fixed a few typos in the documentation
  * Fixed outdated text in documentation
  * Clear publisher's room pointer when leaving and add a reference while executing janus_videoroom_leave_or_unpublish. (#1795)
  * Make sure flags are cleared when getting a close_pc() even when a PeerConnection wasn't created (fixes #1800)
  * Fixed broken check on libwebsockets version (see #1812)
  * Fixed compilation error of WebSocket event handler with older version of libwebsockets (fixes #1812)
  * Added reference to JanusCon to the FAQ for learning material
  * --cwd-path (Current Working Directory) CLI option added (#1804)
  * Added new Nanomsg event handler (#1802)
  * Added new WebSockets event handler (#1799)
  * Hangup Custom Headers (#1809)
  * Wait for keyframe when dropping to lower simulcast layer because of inactivity (fixes #1806)
  * Reduced verbosity of successful mDNS resolves
  * Fix a missing pop on Duktape stack when invoking resumeScheduler.
  * Added warning if libnice version is outdated (at least 0.1.15 recommended)
  * Hook Lua print function(s) to Janus logger (#1782)
  * Write moov atom at the head of the MP4 file (#1791)
  * Support for SIP SUBSCRIBE/NOTIFY in SIP plugin (#1768)
  * sdp-utils: check that janus_sdp_get_codec_rtpmap succeeded (#1785)
  * Add some missing atomic checks in videoroom plugin.
  * Mute participants (#1787)
  * Fixed participant ID being reset in AudioBridge web demo
  * Allow for capturing desktop audio when sharing screen (#1771)
  * Duktape getVersion method added (#1786)
  * SIP plugin: add local interface for SDP binds (#1784)
  * Better async managament of new mountpoints with temp map of IDs (#1732)
  * Fixed potential endless loop in Streaming plugin when binding ports (fixes #1762, replaces #1763)
  * Add command line option to janus-pp-rec to specificy the output format (#1777)
  * Don't remove room for subscriber if not closing PeerConnection (fixes #1761)
  * JavaScript logging improved (#1781)
  * moved destroySession connection condition (#1783)
  * Ignore temporary SSL errors in RabbitMQ transport (see #1769)
  * Fixed typo in SIP demo (DTMF digits message)
  * Typo fix in JANUSSDP.removePayloadType (#1779)
  * Updated version in bower package.json too
  * Fixed broken negotiation in SIP plugin for mandatory SDES-SRTP (fixes #1770)
  * Added option to specify local port when testing STUN server via Admin API
  * Fixed broken responses to incoming SIP INFO and MESSAGE requests
  * Add audio level dBov average to talk events in VideoRoom plugin (#1751)
  * Fixed outdated reference to old configuration files in demos
  * Fixed broken indentation
  * Bumped to version 0.7.5
  * Small tweak to verbose output (see #1740)
  * Fixed handling of offerless reINVITE. (#1740)
  * Hopefully final fix for RTCRtpTransceiver check (see #1759)
  * Improved RTCRtpTransceiver check (see #1759)
  * Fixed RTCRtpTransceiver check for Edge (fixes #1759)
  * Fixed wrong private ID mapping for publishers in VideoRoom (fixes #1760)
  * A couple of fixes after static analysis
  * Fixed warning in SIP plugin (see #1756)
  * Added new announcement request to TextRoom (#1758)
  * Use the correct range for small delta twcc feedbacks (0-255). Handle potential overflows using MAX/MIN short values. (#1757)
  * SIP plugin: Hangup reason_text (#1756)
  * Improve the parsing of "timeout" attribute in RTSP SETUP answer.
  * Adding properties to config must replace old ones (#1753)
  * Fix automatic reconnection in MQTT transport (#1737)
  * Fix again twcc feedback, sticking reference_time to uint64 (see #1733).
  * Tear down PeerConnection if janus_ice_setup_local fails (see #1735)
  * Fixed segfault when closing handles failed due to port exhaustion (see #1735)
  * Added new video to documentation
  * Fixed wrong math for updated BWE reference time (see #1733)
  * add some missing types (#1748)
  * Fixed incorrect conversion of reference time in BWE (thanks @ibc! fixes #1733)
  * Add a stop_recording parameter to mp "disable" request, to let Janus keep recording a disabled mountpoint. (#1749)
  * Update janus.c (#1731)
  * Reset media attributes (#1730)
  * Minor streaming fixes (#1734)
  * Use a mutex around janus_streaming_rtsp_connect_to_server to avoid collisions on used ports.
  * Check room pointer before notifying a join.
  * valgrind: suppress internal openssl warnings (#1739)
  * janus: avoid NULL dereferences of ice_handle->stream (#1742)
  * Fix macro names to not use reserved identifiers (fixes #1725) (#1729)
  * Fixed typo in RTSP configuration
  * Add a reference to any streaming helper pkt queue. (#1686)
  * Added arrival time of packets to .mjr files (backwards compatible) (#1719)
  * Split audio video media addresses (#1727)
  * feat(textroom): listparticipants (#1723)
  * Fixed directives indentation to match code style (see #1709)
  * Add MQTT v5 support (#1709)
  * Send copies of events to handlers when more than one is active
  * Fixed memory leak in MQTT event handler
  * Removed unneeded verbose output when initialising event handlers at startup
  * Fixed a bug where re-INVITE isn't offered to the called party for handling if autoaccept-reinvites=FALSE (#1721)
  * Fixed small leak in SIP plugin
  * Fixed typo
  * Handle plugin message requests asynchronously also when coming from Admin API
  * Tool to convert .mjr files to .pcap (#1718)
  * Fixed wrong timing info in postprocessing summary for audio
  * Fixed broken .wav files when postprocessing G711/G722 recordings (fixes #1716, replaces #1717)
  * Fix typo in textroom_handle_admin_message to stop segfault (#1707)
  * add configurable maxBitrate values for simulcast encodings (#1706)
  * Fixed broken SDP when rejecting audio/video m-line
  * [Fix]: janus.js client bug, 'for in loop on array' is a risk since it's take in account any object defined on array as key, for example, if you has prototyped a array herite it, as Array.prototype.mean = function (){... it will fail =( (#1693)
  * Removed unneeded extra check
  * Allow audio and video to negotiate SRTP separately (SIP plugin) (#1682)
  * Bumped to version 0.7.4
  * Fixed a few typos after static analysis
  * Fixed a few typos after static analysis
  * Fix Janus not sending DATA_CHANNEL_ACK when requested stream id = 0. (#1695)
  * Fix a crash in webm post processing.
  * Fixed broken usage of GSource for RTCP support in RTP forwarders (#1694)
  * Reverted end-of-candidates change done in #1670
  * Added CommCon presentation (multistream support) to list of videos in FAQ
  * Fixed leak in SCTP code (fixes #1687)
  * Fixed broken H.264 simulcast in Streaming plugin
  * Don't print errors on empty candidate strings
  * Fix release sctp resources if the creation of the association fails (#1673)
  * Bump number of SCTP streams to 300, to make Firefox 69 happy (see #1679)
  * Fixed several datachannel issues (fixes #1679)
  * Fixed typo
  * ice: avoid dereferencing component if NULL (#1678)
  * Update bower.json and package.json (#1677)
  * Made AudioBridge create API more consistent with the static config (fixes #1676)
  * Advertize SSRC even when not sending media (fixes #1558)
  * Better check on transceivers support in janus.js
  * Don't set port to 0 when m-line becomes inactive
  * Updated web demos to use the new slowLink info (see #1664)

-------------------------------------------------------------------
Tue Jun 25 09:42:37 UTC 2019 - ancor@suse.com

- Update to version 0.7.3+git20190625.2a86d527:
  * Updated webrtc-adapter version in demos to 6.4.0 (cdnjs link)

-------------------------------------------------------------------
Tue Jun 25 09:33:06 UTC 2019 - ancor@suse.com

- Update to version 0.4.3+git20190625.e790e176:
  * Changed slowlink to use lost packets instead of NACKs, and made it configurable (#1664)
  * Fixed end-of-candidates in janus.js in other methods as well (see #1670)
  * Notify VideoRoom RTP forwarder events via event handlers (fixes #1671)
  * Fixed the end-of-candidates usage in janus.js (fixes 1670)
  * Fix reference leak (#1666)
  * Removed unneeded check on WebRTC state in end_session
  * Added support for notify_joining to videoroom.lua as well
  * Changed default for sender-side bandwidth estimation in VideoRoom to TRUE
  * Made a few enhancements to the Lua VideoRoom plugin example * Support for multiple codecs (like C VideoRoom) * Support for partial subscriptions * Support for require_pvtid
  * Configurable RTP range in Streaming plugin (replaces #1623, fixes #1616) (#1659)
  * Added new documentation page for recordings
  * Updated description of the Streaming plugin
  * Removed very outdated TODO item
  * Fixed typo
  * Updated obsolete documentation of the Admin API
  * Added flag to the Admin API handle_info to only return plugin-specific info
  * Don't allow plugins to generate/relay their own TWCC RTCP packets
  * fix sdes length when adding to compound (#1663)
  * Be more tolerant when parsing b= attributes in SDP (fixes #1662)
  * enable dtls window size on non-MacOS machines (#1660)
  * Added status messages to MQTT transport (#1631)
  * Refactored janus-pp-rec to support command line options (#1656)
  * New Admin API method to make synchronous requests to plugins (#1647)
  * Bumped to version 0.7.3
  * Adding reference to Haskell binding of Janus client protocol using WebSocket transport (#1657)
  * Fixed segfault when changing rooms in AudioBridge (fixes #1655)
  * Fixed segfault in WebSockets transports when using ACL
  * Add libcurl to the streaming plugin flags.
  * Added new Admin API messages Destroy session, detach handle, hangup PeerConnection
  * Don't add ssrc-group if we're not putting any ssrc in the SDP
  * Fixed possible issue in subscriber renegotiation (fix #1651)
  * Set remote candidates when handling an answer if some candidates have already been saved.
  * Fixed typo (fix #1650)
  * Generate error only if added remote candidates is negative.
  * Check the return value of nice_agent_set_remote_candidates.
  * Improvements on writable notifications in WebSockets transport (#1638)
  * Added support for third spatial layer when using VP9 SVC (assuming EnabledByFlag_3SL3TL is used)
  * Set ICE remote credentials when receiving remote SDP, instead of later (#1635)
  * Remove end-of-candidates attribute as well when anonymizing SDP for plugins
  * Fixed leak when RTP forwarding with RTCP feedback (fixes #1605)
  * Add a reference to the session when logging SIP messages (see #1636)
  * Add frame marking parsing function to the rtp fuzzing target.
  * Removed requirement for both sdpMid and sdpMLineIndex to be in trickle messages
  * Small tweak to websockets connection destruction
  * ice: avoid NULL dereference in stats callback (fixes #1633)
  * Apply same fix to SIPre plugin
  * Answer with a busy response if a SIP relayer is still active.
  * Fixed exception in janus.js when using datachannels
  * Added option to locally cleanup handles when destroying a session in janus.js
  * Bumped to version 0.7.2
  * Fixed a few issues saving permanent mountpoints in Streaming plugin (see #1630)
  * Fixed some leftovers in the docs
  * Added sanity checks on createOffer/createAnswer in janus.js
  * A couple of fixes on SIP race conditions after hangups (#1611)
  * Add some comments in fuzzer run.sh to instrument libfuzzer to detect timeout and out-of-memory errors.
  * Add a SDP fuzzer timeout crash file.
  * Normalize fuzzers crash filenames.
  * Abort sdp parsing when a m= line is too long.
  * Separate checks for PeerConnection and getUserMedia support
  * Streamlined navbar in demos and documentation
  * Added link to Slideshare in the FAQ as well
  * Added two more presentations to the videos section in the FAQ
  * New experimental debug mode with disabled WebRTC encryption (#1622)
  * Add version of dependencies to server info (Janus and Admin API) (#1618)
  * Fixed regression in simulcasting when doing SDP munging in janus.js
  * H.264 temporal scalability support via frame-marking extension (#1615)
  * Added link to JanusCon to docs as well
  * Added link to JanusCon to demos navbar
  * Send PLI on all layers, when simulcast is used
  * Fixed Streaming plugin compilation issue for when libcurl isn't used
  * Handle recvfrom failures (#1614)
  * Allow payload type override when creating RTSP mountpoints (#1609)
  * Added convenience method to Admin API to test STUN server
  * Fixed segfault in SIP plugin when using event handlers (see #1613)
  * Added convenience method to Admin API to test address resolving capabilities
  * Added ping/pong request to Admin API as well
  * Fixed regression in Streaming plugin RTCP support
  * Added option to lock RTP forwarding functionality via admin_key
  * Check if the ICE candidate gathering started at all
  * Bumped to version 0.7.1
  * janus.js: fix copy paste error which broke answer (#1607)
  * Added count of received retransmissions to Admin API and event handlers
  * Ported fix from #1601 to master as well
  * ice: fix inline warning (#1602)
  * Added proper management of incoming re-INVITEs to SIP plugin (see #1591) (#1597)
  * Add an interception callback to js API (#1599)
  * Added more videos to the documentation
  * Fixed warning when building SDP fuzzer
  * Initial integration of SDP fuzzing (for SDP utils) (#1594)
  * Use json_loadb instead of janus_loads in RabbitMQ transport
  * [Fuzzers] Use shared libraries when executing locally.
  * Fix some potential crashes in the h264 postprocessor due to invalid packets.
  * Fixed several leaks in SDP utils
  * Notify VideoRoom passive attendees when joining as well, if notify_join is TRUE
  * Streamlined the management of outgoing messages in the RabbitMQ transport
  * Don't use DTLSv1_2_method() when using LibreSSL
  * Make configure script shell compatible
  * Added more videos to the documentation
  * Added explicit check when registering in SIP plugin (fixes #1522)
  * Remove encrypted extensions when exchanging SDPs with plugins (see #1575 and #1581)
  * Don't use old constraints where transceivers are available (fixes #1583)
  * Check destroyed flag before sending on a websocket.
  * Fix broken fuzzer build script.
  * Added new paper to citations in documentation
  * janus_streaming: guard against NULL rtpmaps
  * fuzzers: allow extra CFLAGS and/or LDFLAGS to be appended
  * sdp-utils: avoid extra carriage return for extmap lines
  * Add files via upload
  * Add check-fuzzers target to Makefile. It reuses existing Janus objects to execute regression testing with corpora files.
  * Fuzzer run script: list the files before running with a specified folder, change and move some comments.
  * Arrow functions didn't work in IE
  * Fuzzing run script: use script folder in place of pwd, transform crash file path to an absolute path, detect if the supplied crash file is a file or a folder, rename coverage output files and write them in OUT folder.
  * Use CC as linker before falling back to default value in fuzzing build script.
  * Fix RTP fuzzer building error and add jansson dependency when building fuzzers.
  * Fixed H.264 keyframe detection, especially when simulcasting
  * Don't call getUserMedia when audio and video are false (e.g. when using removeVideo)
  * Changed REMB behavior from 'cap' to 'overwrite', and improved 'no limit' setting EchoTest and VideoCall
  * Better H.264 keyframe check (fixes #1552)
  * Added temporary 'simulcast2' query string parameter to demos to test rid-based simulcasting on Chrome >= 74
  * Force DTLS 1.2 when using older OpenSSL versions
  * Bumped to version 0.7.0
  * More explicit check on replaceTrack support when using Safari (see #1550)
  * Use replaceTrack for Safari as well (fixes #1550)
  * Reduced verbosity of Streaming mountpoint helper threads logs
  * Fixed another typo in Streaming plugin
  * Fixed issue when switching mountpoints powered by helper threads
  * Support for multiple datachannel streams in the same PeerConnection
  * Fixed missing simulcast change notifications at startup
  * Some more fixes on rid-based simulcasting Note: rids in JS must be added from high to low
  * Only add extension IDs to Admin APIs if they were negotiated
  * Fixed typos
  * Fixed typo
  * Increase received counter for any packet received on non-rtx-enabled streams.
  * Calculate jitter only after the first iteration. Increase threshold for rtx detection to 120 ms.
  * Simplify the parsing of SSRCs in SDP for video streams
  * Reset extension IDs if they're not negotiated in the answer
  * Don't pass simulcast attribute along, when anonymizing SDP
  * Updated simulcast fallback in SIP, SIPre and NoSIP plugins
  * Improved support of repaired rid extension
  * Put rid-related stuff in a different object in the Admin API
  * Put rid extension IDs in the Admin API report (fixed)
  * Put rid extension IDs in the Admin API report
  * Added query string parameters to force codecs in EchoTest demo
  * Added experimental support for repaired-rtp-stream-id extension as well
  * Fixed MQTT publish errors (fixes #1535)
  * New method to be fuzzed in rtp_fuzzer. Add a couple of crash files for RTP. Specify crash file as the second argument of run.sh
  * Check for extension length when parsing twcc sequence number.
  * Fix transport-wide sequence number parsing.
  * Fixed broken TWCC negotiation when disabled in VideoRoom config
  * Add Janus type definitions for better developer experience
  * Better support for rid-based simulcasting
  * Do not drop RTP packets with empty payload.
  * Discard outgoing empty RTP packets
  * Added new project to the resources in the docs
  * Fix and enhance RTCP stats calculation for loss and jitter. Fix link quality metric estimation.
  * fuzzers: make jobs and workers configurable via environment (#1542)
  * Support for mid RTP extension, and better extmap negotiation in SDP utils (#1543)
  * Added missing wakeup call
  * Explicitly mark packet as unencrypted, when sending retransmissions via rtx
  * Fixed typo in doxygen docs
  * fix incorrect value for admin_http in instructions
  * Added a few simulcast tweaks in VideoRoom - new boolean property to tell if publisher is simulcast in events - ability to specify substream/temporal layer when joining, for subscribers
  * Fixed typo in RTCP packet, and made sure cname is the same for all m-lines in the SDP
  * Removed folder with self-signed certificate: DTLS certificates are autogenerated anyway if missing, and HTTPS/WSS need valid/better ones
  * Bumped to version 0.6.3
  * Allow opaqueID to be added to Janus API events, if configured
  * Generic fixes from static analysis
  * Fixed missing newline
  * Check for the right method in Janus.isWebrtcSupported of janus.js (fixes #1527)
  * A couple of fixes on Firefox simulcasting in janus.js
  * Added option to negotiate inband FEC for Opus in VideoRoom and EchoTest (#1525)
  * Added option to specify temporary extension when recording AudioBridge rooms, and event handler notification for when recording is over
  * Don't show warnings if we don't know the SSRC yet
  * Configurable TWCC feedback period
  * Fixed check in janus.js
  * Force unified-plan sdpSemantics in janus.js if Chrome >= 72
  * Do not insert a Report Block when sending REMBs.
  * Force plan-b semantics if Chrome is < 72
  * Reset NACK queue only when receiving a KeyFrame with a highest sequence number.
  * Add code for RTCP and RTP fuzzing. (#1492)
  * Added define for number of Opus samples (see #1520)
  * Fixed typo in janus.js (fixes #1521)
  * Link to the math library explicitly for the HTTP event handler (fixes #1517)
  * Update janus.js to use navigator.mediaDevices.getDisplayMedia instead of navigator.getDisplayMedia
  * Close the PeerConnection from the plugin after a successful record/play (fixes #1513)
  * Push local SDP to handlers before the event (fixes #1510)
  * Changed default maxev to 10 in janus.js
  * Use RTCRtpSender.getCapabilities if possible to detect VP8 support in Safari
  * Fixed defaults for allowed publisher's media
  * Fixed multiple watch requests in streaming demo
  * Removed old yes/no references in config files and docs (true/false)
  * Bumped to version 0.6.2
  * Fixed a couple of early decreases (fix originally contributed as a PR in #1501)
  * Fix some wrongs printf formats.
  * Reverted debug console log
  * More fixes to RTP parsing.
  * Disabled mid and rtp-stream RTP extensions (fixes PlanB browsers not working in some demos)
  * Added option to SIP/SIPre/NoSIP plugin to override c= IP in SDP (fixes #1504)
  * Fixed recordings sometimes not destroyed when hanging up SIP sessions (fixes #1500)
  * Increase payload ptr for rtx packets.
  * Fixes for RTP issues discovered while fuzzing.
  * Added check on minimum size for RTCP packets
  * Removed unneeded check (already in helper method)
  * Moved protocols demultiplex helpers to respective headers, to use them in plugins
  * PR comments: memory leak fix and proper comment indentation.
  * Fix infinite loop when an HTTP connection breaks
  * Implement on SIPre plugin. Call-ID on error events.
  * Send call_id to all SIP plugin events related to call.
  * Add pragma to ignore clang warning in g_vsnprintf.
  * Support for custom Call-ID header in SIP plugin.
  * enables extended mount point info by default if no secret is assigned
  * Evaluate RTCP transit with a signed integer.
  * Added Admin API command to stop accepting sessions (e.g., to drain server)
  * Fixed missing prefix when saving Streaming mountpoints with no name to libconfig
  * Make sure element is not null, when saving libconfig files
  * Use transceivers when Chrome >= 72 too
  * Fixed define for TURN REST API (was unnecessary requirement for RTSP support)
  * Fixed typo in verbosity of Streaming plugin log line
  * Fixed deprecated syntax in configs documentation
  * Added more checks when inspecting VP9 payload descriptor
  * Added more checks when inspecting VP8 payload descriptor
  * Added more checks when doing VP9 or H.264 keyframe detection
  * Fixed crash when fuzzying data for VP8 keyframe detection
  * Secure janus_rtcp_remove_nacks.
  * Fix for previous commit.
  * Calculate REMB bitrate in uint64 to make sanitizers happy.
  * Secure janus_rtcp_filter function and avoid a possible memory leak.
  * Updated year in docs and web demos
  * Drop RTCP packet if parsing fails. Avoid possible leak in janus_rtcp_get_nacks. Fix return value in janus_rtcp_cap_remb.
  * Updated README (fixes #1461)
  * Fix broken build due to previous commit.
  * Improve clang compiler detection in configure.ac
  * Remove some unused legacy code.
  * enable jcfg for duktape, fix #1420
  * Bumbed to version 0.6.1
  * Fixed array usage when munging SDP (see #1439)
  * Set correct export-dynamic flag for MacOS.
  * More idiomatic methods to check FCI payloads. Remove methods for NACKs and length checking.
  * Fix some format specifiers.
  * Small changes in logging and docs.
  * Missing sendDtmf success callback call
  * Replace enable with enabled
  * Change log level for rejected RTCP packets.
  * Severl fixes for RTCP parsing bugs discovered while doing fuzz testing.
  * Change a string in the configure summary.
  * Remove AX_APPEND macros to avoid installing another dependency.
  * Use decrypted packet length (buflen) in some calls that mistakenly used the crypted packet length (len).
  * Suppress cast alignment warnings when using clang.
  * Reduced polling times when waiting for candidates
  * events: guarantee loop termination
  * Use compare_and_exchange to avoid a double logging initialization.
  * Specify C language with AC_LANG macro.
  * Use AX_APPEND_LINK_FLAGS to append a flag to the linker.
  * Move export-dynamic in common CFLAGS. Print the matched compiler in the summary.
  * Refactor
  * Added missing params to json validation
  * Sipre plugin - custom headers in accept request
  * Added custom headers in accept request
  * Improve Makefile.am and configure.ac to better support clang compiler.
  * Fix some wrong print formats and variable types.
  * Integrated fixes from #1470 in other RTCP parsing submethods
  * Document 'user_agent' in 'register' and 'code' in 'decline'.
  * rtcp: fix get_remb bugs
  * Documented 'display_name' parameter in SIP plugin's 'register' request.
  * Make sure the merged SDP is sent to event handlers (fixes #1466, see #1467)
  * Fixed payload type selection for RTX (fixes #1469)
  * Fix code execution order
  * Fix a wrong assignment made in previous commit.
  * Avoid media cleanup while a sip thread is still running.
  * Ignore RTCP if it contains no SSRC
  * Set npt in Range header for RTSP PLAY (fixes #1460)
  * Fix an issue when post-processing h264 streams containing STAP-A fragments not in first position.
  * ice: avoid crash on NACK cleanup
  * Check crypto attribute pointer in sip plugins before parsing.
  * Don’t put rtx packets in retransmission buffer
  * Fixed typos (see #1446)
  * Fixed missing quotes in sample configuration
  * Eliminate dead code + make cfg parsing more robust
  * Refactor status messages to be independent of LWT or something
  * Send message after disconnect too
  * Read initial status message from config
  * Nanomsg transport libconfig migration
  * Set retain for initial message equal to LWT retain
  * Adjust sample MQTT EVH config booleans to new format
  * Fix make rule MQTT EVH config
  * Restored link quality calculation check, and clarified it's there to check for NaN (see #1448)
  * Fixed typo in RTCP code (fixes #1448)
  * Fixed broken reference to deprecated configuration file
  * Fix sample config for mqtt evh
  * Disable LWT by default
  * Initialize Last Will and Testament properties for mqttevh
  * Converted MQTT evh config file to jcfg, as it was still missing
  * Change janus_rtcp_fix_report_data signature to avoid references to RTP structs.
  * Remove received SSRC check in janus_rtcp_fix_report_data for incoming RR.
  * Move recording setting forward in echotest message handling.
  * Fixed some small tweaks in documentation
  * Bumped to version 0.6.0
  * Fix typo for videoroomtest reference link in svc test page
  * Check app_handle pointer before doing a hangup or destroying a plugin session.
  * Updated README text
  * sdp-utils: use enum type instead of defines
  * sdp-utils: minor doc corrections
  * Fixed closing websocket when there's no ws
  * Fix SSRC and timestamp in SSRC reports before passing the packet to a plugin.
  * Fixed stuck Publish button when republishing in VideoRoom demo
  * Normalize bitrate reported by Safari
  * Added check for Safari VP8 support in janus.js init
  * Don't remove mid from answer if m-line was rejected
  * Disconnect ws on timeout gateway message
  * Added pcap/text2pcap controls to Admin API demo page
  * Added .pcap info to Admin API, if available
  * Add support for dumping to .pcap directly
  * Fixed datachannel support in the Streaming demo
  * Improved description of sample H.264 mountpoint
  * Added mjr metadata to (some) media containers when postprocessing recordings (see #1189)
  * Updated text in VideoRoom demo that reminded deprecated syntax
  * Make sure a pop is done after a couroutine ends in the Duktape plugin (fixes #1411)
  * When using the TURN REST API, send the API key as both 'api' and 'key' (fixes #1416)
  * Don't spam SRTP protect errors
  * Fixed initial retransmissions wrongly interpreted as losses
  * Updated the way lost packets are counted
  * Added TWCC placeholder (commented out) in VideoRoom configuration
  * Added TWCC placeholder (commented out) in VideoRoom configuration
  * Fixed occasional bogus valuefor lost packets
  * Added info on whether TWCC is enabled or not in Admin API
  * Removed broken/unneeded lock in TextRoom plugin (fixes #1421)
  * Fixed some missing notifications on temporal layer changes in simulcast
  * Better cleanup of plugin sessions at shutdown
  * utils: avoid unneeded casting away of constness
  * utils: constify read-only parameters
  * read RTP padding len into another buffer
  * print RTP header extension type in uppercase hex
  * Better cleanup of HTTP plugin at shutdown
  * Protect the tables destruction with a mutex when shutting down the HTTP plugin
  * Bumped to version 0.5.0
  * Fixed RTP extensions count in postprocessor when there are CSRC bytes
  * Fixed the keyframe detection for H.264
  * Support for a couple of RTP extensions in the postprocessor
  * Fixed broken H.264 simulcast support
  * Fixed multiple 'first keyframe' notifications when postprocessing videos
  * Fixed typo
  * Force pthread mutex for older OpenSSL thread-safeness locking
  * Removed unneeded debug line
  * Allow for predefined number of threads/loops to handle all media
  * Fixed SIP plugin docs and a broken link in the demos (fixes #1404)
  * Fixed some small nits (code style)
  * Fixed deadlock in AudioBridge (fixes #1406)
  * * Fixed a compatibility issue in janus_streaming_rtsp_connect_to_server().
  * Fix HTMLMediaElement.srcObject for older Chrome (< 52)
  * Better refcounting of AudioBridge participants while mixing
  * store transport seq num before dropping packets
  * * Use OPTIONS instead of GET_PARAMETER to keep a live in streaming plugin to avoid some compatibility issues.
  * Streamlined checks for plugin session validity
  * Reversed checks to avoid error messages when pushing events
  * Only free WebRTC stuff once
  * Removed unneeded atomic flag, and moved Admin API loop property
  * Wrap <pre> text (needed for some generated docs)
  * Fixed instructions for libnice, and fixed wrapping in README
  * Refactored handle loop (and thread) as persistent
  * Reverted previous change...
  * Improved atomic checks when quitting the ICE loop
  * Preparse mid when preparsing SDP
  * Made GMutex/pthread mutex choice configurable (configure script)
  * add endian define/include to pp-rtp.h
  * Fixed broken libwebsockets repo link (see #1395 and #1396)
  * Redefine mutexes to use GMutex instead of pthread_mutex_t
  * Added missing info to AudioBridge documentation
  * More conservative checks in AudioBirdge when handling talk events
  * Added some more checks to make sure the plugin handle is not NULL
  * Free plugin session handles before core handles
  * Better parsing of SPS for H.264 non-baseline
  * Use default resolution if postprocessing an H.264 gives a broken one (see #1393)
  * Removed leftover code from SIPre and NoSIP plugins
  * Move silly comment.
  * Add missing operations in video skew.
  * Change logging level in a couple of prints.
  * Fixed broken check on setSinkId in Device Test demo
  * Increase thresholds to 120 milliseconds.
  * Add an evauation and tuning phase to the skew compensation algorithm.
  * Fixed typo in SIP plugin docs (fixes #1391)
  * Fix broken Record-Route support in ACK, and remove deprecated autoack option (fixes #1389)
  * Bumped to version 0.4.5
  * Small edits in some comments from #1386
  * Fix SIP MESSAGE support in SIP plugin (fixes #1388)
  * Fixed comments and indentations
  * Fixed indentation
  * Additional checks when pushing events in Duktape (see #1384)
  * Fixed connectivity establishment when only candidates available are prflx
  * Corrected some comments
  * Limit packet counts per single transport wide cc FB message
  * Generate last chunk of transport wide cc fb msg correctly
  * Don't do a new getUserMedia if we're keeping all tracks
  * Don't do a new getUserMedia for a media if we're not updating it
  * Fixed potential deadlock in Lua and Duktape plugins (see #1384)
  * Fixed leak in the AudioBridge and VideoRoom plugins
  * Fixed leak in the TextRoom plugin
  * Allow EchoTest audio/video codecs to negotiate to be overridden
  * Re-add warning about large packets.
  * Removed unneeded playsinline attribute from <audio> elements
  * Added same #1381 fix to SIPre plugin as well
  * UI tweaks to make that silly iOS thing happy
  * Fix crashing of janus gateway due memory corruption
  * Make sure the file isn't empty before loading the Duktape script
  * Fix code indentation
  * Fixed segfaults when postprocessing some problematic recordings
  * Fix janus_ducktape plugin - remove input js file size limit
  * Introduce OpenSSL BIO that directly writes to the libnice agent.
  * Don't unref event if handler dumped it (the core does already)
  * Made BoringSSL DTLS initial timeout configurable in janus.cfg (fixes #1377)
  * Don't remove JS files (including janus.js) after a make clean (fixes #1378)
  * A couple of fixes related to #1374
  * rtcp: division by zero
  * minor memleak on getifaddrs()
  * mqevh: minor memleak
  * Fixed small nits and typos post #1363
  * Allow more room for the address when parsing candidates (needed for mDNS from Safari)
  * Small nits after merging #1368
  * nosip: better port choice (fixed number of attempts replaced with a full scan within a given range)
  * Applied same NoSIP #1367 fix to SIP and SIPre plugins
  * nosip: resource leak if no answer received
  * close fds only when fd != -1
  * Respond with the list of common codecs, when accepting a SIP call
  * Fixed RTP forwarding regression in the VideoRoom (see #1372)
  * streaming plugin: use even RTP port for RTSP
  * janus_streaming_rtsp_connect_to_server(): handle SETUP error correctly
  * ice: typo (flag cleared multiple times)
  * ice: very minor fix
  * autotools: unconditionally distribute sources
  * autotools: distribute data
  * autotools: ensure man pages are distributed
  * Fix videoroom crash when require_pvtid is true (issue #1362).
  * Added experimental support for mDNS candidates
  * Drop mDNS candidates (no resolve support right now)
  * Added simulcast support to the Record&Play plugin
  * Added some missing context resets at startup/cleanup
  * Updated information on suggested libnice versions
  * Fix parallel build
  * Clarified text in janus.cfg when discussing AWS deployments
  * Fixed warning when compiling event handlers (missing fix after merging #1349)
  * Added simulcasting support to VideoRoom RTP forwarders
  * restore previous logic for detecting codecs during videocall accept
  * clear nacked packet properly in case of rfc4588
  * Handle when retransmission buffer is NULL
  * Added simulcasting support to VideoRoom publisher recorders
  * Refactored simulcasting processing code as a core function
  * Fixed missing temporal layer controls in demos for VP8 simulcasting
  * Allow use of HTTP urls without session/handle in the path
  * Removed unneeded mutex unlock (see #1350)
  * fix (janus): unneeded mutex_unlock() on handle_create() fail
  * Replaced old and confusing 'gateway' label pretty much everwhere
  * Fixed calculation of lost packets, taking successful retransmissions into account
  * Fixed typo in copying sequence number when using RFC4588
  * Fixed broken NACK generation for consecutive losses
  * New MQTT event handler
  * Fixed reporting of event handlers in Admin web page
  * Added new --json command line option to postprocessor to only print JSON header of .mjr files (see #1189)
  * ES6 to ES5
  * Removed unneeded line (copy/paste typo)
  * Update janus.js
  * use dtls timeout value only when DTLSv1_get_timeout is succeed
  * Fixed inconsistency in pointer usage in the VideoRoom
  * Fixed typo in VideoRoom docs
  * Fixed TWCC related change
  * fix (nosip): open local ports according to audio/video presence
  * Notify leaving event in videoroom.lua
  * Added support for the experimental getDisplayMedia API for screensharing to janus.js
  * Fixed typo
  * Fixed leak when using transport-side CC (see post MFhFs6_sczo on group)
  * fix (nosip): pass sdp_update flag to janus_nosip_sdp_process() to trig session parameters update within a relay thread.
  * fix (nosip): set 'sdp_update' flag if more answers received to current session
  * relay REMBs of the lowest peer bandwidth estimation instead of the highest
  * fix (nosip): typo '==' -> '!='
  * Don't touch the bitrate on slowlinks in EchoTest and VideoCall, just notify
  * Added configurable candidates timeout (see #1335)
  * Notify VideoRoom kicked/leaving/unpublished via event handlers (see #1320)
  * Fixed typo
  * Bumped to version 0.4.4
  * Fixed fence around hangup_media in all plugins (see #1337)
  * Typo
  * My solution
  * Removed duplicate check
  * Janus.js reconnect() fix
  * Duplication fix
  * js: use Promise for WebRTC operations
  * url-encoding token and apisecret parameters
  * Add RTCP support to RTP forwarders
  * Tweaked text in the issue placeholder text
  * Moved CONTRIBUTING.md file back to the hidden folder
  * Moved CONTRIBUTING.md file to the root
  * Fixed issues placeholder text
  * Fixed broken preprocessor conditions in WebSockets transport (fixes #1327)
  * Truncate PID file before writing to it
  * Add VP9-SVC support to the Streaming plugin
  * add null terminator to sdes cname item
  * ICE candidate parser - ICE foundation parsing too short
  * Add local volume to janus.js as well (Chrome only)
  * Clear the bitrate interval in the EchoTest demo, if set
  * Fixed typos and missing unlocks in some error cases (see #1317)
  * Added way to notify Lua/Duktape scripts about slowLink events
  * Implement incomingData in echotest.lua demo
  * release sessions_mutex in case of the error
  * Configurable helper threads for Streaming plugin RTP mountpoints
  * Fixed leak in Duktape plugin
  * Fixed RTCP support in Streaming plugin when latching hasn't happened yet
  * Added age to outgoing packets, to avoid sending too old ones in case of issues
  * Don't lock calls to transport_gone in the WebSockets transport (see #1263)
  * Fixed leak in WebSocket transport (fixes #1302)
  * Fixed missing lock in HTTP transport plugin (fixes #1309)
  * Disable Lua and Duktape plugins by default, when configuring
  * Fixed segmentation fault in pp-h264
  * Support screensharing with externally-set constraints
  * Add support for H.264 simulcasting
  * Aligned to new 0.4.1 version
  * Fixed legacy config usage in Streaming plugin after merge
  * Updated Lua sample config file to use libconfig too
  * Fixed syntax error in sample config for Streaming plugin
  * Fixed segfault when .jcfg file is missing
  * Removed example added by mistake
  * Made some array/list-related fixes
  * New configuration file format (using libconfig)

-------------------------------------------------------------------
Fri Jul 13 13:15:19 UTC 2018 - opensuse-packaging@opensuse.org

- Removed the patch to disable RTCP BYE interception
- Added build dependency on lua
- Update to version 0.4.3+git20180711.ca756d17:
  * Added more properties when listing rooms in VideoRoom
  * Fixed documentation references
  * Fixed a couple of leaks in the VideoCall plugin
  * Fixed broken check in TURN REST API code (fixes #1298)
  * Fixed RTCP support in Streaming plugin
  * Updated Streaming plugin configuration
  * Fixed occasional segfault in VideoCall when call fails
  * Handle UDP/DTLS/SCTP negotiations from Firefox Nightly (fixes #1294)
  * Fixed warnings in Lua and Duktape plugins
  * Fixed compilation error with older versions of libwebsockets
  * Small tweaks to just merged #1279 code
  * Added quick helper method to allow plugins to get RR info from RTCP packets
  * per lminiero, revert REMB and RR management of relayed RTCP packets
  * Reset videocall session->incall flag when doing hangup_media.
  * Removed broken call to non-exosting callback in janus.js (fixes #1290)
  * fix sdp find g722 codec bug
  * Fix exception when requesting screen sharing in Chrome
  * throttle RTCP PLI and RR according to the minimum intervals specified in milliseconds in the config for the source. only send the lowest REMB and highest RR loss fraction of all associated listeners to the source.
  * Removed outdated comments
  * Added option to programmatically ask a VideoRoom publisher for a keyframe
  * Improved libwebsockets logging configuration and rendering
  * per lminiero, prefix new relay thread log output with mountpoint name
  * Added flag to post-processor to ignore a number of initial packets
  * Fixe silly typo
  * Updated Janus logo
  * Make sure we only call g_main_loop_quit once (see #1276)
  * Several small Doxygen related fixes
  * Fixed compilation issue on CentOS (see #1283)
  * Bumbed to version 0.4.3
  * Record media added in a VideoRoom publisher renegotiation, if recording (fixes #1281)
  * Don't close WebSocket connections when there's a session timeout (see #1274)
  * save RTCP ports to config when creating/editing mountpoints with permanent=true
  * oops, use the specified RTCP port instead of always using RTP port+1
  * rtcp is spelled RTCP
  * Added new bitrate_cap property when creating VideoRoom rooms (see #1277)
  * Check if iceloop is running before quitting.
  * Fixed some small nits (still related to #1275)
  * Made small fixes to FEC code (see #1275)
  * Fix useinbandfec detection and probation
  * Reset RTP sequence numbers when resuming a long paused subscriber (see #1273)
  * Small tweaks derived from changes in the Lua plugin
  * Make sure the recipient is valid, before accessing it
  * removed offending whitespace
  * per lminiero, reconfigure configure for libsystemd so it is no longer linked and used by default
  * applied patch
  * FEC enabling from SDP negociation
  * Added option to enable lock debugging via janus.cfg or command line
  * init pipefds to -1 in case pipe() fails, call pipe() for rtsp sources as well as rtp sources
  * New Nanomsg transport plugin
  * allow the streaming plugin to relay rtcp packets to and from rtsp servers
  * allow systemd to manage the janus unix domain socket interface
  * Automatically forward incoming PLIs to the media source in the Lua plugin
  * Fixed leak when sending message in MQTT transport plugin
  * Fixed a couple of leaks in the MQTT transport plugin
  * Reduced verbosity in Lua plugin
  * Fixed missing PeerConnection close for VideoRoom subscribers, if configured
  * Change of some AudioBridge properties to atomic checks (see #1266)
  * Removed duplicate code in AudioBridge
  * Allow longpoll and keepalive timers to be configured in janus.js (see #1265)
  * New media plugin to drive logic with JavaScript (via Duktape)
  * Make sure data messages are recorded even if the Lua script implements incomingData
  * Fixed typo in Lua SDP code
  * Properly set subscriber room ID in VideoRoom
  * Additional check on publisher property before accessing it (see #1261)
  * Fixed typos in documentation
  * Removed invocation of close_pc from ad-hoc thread, and added end_session proxy (Lua plugin)
  * Include "openssl/ssl.h" before "openssl/srtp.h".
  * Manually quit the loop when hanging up and ICE never succeeded (fixes #1259)
  * Bumped to 0.4.2
  * Added missing ifdef when pushing SCTP data (fixes #1258)
  * Fixed memory leaks in event handlers (spotted by @zgjzzhw in #1256)
  * Send final stats to event handlers before hanging up
  * On detach, hangup before invalidating the session (fixes #1172)
  * Make sure plugins are configured to enable events, before pushing them
  * Make sure plugins are configured to enable events, before pushing them
  * Fixed occasional broken enforcement of API secret
  * Use defaultDependencies according to documentation.
  * Fixed 'unused variable' warning when libcurl is not used by Streaming plugin
  * Fixed typo in janus.js (see #1252)
  * Address feedback.
  * Remove SCTP thread.
  * Only use "SRTP_MAX_TAG_LEN" bytes for padding.
  * Unlock videoroom in case of errors.
  * Remove duplicate locking of videoroom.
  * Fixed occasional crash in SIPre plugin
  * Fix additional size for packets that will be protected by libsrtp.
  * Remove duplicated code.
  * Bumped to version 0.4.1
  * Removed unused portion of code
  * Streamlines how the plugins docs are organized
  * Revised the way lock are used when joining a VideoRoom as a publisher (see #1239)
  * Use atomic API to get/set flags.
  * Removed unneeded check in janus.js
  * Take into account overridable extension property when extending dependencies
  * Better management of peers in VideoCall code
  * expose janus.js extension config via dependency API
  * Join SCTP thread before releasing resources.
  * Clarified in the docs that for renegotiations and ICE restart you have to check plugin docs as well (see #1238)
  * Clarified item in Streaming plugin documentation
  * Some recorder related fixes, cleanup and removed redundancy
  * Use JSEP trickle parameter when handling answers
  * Fixed broken check introduced in previous commit (custom janus.js dependencies)
  * Check again for queue before pushing packet and add comment about race condition.
  * Removed silly leftover
  * Fix style.
  * Fix the way janus.js dependencies are initialized (by @phsultan, see #1229)
  * Free any queued packets before the queue is destroyed.
  * Only build package to relay if queue exists.
  * Keep ICE component alive while SCTP thread is running.
  * Fixed some audio references (by @tugtugtug, see #1234)
  * Use monotonic time for skew compensation.
  * Fixed ugly typo that caused video delays when starting audio-only
  * Only use a single thread per-PeerConnection instead of two
  * Added config option to set HTTPS ciphers (see #1219, original contribution by @agclark81
  * Fixed occasional race condition when initializing event handlers
  * Added new event handler type, for external content pushed via Admin API (useful for piping info collected via scripts)
  * add space and newline
  * Also only log outgoing SCTP bodies at >= HUGE
  * Fixed info on bound mountpoint ports when choice is left to the plugin
  * fixed codestyle error and add lock back around with janus_videoroom_notify_participants
  * Fixed formatting of time in text2pcap code
  * Removed unused internal SSRCs (VideoRoom)
  * Make sure RTP forwarders get what the publisher sent, if not overwriting the header
  * add helper func to reduce videoroom code > janus_videoroom_codecstr > janus_videoroom_reqfir
  * Fix Streaming plugin compile errors in case libcurl is not installed
  * Added SRTP profile info to Admin API handle details
  * Removed unneeded checks on core reference in Record&Play plugin
  * Removed leftover unlock in VideoRoom plugin
  * Fixed double unlock and a couple of nits/typos in Streaming plugin
  * Fixed double unlock in AudioBridge plugin
  * Added missing unlock in error condition in SCTP code
  * Added missing unlock in error condition in Record&Play plugin
  * Make sure an SRTP profile is negotiated via DTLS (see #1223)
  * Fixed re-initialization of simulcast properties at every configure request in Streaming plugin (see #1220)
  * Added a warning when the DTLS BIO filter is passed a negative size (see #1213)
  * Initialize unused RTCP ports (only needed for RTSP) to avoid closing the wrong file descriptors (see #1221)
  * Remove unused "srtp_mutex".
  * Clear flag if datachannel is rejected by setting the port to 0.
  * Only log SCTP message bodies at >= HUGE level
  * Lower default for DTLS starting MTU
  * Updated README
  * Add host and port in audiobridge rtp forward success response.
  * Implemented latching for RTSP streams (in case Janus is NATted)
  * Re-adapt code
  * Fixed missing srtpsuite/srtpcrypto properties when saving new mountpoints permanently (fixes #1217)
  * Fixed leak in HTTP transport when grouping long poll events
  * Fixed typo in VideoRoom docs (copy/paste leftover)
  * Make sure the size is not negative in the DTLS BIO filter (see #1213)
  * Indentation update
  * Opus FEC implementation
  * Added protocol family to selected-pair handler event
  * Added reason to 'hangup' handler events, and prevented multiple handler events for the same selected-pair trigger
  * More specific info in selected-pair event (event handlers)
  * added FullHD, 4K streams support for getUserMedia via 'fhdres' and '4kres' presets
  * janus.c: increase plugin session ref count while teardown callbacks are live
  * Fixed regression in janus.js when used in Chrome
  * Use transceivers for Firefox >= 59
  * Fixed extra m-lines when adding sendonly media streams in Firefox
  * Better management of extra (rejected) m-lines of already existing type
  * Fixed typo in janus.js log
  * Fixed missing placeholder for no-video publishers in VideoRoom demo
  * Fixed missing SSRC (and mid) in plugin-originated renegotiation (see #1206)
  * Added missing property in VideoRoom listforwarders response
  * Removed references to non-existing API "stop" request from VideoRoom docs
  * Added new reclaim stuff to man page ad README, and added command line shortcut (-m), see #1199
  * Added missing notifications (API+Event Handlers) on kicks in AudioBridge and VideoRoom
  * Fixed RTT computation
  * Don't send Receiver Reports if we're not receiving anything
  * Fixed incorrect docs of VideoRoom API
  * Fixed typo in AudioBridge logs
  * Updated VideoRoom plugin API documentation
  * Fixed a couple of nits in the AudioBridge API docs
  * Added missing item in AudioBridge API docs
  * Updated TextRoom plugin API documentation
  * Fixe a typo in the AudioBridge API documentation
  * Added missing section in NoSIP API documentation
  * Updated Streaming plugin API documentation
  * Fixed typo in AudioBridge API docs
  * Updated NoSIP API documentation
  * Fixed typo in SIP plugin documentation
  * Updated Record&Play API documentation, and added option to specify target ID when recording
  * Updated AudioBridge API documentation
  * Updated SIP plugin API documentation
  * Updated VideoCall API documentation, and added missing simulcast fields in set request
  * Updated EchoTest API documentation
  * lminiero reconnect patch
  * Increase thresholds for skew detection.
  * Fixed leak when creating static AudioBridge RTP forwarders
  * Fixed JSON leak in Janus core
  * remove unnecessary field in claim response, fix log
  * changed session_over signature. Many changes related to session claims -- hopefully less naive now
  * change back log
  * session_claimed callback
  * s/tranport/transport
  * Fixed deadlock in VideoRoom when notify_joining=true (fixes #1204)
  * remove secret session id
  * fix default value in sample
  * fixes
  * secret session id
  * dont allow claim if no timeout specified
  * add ability to reclaim session
  * Updated date in documentation
  * Bumped version number(s), and disabled refcount debugging by default
  * Fixed a couple of nits (code style)
  * Removed swap (?) file added by mistake in #1064
  * Added missing unlock in error branch (Streaming plugin)
  * Fixed broken indentation
  * removed Chrome/FF versions checks for getUserMedia constraints, use unified format now
  * Adds dynamic properties (PINs, Secrets, etc) for Streaming Plugin
  * Fixed typo introduced in previous commit
  * Allow VideoRoom subscribers not to hangup PeerConnection automatically when publisher goes away
  * Removed overly verbose debug line (see #1196)
  * Fixed Streaming plugin method signature when libcurl is not available
  * fixed warning when used from C++ plugin
  * Added a couple new projects to the resources (docs)
  * Rename participants_mutex, add room_id element to listener, add a lock in janus_videoroom_participant_joining.
  * Allow first RTSP connection to fail, if so configured (see #1186)
  * Minor fixes in accessing videoroom from publisher.
  * Fixed deadlock when destroying AudioBridge rooms
  * Check MHD_OPTION_HTTPS_KEY_PASSWORD support programmatically, not in configure
  * Protect stream in janus_plugin_handle_sdp with handle mutex.
  * Updated DTMF related code to use new specs
  * Fixed crash when creating AudioBridge room
  * Update README.md
  * Don't wait for candidates-done if frull-trickle is enabled
  * Fixed missing cleanup in Makefile (fixes #1184)
  * Some more fixes on small nits and typos (static analysis)
  * Fixed small nits and typos (static analysis)
  * Make sure libmicrohttpd is at least 0.9.40 (needed for MHD_OPTION_HTTPS_KEY_PASSWORD)
  * Add a cap to rooms created in the screensharing demo
  * Added new project to the resources in the docs
  * optimised an if/else case when media.video is an object
  * move getUserMedia constraints for Chrome >= 59 to ideal
  * added support of getUserMedia constraints for Chrome >= 59
  * removed pre Firefox38 support
  * removed pre Firefox38 support
  * Allow configuration of passphrase, if needed, when configuring certificates (DTLS, HTTPS, WSS)
  * Resolve names for TURN servers when getting them through API (fixes #1176)
  * Fixed wrong numeric version identifier
  * Make sure lock debug lines end up in the log file, if configured
  * Fixed uncaught exception in janus.js
  * Added content type to fetch requests in janus.js
  * Added possibility to set per-handle token when attaching in janus.js
  * Fixed leak in AudioBridge (missing refcount for participants)
  * Removed extra verbose lines in TextRoom plugin
  * Tweaks to a couple of g_usleep calls
  * Added force_tcp option when registering in SIP plugin (force_udp=FALSE does NOT force TCP)
  * Removed unneeded checks
  * Removed unneeded checks
  * Fixed wrong logs when debugging refcount initializations
  * Fixed memory leak in AudioBridge, and added indication on who has a valid PeerConnection
  * String manipulation fixes and cleanup
  * expose janus_auth_check_signature* via callbacks
  * Add realms to signed token authentication methods
  * preliminary signed-token support for videorooms
  * Fix exit/warning with -A and disabled admin API
  * add documentation for signed token auth
  * add command line switch for token-auth-secret
  * Implement signed token authentication for the API
  * Fix typo in janus.js cleanupWebrtc.
  * Make sure SRTP_AEAD_AES_256_GCM is defined, and disable AES-GCM otherwise
  * Fixed typos in post-processing code
  * Fixed small nits to better adhere to code style
  * Fixed error messages in WebSockets transport plugin
  * Added configure check for breaking change in libre 0.5.7 (fixes #1165)
  * Fix filter stuck at NaN value
  * Fixes after review
  * Fix logs
  * Link quality stats
  * Fixed typo in docs (see #1167)
  * Make sure libsrtp was compiled with AES-GCM support
  * Added different SRTP profiles to SIP, SIPre and NoSIP plugins
  * Removed unneeded stringifier method for SRTP profile
  * Add support for SRTP with AES-GCM
  * Fixed typo (see #1163)
  * Added check to fail if Doxygen is too new (recent versions mess up menus)
  * Add `mountpoint_listeners` to Streaming Plugin
  * Added basic tweaks to RabbitMQ event handler as well
  * Updated post-processor to use new FFmpeg APIs, when available
  * Removed unneeded verbosity when detecting VP9 keyframes
  * Use replaceTrack in Firefox when renegotiating and changing devices
  * fix - videocall plugin data channel recording
  * Removed outdated text from the VideoRoom documentation
  * Fixed command line inconsistencies in README and man file
  * Bumped to version 0.3.1
  * Changed start toggle in demos (see #1158)
  * Added missing va_end() in case of error
  * Request/response mechanism for Event Handlers (via Admin API)
  * Fixed typo in configuration file
  * Override MHD panic function to avoid unnecessary aborts (see #1154)
  * Added detection of RTP duplicates when RFC4588 is involved
  * Fixed typo
  * Fixed broken check when checking if a remote stream has video, in demos
  * Replace fix to janus_videoroom_leave_or_unpublish with a more robust solution.
  * Fix crash on janus_videoroom_leave_or_unpublish occurring when room has already been freed.
  * Fixed typo
  * fixed parse width/height from sps
  * Fixed typo when freeing stream resources
  * Initialize retransmission value for queued packets
  * Add opaque_id to all handle-related events (handlers), and not just 'attached'
  * Add collision attribute to permanent rtp mountpoints.
  * Remove blank lines.
  * Moving average estimation for RTP clock skew compensation. Detection of multiple SSRCs collision onto same streaming mountpoint.
  * Fixed re-adding a media stream after it was removed in a renegotiation
  * Added missing callbacks to SIP demos
  * Fixed a few leaks in the SIP plugin
  * Fixed a couple of leaks in RabbitMQ transport
  * Fixed a couple of leaks in the VideoRoom plugin
  * Fixed leak in SDP parsing
  * Fixed management of removing a m-line when doing VideoRoom renegotiations (see #1151)
  * Initialize missing fields in janus_videoroom_message (not a g_malloc0)
  * Added a notifyDestroyed property to destroy() in janus.js (fixes #1132)
  * Check return error of select in STUN test (fixes #1125)
  * Fixed error in logs when event handlers have not been not enabled (fixes #1128)
  * disabling md5, RC4, SSL 2/3, and compression
  * Fixed typo in janus.js (see #1143)
  * Added a couple of mobile-related projects to the resources (docs)
  * text2pcap: remove redundant fopen()
  * Fixed typos when handling G.711 in SDP
  * text2pcap: include missing errno.h header
  * text2pcap: fix possible memory leak
  * Stop using an unlimited thread pool for all requests
  * Added new command line flag to README
  * Turn RFC4588 support off by default
  * transports/janus_http: fix allocation bug
  * streaming: fix usage of getsockname()
  * Add full-trickle support to Janus local candidates
  * plugins: remove incorrect comment
  * Remove unneeded castings from memory allocations
  * Avoid branching decision between g_malloc() and g_realloc()
  * Remove unneeded allocations checking
  * Fix redundant memory initializations
  * Skip duplicate packets when post-processing
  * Made RFC4588 support configurable (on by default)
  * Don't do RTCP stats on packet if it's a retransmission Also fixed a typo when handling an incoming retransmission
  * Added support for RFC4588 retransmissions (rtx/90000)
  * Fixed SSRC typos when enabling simulcast in Chrome (janus.js)
  * Clarified that, as of now, transport wide CC doesn't work as expected when simulcasting
  * Remove (again) useless NULL pointer checks.
  * Code-style related cleanup
  * Fixed false positive SSL_read:uninitialized error message
  * Fix false negatives in media detection algorithm.
  * Fix unreferences on participant and videoroom in message handler.
  * Added renegotiation support to videoroom.lua (no add streams though)
  * Remove useless checks for pointers before g_free.
  * Fix invalid unreference on publisher in videoroom handler.
  * Check some participant pointers before doing g_free in videoroom.
  * Fixed missing reference when renegotiating (VideoRoom)
  * Optimized SRTP forwarders by sharing SRTP contexts where possible
  * Added SRTP forwarding to AudioBridge plugin too
  * SRTP support in Streaming mountpoints and RTP forwarders
  * Fixed crash when calling 'exists' with no room in videoroom.lua
  * Added a timeout on DTLS handshakes (20s) Added tweaks and fixes to DTLS, ICE and renegotiation management
  * Removed loop initial declarations (fixes #1131)
  * Don't require profile to be baseline when postprocessing H.264
  * Added srtp to brew recipe for MacOS installations (see #898)
  * Use 'pcmu' and 'pcma' in .mjr files, not 'g711'
  * Improved installation instructions for MacOS (see #898)
  * Fixed some nits from #1118, and made default for transport cc FALSE
  * Fix rtcp transport wide fb serialization
  * Fix formatting issues and inline/static
  * Fixes
  * Implemented missing pieces
  * Enable transport wide cc reception per ice session and store <transSeqNum,time> on ice session map
  * Pass transport wide cc usage and ext id to ice session
  * Add transport-wide-cc header negotiation and parsing
  * Simplify internals by always assuming single stream/component
  * Move versioning info from janus.c to in version.h/.c (autogenerated) Added versioning (and commit/build) info to janus-pp-rec too
  * Reset latest FIR/PLI timer when we send a one and at incoming keyframes
  * Close old tracks right away, when replacing them
  * Reduced verbosity of a some warning lines
  * Fixed typo
  * Improved Lua plugin documentation, and allowed Lua scripts to override plugin namespace/versioning info (example in echotest.lua)
  * Fixed anchor conflict in documentation
  * janus_recordplay: fix log error
  * Fixed libsrtp summary in configure (see 1120)
  * janus_voicemail: fix tautology
  * janus_videoroom: fix tautology
  * janus_textroom: fix tautology
  * janus_streaming: fix tautology
  * janus_sipre: fix tautology
  * janus_sip: fix tautology
  * janus_recordplay: fix tautology
  * janus_nosip: fix tautology
  * janus_echotest: fix tautology
  * janus_audiobridge: fix tautology
  * Added missing reference that was causing leaks
  * Fixed typos
  * Updated footer in demos and documentation
  * Accepts all combinations of dir and filename warning the user for unsupported ones.
  * Support 407 response for REGISTER
  * Fix comments style
  * Recursive mkdir when a full path has been passed as recording filename.
  * Removed outdated dependency from docs
  * Simplify internals by always assuming single stream/component
  * Added a warning for when libwebsockets is compiled without ipv6 support
  * Estimate RTT via RTCP (to improve)
  * Sending/receiving keyframes cleans video retransmit buffers/NACK queues
  * Fixed typo
  * Modified version of adapter.js in demos (6.0.3)
  * Fixed link of one of the new videos in the FAQ
  * Added some more video links to the material in the FAQ
  * Fixed some more typos when parsing SDP SSRCs
  * Fixed typo when parsing SDP video SSRC
  * Fixed typo in documentation section name
  * Some more fixes to compilation errors when datachannels are disabled
  * Bumped to version 0.3.0
  * Fixed typos
  * Improved management of incoming NACKs
  * Fixed janus.js issue when adding streams to a datachannel only PC
  * Fixed event sent to handlers related to simulcast video streams
  * Got rid of GList traversal to calculate lastsec bytes
  * Add missing semicolon
  * Close web socket connection when destroying session
  * Fixed some nots and typos
  * Removed unneeded check
  * Add a reference to the session in janus_lua_hangup_media
  * Reduced unneeded verbosity when overwriting SSRCs in RTCP
  * Several RTCP related changes Simplified timing for outgoing RR/SR packets (single trigger) Added RR packets for all simulcast video streams RTCP SSRCs overwritten only for packets originated by plugins Core sets SSRCs manually, when originating RTCP itself Better logging of incoming packets, and cleanup here and there
  * Handle RTCP for all remote SSRCs, including video simulcast
  * Fixed nits and typos
  * Make close_pc and end_session calls truly asynchronous (see #1109)
  * Added janus.js renegotiation controls to the docs
  * Be a bit more tolerant of ICMP errors on RTP in SIP, SIPre and NoSIP plugins (fixes #1095)
  * When postprocessing Opus, flush when closinf the file too
  * Make sure the Opus file is flushed when postprocessing
  * Fixed some typos (static analysis)
  * Updated other demos to handle updates to local and remote streams
  * Only remove old tracks after the new getUserMedia has succeeded
  * Updated stream management in demos to support renegotiations
  * Only show audio and video tracks if there's a local stream
  * Fixed access to non-existing remote stream when removing track
  * Fixed nits in EchoTest and Device selection demos
  * Modified Device selection demo to use renegotiations
  * Added renegotiation options (add/remove/replace audio/video/data) to janus.js
  * Turn recvonly offer to inactive answer, in EchoTest
  * Renamed 'refresh' attributes to somethign more meaningful (update, restart)
  * Improved renegotiation management in VideoCall plugin
  * Added missing return statement in videocall demo
  * Fix not initialized old_rooms in janus_textroom_destroy. Re-evaluate as needed 'now' timestamp in textroom and streaming watchdog.
  * Instruct ESLint to ignore Janus JS module contents when linting your project.
  * Renegotiation on publisher triggers renegotiation on viewers
  * Fixed typo when parsing remote media direction
  * Better management of inactive streams in SDP (and send/recv properties)
  * Keep track of, and enforce, media directions negotiated in SDP
  * Allow datachannels to be added in a renegotiation
  * Added some more references in plugin-to-core messaging
  * Several negotiation related fixesordering of BUNDLE items, stream IDs when ordering of m-lines is different, SRTP setup when audio and video only appears later on
  * Fixed problem of printing non-terminated string when logging in SCTP code
  * Make sure the send thread always quits the ICE loop, even when interrupted
  * Make sure a handle detach doesn't break the libnice loop itself
  * Added some more reference counters during shutdown
  * Fixed typo
  * Generate local values when a new media stream is added
  * Update media flags after a renegotiation
  * Removed extra unneeded debug lines
  * Integrated old ICE restart patch, and fixed SSRC source change support
  * Reduced verbosity when postprocessing VP8/VP9 mjr files
  * Fixed missing unlock in websockets timeout management
  * Add some extra iceloop checks.
  * Renegotiation support
  * Remove unnecessary handle references in ice trickles. Remove waiting for glib loop in janus_ice_webrtc_hangup. Wait in janus_ice_thread for send_thread before calling janus_ice_webrtc_free.
  * Add reference to DTLS while setting up sctp.
  * Fixed WebRTC cleanup never happening due to missing unrefs
  * Fixed deadlock introduced by recent merge
  * Fixed typo (double check)
  * Fixed typo
  * Added missing attributes when saving permanent streaming mountpoints (fixes #1096)
  * Added option to force UDP when registering in the SIP plugin
  * Only list RTSP info when querying mountpoint if libcurl is available
  * Removed extra call to janus_ice_webrtc_free
  * Return more info about mountpoints if the admin secret is provided
  * Added missing properties to permanent save in AudioBridge and VideoRoom
  * Make sure an alert trigger is only enqueued if the send thread exists (see #1083)
  * Aligned to recent mergesin refcount
  * More compact cleanup of retransmit buffer
  * Fixed assertion when accessing non-existing GQueue
  * Removed extra mutex unlock in SIP plugin
  * Restored 'stopping' event to handlers in Streaming plugin
  * Only use g_async_queue_push_front if glib is recent enough
  * Reverted g_async_queue_push_front to g_async_queue_push (which needs glib 2.46 that may be too recent)
  * Fixed small nits (coding style)
  * Set lws count_threads to 1 before creating lws context.
  * Added onended event to track screensharing from UI button in demo
  * Use active plugin sessions map, in place of old sessions map.
  * Fixes #1094
  * Add missing janus_plugin_session_is_alive checks.
  * Remove some debugging logs.
  * Use active sessions lookup in every plugin (excluding videocall). Make a synchronous hangup_media in streaming and recordplay.
  * Fixed typo in Record&Play (caused old recordings not to replay)
  * Handle MSG_EOR in datachannels
  * Fixed include paths of TextRoom plugin
  * Added 'deprecated' warning when using old listener ptype in refcount VideoRoom
  * Log error codes for SCTP-related errors
  * Make sure pending messages are sent before closing a Unix Socket for timeout (fixes #1009)
  * Fixed typo in VideoRoom demo (fixes #1088)
  * Use GQueue instead of GList, when last item is important
  * Allow Streaming viewers to temporarily disable/enable audio/video/data
  * Don't use RTCP BYE as DTLS alerts to close PeerConnections
  * Make sure SDP rid attributes are parsed before ssrc (fixes #1072)
  * Added option to override threshold for detecting timestamp resets
  * Added missing PLI when restoring subscriber's video with configure (fixes #1087)
  * Added missing transaction ID to error (fixes #1084)
  * Fixed check of Jansson version in configure (fixes #1077)
  * Update README for certificates - 2048 bit key in the example - Pointer to letsencrypt.org
  * Update janus_audiobridge.c
  * fixing media_hangup deadlock
  * Add Thread ID callback in lws_protocols[0] to help identifying caller of `lws_callback_on_writable` and as such speed up lws reactivity
  * Add SSL/TLS support for MQTT plugin
  * Experiments to improve performance of the ICE send thread
  * Made tracking functions to debug reference counters defines instead
  * Fixed leak in Streaming plugin (unneeded extra reference)
  * Fixed STUN check at startup in IPv6 network (fixes #1053)
  * Fix missing lua_getglobal in janus_lua_incoming_rtp.
  * Always push 4 args in janus_lua_handle_message.
  * Fix nargs in janus_lua_handle_message and other minor changes.
  * Fix Lua stack overflow. Optimize Lua stack usage.
  * Aligned to new signature for destroying recorders
  * Added refcount to recorder and SDP utils too (plus fix to Streaming crash)
  * Better cleanup of Lua plugin media recipients when a session is destroyed
  * Fix room leak in videoroom plugin.
  * Fixed leak when destroying ICE handle (packet queue)
  * Fixed leak in Lua plugin
  * Removed unnecessary pcall wrapper when resuming coroutines
  * Added missing local statemement in Lua SDP helper utilities too
  * Added 'local' statement to variables that needed to be, in Lua sample scripts
  * Use ipairs instead of pairs when iterating on pending coroutines
  * Fixed typo when parsing Lua plugin config
  * Added checks on validity of message to parse in Lua
  * Added hack in videoroom.lua to fix the empty-array-to-object JSON encoding
  * Added example of notifyEvent to trigger event handlers in Lua
  * Added mechanism for timing asynchronous callbacks from Lua
  * Fixed broken room bitrate/PLI interval enforcement
  * Fixed memory leak when handling Lua plugin messages
  * Fixed broken cleanup of participants list
  * Fixed missing check on valid session
  * Make sure a leaving user is removed from the list of participants
  * Fixed definitions of errors in videoroom.lua
  * Fixed typo in Lua plugin
  * Make sure references are increased before starting threads
  * Added reference counters to new SIPre and NoSIP plugins
  * Additional check before iterating on counters hashtable on shutdown
  * Renamed extern variables in Lua plugin to avoid conflicts with core
  * Make sure that a PLI is sent when starting a video recording from a Lua script
  * Made extern some more stuff as well (Lua plugin)
  * Moved lua state and mutex definition to janus_lua_data.h
  * Added a generateOffer method to the Lua SDP utilities
  * Externalized Lua session definition, to make it accessible from the extra hooks
  * Added missing janus_recorder_save_frame calls to the Lua plugin
  * Fixed validation of number of arguments for Lua's startRecording
  * Removed extra line added by mistake in VideoRoom demo
  * Added support for configuration files to the Lua scripts
  * Changed startRecording API to allow passing a folder, and fixed a typo in videoroom.lua
  * Many fixes in the Lua plugin Moved close_pc calls to dedicated threads to avoid deadlocks Cleanup of recipients (and references) on hangup_media in C code too Locking of Lua state at shutdown to avoid race conditions
  * Check both lua and lua5.3 with pkg-config, due to distros differences
  * Use janus_process_error in order to avoid (char *) casting.
  * Added references to Lua to both README and documentation
  * Fixed some leaks, and made pushing of events+SDP asynchronous
  * Notify hangup to users only when janus_ice_webrtc_free is called.
  * Cast away const qualifier from a char ptr to avoid compiler warnings.
  * Revert send_thread ptr check in janus_ice_webrtc_hangup.
  * Modified videoroom.lua to start using the Lua SDP helper
  * Fixed a couple of occasional crashes in the Streaming plugin
  * New media plugin to drive logic with Lua
  * Add error symbol for wrong WebRTC status.
  * Reject jsep and trickle while cleaning a previous session.
  * Remove send_thread ptr check in janus_ice_webrtc_hangup.
  * Check if ice send thread is ended before freeing session. Front push dtls alert message in case of hangup.
  * Lock sessions in janus streaming plugin.
  * Use session table in videoroom plugin to avoid using invalid sessions.
  * Quit the main loop in janus_ice_handles_check in case the loop is still running.
  * Wait for handle icethread end before scheduling handle destruction.
  * Remove sleep in janus_ice_thread after loop unlocking.
  * Check iceloop ptr.
  * Move iceloop running check from janus_ice_webrtc_free to janus_ice_handles_check.
  * Improve iceloop lifecycle management.
  * Fixed extra spaces
  * Make sure we don't show the counter at the end unless REFCOUNT_DEBUG is set
  * Fixed crash in VoiceMail
  * Fixed crash in VideoRoom, introduced after a recent merge (see #763)
  * Fixed typo in AudioBridge code
  * Fixed typo
  * Check all mallocs for failures (see #756)
  * For rtcp-mux, only get rid of RTCP component when it exists
  * Fixed occasional crash when destroying videorooms (see #736)
  * Fixed crash in VideoRoom when reporting events
  * Rename to janus_create_message
  * Function to create reply with common parameters
  * Handle "add_token" in janus_request_allow_token.
  * Reduce code duplication in janus.c with validation and new functions.
  * Increase idle time for thread pool.
  * Use nodebug variants of session_get_publisher and publisher_dereference in high-traffic functions.
  * Always use g_atomic_int_get when checking room->destroyed.
  * Don't assume that LWS_CALLBACK_ESTABLISHED is the first callback.
  * Put the mutex for websockets into janus_transport_session.
  * Use g_clear_pointer to decrease refcounts. Missing room refcount decrease Dereference videoroom in error block in janus_videoroom_handle_message. Handle race condition for subscribers during janus_videoroom_hangup_media. Protect session publisher with a lock and increase refcount. Simpler cleanup with g_clear_pointer Don't clear msg->handle->plugin_handle in janus_videoroom_message_free. Remove tests for NULL in *_dereference functions. Don't clear publisher->room even after decreasing the refcount. Include pointer addresses in some log messages.
  * Added MQTT transport to the refcount branch as well
  * Fixed indentation
  * Handle LWS_CALLBACK_WSI_DESTROY for AdminWSS, too
  * Handle LWS_CALLBACK_WSI_DESTROY
  * Don't use references for RTP forwarders
  * Reset participant pointer in session when it has been destroyed
  * Fixed typo introduced by #634
  * Increase publisher refcount before unlocking videoroom->mutex.
  * Fixed typo introduced by merge with master
  * TextRoom participants members via refcount
  * More compact checks (as in #628)
  * Make destroy and dereference functions more compact.
  * Decrease refcount via janus_ice_handle_dereference.
  * Maintain refcount for members of janus_videoroom.participants.
  * Functions for manipulating ice_handles in janus_session
  * Fixed deadlocks when destroying handle (and made refcount debugging enabled by default, while testing the PR)
  * Removed outdated log line
  * Moved janus_mutex_lock(&session->mutex) outside of janus_ice_handle_destroy.
  * Reset plugin_handle pointer to NULL after a destroy_session
  * Check if handle has been destroyed before passing stuff to plugins
  * Added missing session lock and expanded another lock.
  * Unref handle before session when API message has been processed
  * Attempt to fix occasional race condition when detaching handle
  * Include refcount.h in files that use janus_refcount.
  * Interrupt poll when destroying RTP mountpoint (faster cleanup)
  * Add explicit refcount.h include to ice.h
  * Less verbosity
  * Fix broken merge of SIP-SRTP branch
  * A few enhancements to VideoCall plugin, and nit-fixing in SIP plugin
  * Fixed a couple of references not being removed in the videoroom plugin
  * Restored g_thread_unref in threads we don't join explicitly
  * Restored g_thread_unref in threads we don't join explicitly
  * Some more rounds at fixing/improving the reference counters branch
  * Changes to the modular transports to integrate reference counters support New janus_transport_session struct to bridge core and transports Fixed some bugs and typos from previous commits
  * A couple of web demos changes to reflect the big RC commit
  * First take at refactoring Janus to use reference counters
  * Rename the free refcount init argument
  * Added reference counter implementation (integration in code still WIP)

-------------------------------------------------------------------
Fri Jul 13 12:04:53 UTC 2018 - ancor@suse.com

- Fixed pkg-config ambiguity

-------------------------------------------------------------------
Sun Nov 19 23:33:29 UTC 2017 - opensuse-packaging@opensuse.org

- Added patch to disable RTCP BYE interception
  * See https://groups.google.com/d/topic/meetecho-janus/4XtfbYB7Jvc/discussion
- Update to version 0.2.6+git20171118.96b8669:
  * Bower: update webrtc-adapter to 5.0.1
  * Fix SSL library detection
  * Add openssl pre 1.1 api macro
  * Don't use es6 specific code

-------------------------------------------------------------------
Wed Nov 15 21:58:43 UTC 2017 - opensuse-packaging@opensuse.org

- Update to version 0.2.6+git20171113.ff8e65e:
  * set Access-Control-Max-Age header
  * Several new helper methods for SDP utilities
  * New Janus plugin, NoSIP, for legacy interop without touching signalling
  * Removed info string, cleaned up a bit, and refactored code to limit reduncancy
  * Modified NoSIP plugin code to use #796 and #804 (RTP context and SRTP stuff)
  * Made incoming_rtcp handler more compact too
  * Integrated updated IP utilities in NoSIP plugin
  * Deallocate the local_ip when getting rid of the plugin
  * First commit of SIPre plugin placeholder (WIP, still broken)
  * Moved IP self-detect of NoSIP plugin outside of the config parse code
  * Aligned SIPre plugin to IP utils
  * Use mqueue to make sure libre calls are done on the loop thread
  * Updated code to reflect latest feedback
  * Initialize libre in the loop thread
  * Working REGISTER
  * Working outgoing INVITE (incoming calls, BYEs, etc. still WIP)
  * Refactored message queue payload, and (almost) working incoming calls
  * Removed unneeded memset
  * Fixed endless retransmissions on incoming calls
  * Better management of BYEs and call cleanup
  * Added audio output device selection, if available, to devicetest demo (fixes #869)
  * Detect RTCP BYE messages and hangup in case
  * Attempt to provide an issue template as a guideline for issue reports
  * Moved CONTRIBUTING.md to subfolder
  * Fixed typo when talking about Opus complexity in AudioBridge
  * Fix: broken check for whether or not data channels were actually requested.
  * fix TextRoom name in launching handler error
  * Fixed typo in HTTP plugin (fixes #882)
  * Added new project to the resources documentation page
  * Made username mandatory when registering, guest or not (fixes #885)
  * Integrated fix made for #885 in SIPre plugin as well
  * New configure options to disable all plugins
  * Fix on nasty race condition (SRTP contexts for outgoing traffic)
  * Removed broken warning on lack of VP9 support in postprocessor (see #878)
  * change packet queueing log level
  * update webrtc-adapter 3.1.5 -> 3.4.3
  * use ontrack instead of onaddstream
  * Aligned to configure-related changes in master
  * Aligned to configure-related changes in master
  * Make sure the agent for a handle is not created twice
  * Aligned to changes to janus.js (intrack)
  * Allow custom headers in REGISTER too, in SIP plugin
  * Added support to custom headers in REGISTER and INVITE in SIPre plugin
  * listDevices support for custom GUM parameters
  * Reverted ontrack/onaddstream replacement, as it broke Firefox (see #894)
  * Changed some TODOs in FIXMEs
  * updating indents to use tabs instead of spaces
  * prevent unintended recursion
  * Fix sdp parsing for audio in streaming plugin
  * Small RTCP fixes
  * Better management of hangup conditions
  * Fixed extra spaces
  * Parse (and notify) display name when receiving incoming INVITEs
  * Moved stack to per-user property (but still a single thread and queue)
  * Fixed crash when shutting Janus down
  * Implemented DTMF via SIP INFO
  * Send an event back when a DTMF has been sent via SIP INFO as requested
  * Send SR way less frequently
  * Implemented hold/unhold (still WIP)
  * Made expires in REGISTER configurable/overridable
  * Temporarily disabled HA1 REGISTER option in SIPre demo
  * First attempts at getting re-INVITES to work in SIPre plugin
  * Talking / stopped talking events, instead of talking-only repeats
  * fix for unititialized event handler close
  * Allow handles to force BUNDLE/rtcp-mux via API without waiting for negotiation
  * Fixed broken re-INVITE support in SIPre plugin
  * Send incoming/outgoing SIP messages to event handlers when using SIPre plugin, if sip_set_trace is available in libre
  * Cleanup of log verbosity in SIPre plugin
  * Fixed 486 response when we're busy in another call (SIPre plugin)
  * Added audio to screen sharing
  * bower: external webrtc-adapter instead of bundled one
  * FIX on adding stream. Chat recordings were messed up.
  * Fixed check in janus.js that was breaking the new Safari WebRTC support
  * Unmute screensharing video, when acting as a viewer (see #915)
  * Bumbed to version 0.2.4
  * Added the possibility to specify an outbound proxy in the SIP plugin
  * Some REGISTER-related fixes in the SIP plugin, and improved SIP demo UI/options
  * More meaningful errors in some demos when trying to join a non-existing room
  * Fixed UI for the table in the demos page
  * Update remove video resolution in demos dynamically
  * First integration of VP9 SVC support in the VideoRoom plugin
  * Removed excessive verbosity
  * Allow binding to 0.0.0.0 in HTTP transport plugin (fixes #923, see #924)
  * Use the authuser part when registering in the SIP plugin, if provided
  * Better verbosity when trying to do a REGISTER in the SIP plugin
  * Fixed typos in SIP plugin docs
  * Aligned SIPre plugin to recent SIP plugin features
  * Add a DNS client when allocating a SIPre stack
  * Made event handlers media statistics timer configurable (was one per second before)
  * Clarified in the docs that janus.js depends on the webrtc-adapter shim (see #926)
  * Give ICE 5 seconds before considering failed a definitive state
  * Allow prompting for server/secret in admin.html (prompt for secret by default)
  * Added support for outbound proxies to the SIPre plugin
  * Added support for offerless INVITEs to the SIP plugin
  * Make sure an 'accepted' event has the transaction ID of the 'accept' that originated it, when doing offerless INVITEs
  * Fixed broken authentication when sending INVITE in SIP plugin
  * Small fixes to devices test demo
  * Fixed some getStats related stuff for Firefox and Safari
  * Better management of POLLERR errors in NoSIP plugin
  * Added support for offerless INVITEs to the SIPre plugin
  * First experiments with VP8 simulcasting (Chrome and Firefox, WIP)
  * Removed duplicate sctmap attribute from SDP when negotiating data channels
  * Bugfix missing variable timePassed
  * Fixed leftover code in janus[.nojquery].js
  * Added support for INFO and MESSAGE to the SIP plugin
  * Added configurable bundlePolicy setting to janus[.nojquery].js
  * Added support for SIP INFO to the SIPre plugin
  * Implemented early media (183) support in the SIP plugin
  * fix for ES5 browsers
  * Fixed rare race condition at startup in TextRoom plugin (fixes #935)
  * fix for ES5 browsers
  * Implemented early media (183 Session Progress) in SIPre plugin
  * Cleaned up some currently unused features
  * Protect access to SIP plugin hashtables
  * Use tracks instead of streams in janus.js
  * Don't set default bundlePolicy to max-bundle (seems to break video on Firefox 56)
  * Allow to override build date
  * Fixed a couple of typos, and improved a couple of methods
  * Fixed issue with usernames containing a semicolon in Sofia SIP (needs patched library)
  * Initial experiments with VP8 temporal scalability
  * Experimenting with VP8 simulcasting in VideoRoom plugin as well
  * Fixed typo, and enriched VideoRoom specific Admin API info when simulcasting
  * Fixed rare crash when looping on interfaces in HTTP transport (fixes #945)
  * Updated reference to webrtc-adapter (4.1.1)
  * Make sure only registered users can place a call in the SIP plugin
  * Fixed missing capping of REMB feedback if it was using the audio SSRC
  * Fixed SDP munging when simulcasting
  * Experiments with automatic simulcast substream changes (EchoTest)
  * Fixed occasional crash when fixing SDP attributes
  * Reorganized simulcast-related JS code to make integration cleaner
  * Reorganized how we munge SDP when simulcasting as a method, and added to createAnswer (but not for Chrome, it breaks video there)
  * Added simulcasting support to VideoCall plugin
  * Some fixes and enhancements in VideoRoom simulcasting
  * Added placeholder code for simulcasting in RecordPlay and SIP plugins
  * Don't enable simulcasting if the negotiated codec is not VP8
  * Version bump for plugins
  * Added simulcast support to RTP live mountpoints in the Streaming plugin
  * Extended RTP forwarding in VideoRoom plugin to support simulcasting
  * Updated plugin to send events at layer changes, and demo to use those (as VP8 simulcasting stuff does)
  * Fixed getBitrate calls for older versions of Chrome
  * Fixed broken guests in SIP plugin, and added way for them to send authenticated INVITEs
  * rtp: simplify janus_rtp_switching_context_reset
  * Fixed typo and enriched plugin specific queries in SIP plugin
  * Added libresls support
  * Add crypt_lib = "LibreSSL" to dtls.c if LibreSSL is defined.
  * Fixed end-of-candidates issue with Edge (see https://github.com/webrtc/adapter/issues/605)
  * Change lws_get_peer_addresses with lws_get_peer_simple to avoid slow reverse-DNS requests.
  * Check for lws_get_peer_simple availability and fallback to libwebsockets_get_peer_addresses eventually.
  * Fix macros checking.
  * Add the end-of-candidates SDP attribute on the server side, not in JavaScript
  * Better lock management when processing requests in plugins
  * Added LibreSSL detection in configure.ac
  * Added local and remote candidates to event handlers (fixes #959)
  * Made a few changes to how a SIPre session is destroyed
  * Added support for authenticated INVITEs, in case the user registered as guest
  * Moved removal of session from hashtable in VideoRoom
  * Made several small fixes to address Coverify static analysis
  * Some more post-static analysis fixes
  * Added some checks/fixes already made in master
  * Added some checks/fixes already made in master
  * Made Access-Control-Allow-Origin response configurable in HTTP plugin
  * Moved VP9 SVC parsing to utils, so that they can be used in other plugins if needed
  * Only send a FIR/PLI for new subscribers when the PeerConnection is up
  * Fixed CORS management in HTTP transport plugin, and made it more compact
  * Use a safer and more conservative starting MTU value for DTLS
  * Bumped to version 0.2.5
  * Added link to blog post in the VP9 SVC demo
  * Reduced verbosity of a few warnings
  * Make sure handles are detached immediately when a session is destroyed
  * Honor trickle configuration when using ext. stream
  * Added ability for VideoRoom listeners to not negotiate support for media the publishers are sending
  * Added option for Streaming viewers to subscribe to a subset of a mountpoint media
  * Added a new paper to the Janus-related bibliography
  * Fixed some typos, leaks, and broken checks
  * Added some checks on the availability of an SDP in the SIP plugin
  * Implemented way to edit some room properties dynamically in AudioBridge, VideoRoom and TextRoom (see #939)
  * Make sure handles are removed right away in case of a session timeout (see #963)
  * mutex: add alias for PTHREAD_MUTEX_INITIALIZER
  * plugins: use JANUS_MUTEX_INITIALIZER
  * transports: use JANUS_MUTEX_INITIALIZER
  * turnrest: use JANUS_MUTEX_INITIALIZER
  * Fixed typo in AudioBridge docs
  * Only do NACKs for a specific medium if they were negotiated
  * documentation: add default value of videoroom publishers
  * documentation: fix for a settings name
  * Explicit mode non-interleaved
  * Fixed RTP forwarding not working for video in VideoRoom plugin (fixes #975)
  * Don't offer medium to VideoRoom subscribers if recvonly/inactive (fixes #985)
  * Updated VP9 SVC demo to use tracks, and bumped adapter to 5.0.1
  * Aligned NoSIP branch to latest changes
  * Added configurable ping/pong to WebSockets transport (see #986)
  * Better management of session timeouts and destruction (see #986)
  * fix rtcp delay calculation
  * Only use new WebSockets ping/pong stuff if libwebsockets >= 2.1
  * Removed typo/leftover from stats evaluation in janus.js (fixes #976)
  * Allow rtpmap/fmtp to be overridden when creating RTSP mountpoints (fixes #972)
  * Wrap build date in quotes, when creating version.c (see #990)
  * Fixed copy/paste error (see #972)
  * Fixed static mutex initialization in HTTP transport, as per recent changes
  * Made Record&Play plugin codec-agnostic (was opus/vp8 only before)
  * Added text2pcap support for handles traffic
  * Add PHPClass to Resources
  * Added option to truncate packets when dumping via text2pcap
  * Added printf format to text2pcap dump method, and session/handle info when dumping
  * Added G.722 support to Record&Play and janus-pp-rec
  * Added some more video material to the FAQ on Janus in the docs
  * Added some more video material to the FAQ on Janus in the docs
  * Added index and anchors to the Janus FAQ in the docs
  * Fixed the contact link in the Support HTML page
  * Notify VideoRoom publishers and subscribers about codecs This solves the problem of subscribers getting an SDP they can't process (e.g., Safari with an SDP containing VP8), as when receiving info on a new publisher they can look at the codec and decide to only subscribe to the media streams they support This also helps publishers hide the local webcam display, if it turns out their video stream has been rejected (e.g., Safari publishing H.264 only in a VP8 room)
  * Changed the commented sample values for the WebSockets pingpong (see #988)
  * Make sure there's only one a=end-of-candidates per m-line (fixes #995)
  * Fixed playout of G.711 and G.722 recordings in Record&Play
  * Fixed race condition when starting WebSockets threads and initialized still 0
  * Fix for the SSL+WebSockets crashes (see #913)
  * Changed check for libwebsockets in configure.ac, to make sure vhosts are supported
  * Fixed typo where we sent the wrong info about video codecs in the Videoroom (fixes #1000)
  * Check for iceloop status before quitting the loop.
  * Ignore EINTR for Streaming mountpoints (fixes #1004)
  * Fixed leak in Record&Play, and dead code in VideoRoom
  * Fixed occasional deadlock when shutting down Janus
  * Ignore ontrack event if there's no stream info
  * Append latest glib.supp to janus-valgrind.supp
  * Changed flags to dlopen to load modules
  * Moved websockets ping/pong setup from the vhosts to the main context
  * Fixed flags for NACK management after SDP parsing
  * Fixed broken stats/NACKs for incoming streams (see #999)
  * Decouple `janus.js` from its dependencies.
  * Added documentation updates for decoupled dependencies:
  * Fix a regression in websocket ACL.
  * Move SRTP specific definitions from rtp.h to a new header
  * Don't ignore ICMP errors on RTP packets in the SIP plugin
  * Fixed typo in comment
  * Add support for static configuration of RTP forwarder in audiobridge plugin
  * Mention static RTP forwarder in the documentation
  * Allow configuration of static RTP forwarder ID
  * Fix issue #1030 by introducing a new flag to videoroom for joining events
  * Updated NoSIP plugin to use new rtpsrtp.h header
  * Updated SIPre plugin to use new rtpsrtp.h header
  * User forward declaration of json_t in plugin.h
  * Add documentation to notify_joining
  * Made clearer that the new options added in #1027 are optional
  * Update change based on review and feedback by @lminiero
  * Fixed typo in rtp.c when resetting sequence numbers (should fix #992 and #1032)
  * Add support for building JavaScript modules from html/janus.js.
  * Integrate support for building JavaScript modules within the build system.
  * Document available JavaScript module formats/syntaxes and how to build them.
  * Changed check when enforcing RTP switching header changes
  * Added configurable port range for RTP/RTCP ports in the SIP plugin
  * Added configurable port range for RTP/RTCP ports in the NoSIP plugin
  * Moved NoSIP links in the demos to highlight the different nature of the plugin
  * Fixed broken enforcement of new RTP/RTCP range in NoSIP plugin
  * Clarified that the SIPre demo is not currently as stable as the older Sofia one
  * Added configurable port range for RTP/RTCP ports in the SIPre plugin
  * Fix compilation issue with libavcodec < 55.28
  * Use supplied WebRTC adapter throughout file
  * Fixed a couple of nits in the new js-modules documentation (see #1029)
  * Add js modules to dist target too, if built
  * Added configure flag (--enable-all-js-modules) to enable all JS modules
  * Ugly hack to fix nits (double dashes) in documentation
  * Added minijanus.js project to the resources in the docs
  * Additional changes before iterating on hashtables
  * Bumped to version 0.2.6
  * Fixed typo in SIPre demo and the navigation bar
  * Made EchoTest and VideoCall recordings aware of negotiated codecs
  * Made a few recorder properties atomic
  * Check that icectx is not NULL before invoking g_main_context_wakeup.
  * Removed spurious debugging line from EchoTest
  * Made RTCP BYE management more tolerant, to accomodate older Firefox 52
  * fix libressl call in dtls.c
  * Fixed some checks in the web demos
  * Allow multiple codecs in VideoRoom, and publishers to choose which one
  * Make sure codecs match when switching publishers in a VideoRoom
  * Add optional exchange for RabbitMQ transport
  * RabbitMQ event handler
  * Added info on emitter (server name) to Janus handlers events
  * Added EINTR checks for all poll() calls (fixes pfunix issue at startup)
  * Added way to provide custom PeerConnection constraints (see #1028)
  * Fixed typo in RabbitMQ transport comment
  * Fixes after review
  * Sample config file fixes
  * Improved RTP headers rewriting in case of context switches
  * LWS_MAX_SMP=1 in README LWS build instructions
  * Add Rust plugin wrapper to resources page
  * Updated reference to stable version of libwebsockets in README
  * Streaming plugin UDP multicast socket bind option to `SO_REUSEADDR`
  * brew installation of libcurl is now called just `curl`

-------------------------------------------------------------------
Tue Jun 06 21:36:48 UTC 2017 - opensuse-packaging@opensuse.org

- Update to version 0.2.3+git20170427.d2ec9c5:
  * Check participant->room before using it at janus_videoroom_leave_or_unpublish
  * Check for videoroom listener at janus_videoroom_incoming_rtcp
  * Fixed outdated line in documentation (fixes #730)
  * Fix test pipeline for the streams plugin
  * Fix setting a XHR.status property that has only a getter
  * Fixed duplicate assignment (fixes #735)
  * Fixed exception in videoroom demo JS code
  * Fixed leftover g_free in a couple of transport plugins (should have been json_decref)
  * Fix missed jquery $.ajax to nojquery Janus.ajax
  * Added optional identifier to match VideoRoom subscribers to a participant
  * ACL and kick support in AudioBridge, VideoRoom and TextRoom
  * Removed unused commented lines in janus.js
  * Opaque identifier to contextualise handles
  * Added opaque ID to documentation
  * Added another paper (Jattack) to publications page
  * Deallocate opaque ID when destroying handle
  * Transport-related events
  * Unref events queue when shutting down
  * Added opaqueId to all demos to demonstrate intra-session handles correlation
  * Removed unused property from AudioBridge
  * Return permament/volatile status as a response to create rooms/mountpoints
  * Reconnect sockets to new IP as well
  * Removed redundant attribute in Streaming plugin event
  * Fixed #754, and added error message in case of missing/invalid IP
  * Fixed crashes in VideoCall when event andlers are enabled (fixes #749)
  * Increase lifetime of remote candidates before they're enforced (fixes #738)
  * Allow configuring SSRC when creating RTP forwarders (AudioBridge, VideoRoom)
  * Fixed typo, and clarified doc for AudioBridge
  * Make sure private IDs in VideoRoom are unique (fixes #755)
  * Allow Streaming plugin to relay datachannels, and VideoRoom to forward them
  * Removed verbose debugging text
  * Added aditional function to get dBov audio level so that average can be calculated for X amount of packets. This can be done also, probably, by converting int janus_rtp_header_extension_parse_audio_level from into to void to return both v and level. videoroom incoming_rtp modified to calculate average level of dBov but event to users is not working.
  * Event is working
  * Fixed indentation and changed number of packets to 150 (3s) and acumulated dBov 3000 (~20 average)
  * Updated date in footer
  * Added accept/reject buttons to VideoCall demo
  * Added FOSDEM2017 presentation on Event Handlers to video resources in FAQ
  * Added DevDay Napoli presentation to video resources in FAQ
  * Fix typo in textroom plugin log message for list command
  * Add LWS_SERVER_OPTION_DO_SSL_GLOBAL_INIT for secure websockets if supported (fixes #768)
  * Allow some TextRoom commands to be sent via Janus API
  * Added withCredentials support to XHR requests in janus[.nojquery].js (fixes #742)
  * Updated janus.js documentation
  * Reply with created/destroyed when requests come from Janus API (fixes #765)
  * Changed default MAX nack queue to 300ms instead of 1 second
  * Configurable timeout for the 'not receiving audio/video' events
  * Updated admin.js and documentation
  * Configurable session timeout value
  * Option to add temporary extension while recording
  * Added aditional function to get dBov audio level so that average can be calculated for X amount of packets. This can be done also, probably, by converting int janus_rtp_header_extension_parse_audio_level from into to void to return both v and level. videoroom incoming_rtp modified to calculate average level of dBov but event to users is not working.
  * Event is working
  * Fixed indentation and changed number of packets to 150 (3s) and acumulated dBov 3000 (~20 average)
  * * Proper use of level var of dBov * Proper notification of participants TODO event notification
  * * removed reference to old method
  * added event notification
  * Don't use mountpoint property of session directly (see #777)
  * Reference third party js/css files externally (see #778)
  * Added license header to adapter.js (fixes #781)
  * Fixed broken link to css
  * Several new helper methods for SDP utilities
  * Fixed relative paths to navbar.html and footer.html in docs placeholder
  * Externalized adapter.js and removed automatic loading of jquery/adapter from janus.js
  * add custom libnice install instructions to README
  * Allow VideoRoom publishers to force the plugin to drop their data messages
  * Allow updating display value via configure command
  * include display in parameter validation, get rid of extra if statement
  * Allow websockets server to bind to IP instead of certificate name (fixes #772)
  * Print when we're using BoringSSL (and turned some related warnings in infos)
  * remove recommendation to install newer version of libnice
  * enhancements to BoringSSL handling in autoconf
  * fix path typo in README
  * free old display, make setting new display more compact
  * Cleaned up the log notification about the crypto lib in use
  * Changed a few warnings to debug messages in janus[.nojquery].js (fixes #791)
  * add janus-event-server to resources page Simple plugin-based server to receive/process events from Janus
  * Add utility functions to map a network device name or IP address to a network interface. Introduce the janus_network_query_config object type to describe how a 'matching' network interface should be determined.
  * Permit user configurable network device selection for listening to multicast RTP and RTSP streams in Janus. Since multicast addresses only make sense in the context of specific network interfaces, this change enables Janus to stream RTP/RTSP received on abitrary interfaces reliably to its WebRTC clients/peers.
  * Support the datasctpnetwork configuration option for RTP streams. The new options may be used to restrict Janus to a specific interface (IP) when listening to a data channel (SCTP). The value type/syntax of the option is: either a device name or its IP address.
  * New iceState event in janus[.nojquery].js, and enriched webrtcState event
  * Small changes to #776, and added related doc info to conf file
  * Small fixes to #786, and updated example in AudioBridge docs
  * Made RTP context and rewriting part of the core, rather than plugins
  * Make sure the PeerConnection is valid before invoking the iceState callback
  * configurable screensharing framerate
  * Update janus.js
  * Revert "Update janus.js"
  * made same changes to nojquery
  * tabs not spaces in nojquery
  * Added 'retransmissions' counter to DTLS contexts, available in Admin API and event handlers
  * Better integration of new IP tools in Janus core and plugins
  * Moved most of SRTP-related stuff to rtp.h/.c (cleans dtls and janus_sip)
  * Require libsrtp >= 1.5 (1.4 will be rejected)
  * Reduced verbosity of a couple of debug lines
  * Fixed typos in configure.ac
  * Documented new media.screenshareFrameRate property (see #802)
  * Changed API of janus_network_detect_local_ip to better fit ip-utils, and added wrapper (integrated in janus.c and janus_sip.c) that returns an allocated string
  * Fixed attach and reattach media over iOS devices
  * Updated browser detection condition. Now, we use adapter.browserDetails
  * Updated browser detection condition. Now, we use UserAgent if user navigated from mobile Safari
  * Fixed indentation issue
  * fix typo
  * Further cleanup of ip-utils related code
  * try/catch JSON.parse in janus.nojquery.js (see #807)
  * Switched to version 0.2.3
  * Fixed a couple nits in the README
  * Made the DTLS set-timeout feature more evident in both README and configure
  * Add ICE Lite status to the Janus info
  * Make sure reply is initialised (TextRoom plugin)
  * Make SRTP errors way less spammy (unless debug=7 is used)
  * Integrated SDP utils in Record&Play plugin too
  * Removed leak in AudioBridge plugin
  * Fixes missing directory (full path) when renaming from temp extension
  * Updates Janus manpage to addres recent cmdline argument addition (fixes #812)
  * Install only headers needed by third party components (plugins), and add proper subdirectories for API headers (plugins, transports, events) (fixes #811)
  * Make sure apierror.h is installed with the headers too
  * Removed unused code
  * Fixed, issues around my original commit
  * Fixed typo introduced in #810
  * Make sure config.h is installed with the headers too
  * Stripped ICC profile from arrow image in demos (fixes #815)
  * Make sure cur_seq is a valid pointer
  * If a Streaming recorder fails, return an error
  * Fixed typo when matching user agent (introduced in #808)
  * Return a valid event after an AudioBridge leave (fixes #816)
  * Fixed typos in docs
  * Updated date in docs
  * Fixed broken re-INVITE management in SIP plugin
  * Fixed compilation warning in ice.c (fixes #818)
  * Unlink UnixSockets when shutting down (fixes #819)
  * Added Debian repo to resources
  * Removed usages of /tmp, and used placeholders where relevant (see #814)
  * Username/Password fix
  * Moved IP self-detect of SIP plugin outside of the config parse code
  * Fixed memory leak username/password
  * Fixed crash when disabling non-RTP mountpoints
  * Don't print errors if transports are simply disabled by configuration
  * Fixed broken link in EchoTest demo (fixes #827)
  * Fixed error in HTTP module (reported on group)
  * Clarify whether a room (AudioBridge, VideoRoom, TextRoom) is PIN-protected when answering a list request (fixes #826)
  * Check PIN in TextRoom, if available (fixes #830)
  * Option to add temporary extension while recording
  * Fixed removed data channel code
  * add ssl support for the rabbitmq transport
  * Make sure RTCP buffers are reset before they're written to (fixes #833)
  * Make sure new_head is not NULL before accessing it
  * Configurable parameters for audio level in room
  * Only in case of audio packets
  * Reset participation type after a leave in the VideoRoom
  * open RTCP port for RTSP streams
  * Better management of VideoRoom kick
  * open rtcp port for rtsp streams (change janus_streaming_rtsp_parse_sdp signature)
  * Made janus_streaming_rtsp_parse_sdp static
  * Use g_file_get_contents instead of fopen/fseek/fread, using the former causes AddressSanitizer to report a heap-buffer-overflow
  * bind random ports in rtsp
  * Updated configure.ac requirements: g_clear_pointer was added 2.34 and not 2.32
  * RTSP: bind random RTP port, adjacent RTCP port
  * Added git commit + compile time information to the Janus logs
  * Fixed broken section titles in README
  * Fix leak in janus_sctp_incoming_data, callback used by usrsctp_socket didn't free passed data
  * Fixed fmtp parsing in Streaming plugin for RTSP The SDP session name is now the mountpoint name, in the Streaming plugin
  * Support for on-hold in SIP plugin
  * Audio room event Working from config static rooms Typo in comments
  * Switched from User->Display to user_id
  * Optional forcing of private IDs for subscriptions for better kick in VideoRoom
  * * config file samples removed * fixed indentation * mutex lock/unlock added to both plugins to wrap message dispatch
  * Covered > 0 cases for int values Reseting on leaving
  * Fixed occasional deadlock when kicking
  * Fixed (optional) rabbitmq-c compilation steps in README (see #847)
  * Modified rabbitmq-c instructions to better adhere to official guidelines
  * Reverted instructions (hopefully for the last time)
  * * moved example in sample file to proper place * added 'nulling' on changeroom event
  * Added check if this is not video packet
  * Changed the shape of event
  * Small post-merge fixes to AudioBridge and VideoRoom code
  * Attempt to fix race condition when kicking publishers and their subscriptions
  * Fixed bug where session_timeout=0 config setting was now honored
  * Cleaned up warnings from #760
  * Reconnect RTSP stream if it goes down (Streaming plugin)
  * Fixed generation of version.c when not on a git repo (fixes #860)
  * Log when we managed to reconnect
  * Don't do anything until the RTSP stream is reconnected
  * Fixed deadlock when connecting to an RTSP server at creation time fails
  * Fixed broken AudioBridge/VideoRoom PeerConnections from Firefox Nightly, due to new checks on extmap direction
  * Added versioning details to /info results
  * Allow video/audio port of 0 for RTP streaming, also add 'audio_port' and 'video_port' to find the randomly generated port
  * Fixed broken indentation (spaces)
  * Use MHD_USE_AUTO if libmicrohttpd is recent enough
  * Specify MHD_USE_AUTO_INTERNAL_THREAD or MHD_USE_POLL_INTERNALLY explicitly, when doing MHD_USE_THREAD_PER_CONNECTION
  * Added elixir-janus to the Resources documentation page
  * Reverted MHD_USE_TLS back to MHD_USE_SSL
  * Don't detach handles when you're destroying a session in janus.[nojquery].js
  * fix #871 detaching of already detached plugin
  * mode plugin detached status check before request construction
  * Fixed no_media_timer log line
  * Make sure s-values in SDP are always simple (fixes #874)

-------------------------------------------------------------------
Thu Jan  5 10:52:05 UTC 2017 - ancor@suse.com

- janus-pp-rec manpages added only when needed

-------------------------------------------------------------------
Thu Jan  5 10:04:39 UTC 2017 - ancor@suse.com

- Added manpages to the package

-------------------------------------------------------------------
Wed Jan 04 16:35:01 UTC 2017 - opensuse-packaging@opensuse.org

- Update to version 0.2.2+git20170104.3f4cdb2:
  * Fix getting min/max values in janus_rtp_header_extenstion_parse_playout_delay
  * Updated the obsoleted FAQ items in the documentation
  * Added license exception to explicitly allow linking to OpenSSL (fixes #713)
  * Use real time instead of monotonic time for events in event handlers
  * Fixed truncated error messages in textroom (fixes #720)
  * fix typo thanks to lintian
  * Added manpages for janus and janus-pp-rec (addresses #723)
  * Added events_folder property to janus.cfg (fixes #728)
  * Fixed libcurl-related headers leak (sample event handler, textroom)
  * Fixed events-related leak when handlers are enabled but none's available (should fix #727)

-------------------------------------------------------------------
Tue Dec 20 10:38:11 UTC 2016 - opensuse-packaging@opensuse.org

- Update to version 0.2.2+git20161216.fef96f3:
  * Event handler plugins, first draft
  * Only forward events a handler is subscribed to
  * Allow for the events to be disabled completely (broadcast=no in [events] of janus.cfg)
  * Fixed typo
  * More events, in particular from other plugins than the EchoTest, and added examples to the sample handler plugin
  * Added incoming SIP messages to the events (still missing outgoing)
  * Added outgoing SIP messages to events (to improve/fix)
  * Added queue and thread for actually broadcasting events to handler plugins
  * Added some RTCP and media related statistics to the events, triggered each second
  * Disable event handlers by default; added command line flag to enable them
  * Add display name to joined event in VideoRoom
  * New SDP utilities to replace Sofia SIP SDP stack
  * Made Sofia SIP a dependency for only the SIP plugin, cleaned up configure.ac and Makefile.am, added enumeration for media direction, and used new SDP utils in VideoRoom plugin too
  * Return reason for SDP parsing errors Renamed some methods Optimized some parsing/processing rounds
  * Added plugin configuration for whether or not to shoot plugin-specific events (even when global configuration is yes)
  * Added helper method to remove payload types from SDP
  * Helper method to free an SDP attribute
  * Support session level connection data
  * Converted SIP plugin to use the new SDP utils
  * First take at supporting re-invites/updates in SIP plugin with new SDP utils
  * Removed unneeded sdp_parser property
  * Revert "First take at supporting re-invites/updates in SIP plugin with new SDP utils"
  * First take at supporting re-invites/updates in SIP plugin (uses #578)
  * Set pointers to NULL after a g_list_free
  * Increased size of pollfd array to account for pipe file descriptor
  * Increase plugin API version
  * Fixed merge introduced error
  * Fixed memory leak
  * Style
  * Fix compilation
  * Fix crash if attribute value is empty
  * sip: reply with 488 if offer doesn't contain audio or video
  * Updates to janus.js/adapter.js
  * Removed unneeded pragma
  * Removed unneeded checks before g_free
  * Larger buffer when parsing crypto
  * Added JANUS_SDP_DEFAULT (=JANUS_SDP_SENDRECV)
  * Don't write direction attribute if it's JANUS_SDP_DEFAULT
  * Added fmts list, and fixed datachannels negotiation
  * Notify on log when we skip a JS dependency because it's already loaded
  * Invoke previous onbeforeunload callbacks at page close, if set before ours
  * Use urls instead of url in iceServers
  * Clarified that the sample plugin needs libvurl in the README
  * Added display to all participant-related events
  * Implemented event grouping and HTTP auth+timeout in sample event handler
  * Made basic authentication the only supported method, for now
  * Added new category of events (core)
  * Added Raspberry Pi resources (UV4L) to the docs
  * Fixed typo
  * Allow plugins to send out-of-context events (no associated session/handle) to event handlers
  * Handle LWS_CALLBACK_WSI_DESTROY
  * Use g_ascii_strtoull instead of atol where applicable
  * Fixed indentation
  * Use g_ascii_strtoull instead of atol where applicable (pt.2)
  * Bump version number
  * Assign new value before freeing old value to avoid state with freed value.
  * Remove code duplication between regular and admin web sockets.
  * Added support of MQTT transport
  * Unload and skip plugin if init failed (see discussion in #645)
  * Fixed indentation of MQTT cfg file
  * Fixed wrong casts when closing plugins
  * Fixed typo in documentation
  * Fixed small typos in documentation
  * Fixed small typos in documentation
  * Fixes to get Janus working with Edge again (see #651)
  * Don't drop video support on Edge, leave it to the application
  * Fixed a few leaks
  * typo fix
  * typo fix
  * fix compile failed on Mac
  * add compile on macOS to README
  * update README
  * update README
  * Changed the verbosity of some log messages in WebSockets plugin
  * fix documentation
  * Updated instructions for building libsrtp (1.5.4)
  * Make sure there's always an event to return in HTTP long poll
  * Fix crash in SIP plugin when no remote IP is found for RTP in the SDP
  * Added simple retransmission mechanism to the sample event handler plugin
  * Reset retransmission counter after a success
  * Fixed checks for adapter.js (were broken in Chrome 56)
  * Fix check for auth token support in admin.html/js
  * Added setting to modify own volume (percent) in audiobridge (see #668)
  * Fix SSRCs in RTCP before encrypting and not after, in SIP plugin
  * Added supervisor sample to the documentation
  * Updated resources list
  * update README
  * support for old macOS
  * New debugging level in janus.js (vdebug), a few changes in JS logging, and new slowLink event handler (example in echotest.js), plus updates to documentation
  * Use free instead of g_free for strings allocated by json_dumps (fixes #679)
  * Update README.md
  * Removed extra unlock (see #694)
  * Mention libcurl as optional dependency in the documentation (see #691)
  * Add optional authentication support to RTSP streaming (see issue #692)
  * Edited README guidelines for MacOS (see #666)
  * Fixed error when compiling Streaming plugin without libcurl
  * Made configure smarter (see issue #689)
  * support for setting an iceTransportPolicy
  * Lock forwarder mutex before using forwarder hash table (pull #686)
  * Added support for (some) RTP extensions
  * Added playout-delay to the RTP extensions
  * Automatically try using SIP INFO for DTMF in SIP demo when not on Chrome
  * Fixed typo
  * Implemented timeout/GET_PARAMETER support for RTSP in Streaming plugin
  * Make negotiation of audio-level RTP ext in AudioBridge configurable
  * Make negotiation of new RTP extensions in VideoRoom configurable
  * Fixed --disable-unix-sockets check in configure.ac (fixes #701)
  * Bumbed version number and small fixes to the docs
  * Autodetect libsrtp version (1.5.x vs 2.0.x)
  * Updated code to reflect API changes in case libsrtp2 is detected
  * Shimmed libsrtp2 API
  * Include publisher's internal_audio_ssrc and internal_video_ssrc in plugin_videoroom listparticipants
  * fix MACOS endianness issue (due to lack of standart environment variables) + make janus-pp-rec compile on MACOS
  * Fixed async/sync AJAX request for detach/destroy (fixes #704)
  * Reduced verbosity of a couple of transport related messages
  * Modified RTCP code to recognize XR packets
  * Added failIfNoAudio/failIfNoVideo capture-related flags to janus.js, both default to false (fixes #705)
  * Fixed leak when reporting media-type events to handlers
  * Fixed typo when skipping bytes in post-processing
  * Added support for libsrtp2 to SIP plugin too (fixes #709)
  * Removed leftover linking reference in Makefile.am (see #709)
  * Fixed uncaught typeError for slowLink in janus.ks (fixes issue #710)
  * Handled case of Aggregate Control containing the URL already (RTSP)
  * Added check on target extension when post-processing .mjr files

-------------------------------------------------------------------
Thu Sep 29 05:57:35 UTC 2016 - opensuse-packaging@opensuse.org

- Update to version 0.2.0+git20160928.fa78e35:
  * Reply with sendonly if AudioBrdge peer is recvonly (fixes #629)
  * Make notification on dropped packets less frequent in AudioBridge (see #626)
  * Fixed typo and period check in AudioBridge
  * Better check for SPS in H.264 post-processor, and NAL parse debugging
  * Parse STAP-A packets when processing H.264 recordings (fixes #630)
  * Return an event to publishers leaving
  * Fixed typo (see #620)

-------------------------------------------------------------------
Fri Sep 23 09:31:46 UTC 2016 - opensuse-packaging@opensuse.org

- Update to version 0.2.0+git20160922.38db2c9:
  * Optimization of core-to-plugin communication
  * Make sure the result content is a JSON object
  * Combined result content check
  * Use json_true() and json_false() where we used 0/1 integers or true/false strings
  * Made media event use boolean as well
  * Use janus_process_error_string when error is a complete string
  * Moved NACKs counters/timers to janus_ice_stats (before there was ambiguity on direction), and added new core-level 'slowlink' event
  * Fix new check and local variable setup
  * Fix new check and local variable setup
  * fix processing vp8 with no extended bit
  * fix indents
  * fix indents
  * fix indents again
  * Fixes for 64-bit identifiers
  * Fix for issues #509 and #574
  * RTSP PLAY request URL should not have a slash appended
  * Avoid shadowing dup
  * Fixed leaks and typos in Record&Play plugin
  * Removed references to deprecated lws_get_internal_extensions()
  * Avoid warning when libcurl is not available
  * Don't ignore return value of read
  * Fixed sscanf and format related warning
  * Don't ignore return value of fread
  * Removed unneeded extra verbosity for candidated in janus.js
  * Allow for the configurable recording of the contribution of a single participant (AudioBridge)
  * Explicitly detach libnice data notifiers when hanging up
  * Rudimentary handling of SIP session-refresh (keepalive) added.
  * Fixed a couple of potential leaks in SIP plugin
  * Removed extra/unneeded calls to json_decref
  * Fixed typo (wrong check in admin API)
  * Fixed typo in HTTP transport plugin
  * Added 'exists' request to the textroom plugin
  * add support for 'ack' field in textroom messages
  * ads 'ack' to message parameters
  * fix atom tab -> 2 space issue
  * try char fix
  * try hard tabs
  * try hard tabs
  * reset tabs
  * make ack comment more clear
  * Changed some comments to #584, and fixed leak in TextRoom plugin
  * Fixed typo introduced in #577
  * Reject attempts to start SIP calls with datachannels (fixes #581)
  * Made Record&Play more tolerant with playout (broken files just skipped)
  * Documented use of deviceId (fixes #591)
  * Changed naming of threads, fixed wav header in audiobridge recording, anticipated sessions stuff in Janus startup (to avoid issues when some of the transport plugins drag and requests start arriving)
  * Removed extra verbose line
  * Fixed count of packets in large files when postprocessing
  * Fix for post-processing (timestamp resets + retransmissions)
  * Close socket descriptors on error
  * Address comments
  * Parse end-of-candidates
  * Use `var` keyword before declaring charSet var.
  * Fix the same problem in janus.nojquery.js as well
  * Fix for crashes during shutting down
  * Fix: mountpoints_mutex should be locked
  * Sync the port with the demos
  * Fixes after review
  * Remove condition check
  * Made plugin response more concise (code suggested by @andreasg123)
  * Fixed VideoRoom publish when datachannels are negotiated but not supported
  * Added configuration options to transport plugins to control how JSON output is serialized (default=indented, plain, compact)
  * Don't parse attributes for an m-line not associated to any stream
  * Fixed indentation
  * Increase plugin API version
  * Added JSON serialization options to TextRoom plugin as well (for data channels)
  * Changes to DTLS BIO filter for OpenSSL 1.1.0
  * Fixed a couple of leaks/checks
  * Removed some unneeded extra verbosity
  * Add size of queued packets queue to Admin API info
  * Fixed typo
  * Found a bug in janus_sip.c when the sip stack receives an INVITE without SDP after the inital invite.  In this call flow, Janus was assuming the invite would always have an SDP and would segfault when receving an invite without one.
  * Fixed VideoCall media setup
  * Initialize BIO filter at startup
  * Changing log setting on invite without SIP to LOG_WARN
  * Fixed warnings
  * Fix processing SDPs with value-less attributes
  * Fixed typo in configuration example (streaming plugin)
  * Fixed typos in VideoRoom plugin
  * Implemented RTP forwarding for AudioBridge's mix
  * Conditional check of PIX_FMT_YUV420P availability in pprec (issue #622)
  * Allow port re-use in Streaming mountpoints if it's for multicast (issue #617)
  * Allow AudioBridge RTP forwarder to relay a mix even when the room is empty
  * Have VideoCall plugin close PeerConnections on hangup (issue #616)
  * Show if RTP streaming mountpoint is recording in info request
  * Fix sequence numbers when media is resumed after a configure/false (issue #620)

-------------------------------------------------------------------
Fri Jul 01 15:33:44 UTC 2016 - opensuse-packaging@opensuse.org

- Update to version 0.1.2+git20160701.d57beb5:
  * New transport module (Unix Sockets)
  * Addressed review by @saghul, and added call to transport_gone on disconnection which was missing
  * Added check for SOCK_SEQPACKET in configure.ac
  * Addressed further feedback
  * Define UNIX_MAX_PATH if undefined, and helper method for creating socket
  * Fixed portable definition of UNIX_PATH_MAX
  * Use recvmsg() for incoming messages, and check MSG_TRUNC
  * Check EAGAIN as well when reading
  * Handle POLLERR and POLLHUP in Unix Sockets poll
  * Reset socketpair after a POLLERR
  * Added SOCK_DGRAM support to the Unix Sockets transport module
  * Move initial declaration outside of the loop
  * First take at RTCP SR/RR in core
  * Device selection in janus.js and new demo
  * Fixed broken screensharing
  * Removed unneeded verbosity in listDevices
  * Added RR/SR termination, and filtering of outgoing packets (REMB generation)
  * Minor fixes for coding convention
  * Added missing doc info
  * Fix Sofia SIP when both Record-Route and Contact are there
  * Better management of missing capture devices (see #489)
  * Fixed check in latest commit (see #489)
  * Fix broken VideoCall plugin for recent Chrome versions
  * Handle padding in RTP when postprocessing
  * Fixed typo
  * Fixed typo
  * Fixed typo
  * Integrated capture devices fix in janus.nojquery.js as well
  * Handle media event in janus.js
  * Documented new media event handler in janus.js
  * Support for other codecs and formats in recorder and post-processor
  * Fixed typo (extra debug line causing wrong return)
  * Fixed typo
  * Pass right codec information to the recorder in the SIP plugin
  * Reduced verbosity introduced in latest commit
  * Bump plugin version to force developers to be aware of API changes
  * Handle rec_dir even if record is false in VideoRoom plugin
  * Adjustments to postprocessor logging
  * Adjustments to postprocessor logging
  * Clarified that the license for the janus.js and janus.nojquery.js JavaScript libraries is MIT and not GPLv3
  * Some more examples in the deploy documentation
  * Moved some WS stuff in the deploy documentation
  * janus-pp-rec should always janus_log_destroy() at exit
  * Reject datachannels in AudioBridge plugin, if offered (see #501)
  * emacs.el to set the Janus coding style in Emacs
  * Fixed segfault when processing recordings with old header
  * Fixed VP8 post processing
  * Fixed nits from code review
  * core: use RTLD_LOCAL when loading plugins and transports
  * Better error notification in case of screensharing errors
  * handle NULL error argument to janus_ice_trickle_parse()
  * fix janus build on mac os x, add openssl CFLAGS
  * Added node-janus project to the resources in the docs
  * Fix Janus.isWebrtcSupported
  * Remove redundant whitespaces
  * fixbug postprocessing for opus using DTX
  * Differentiate screen and window sharing in Firefox
  * move early janus_mutex_unlock(&rooms_mutex)
  * fixbug postprocessing for opus using DTX
  * Refactored web pages and demos
  * new JANUS_VALIDATE_JSON_OBJECT macros
  * two tiny fixes for JANUS_VALIDATE_JSON changes
  * Allow configuration of a name for the server instance
  * Proceeding call state added
  * New webrtcState event in JS API to be notified when PC goes up/down (and a few updated demos to use this)
  * Early media for session progress
  * rolled back changes for early media
  * Fixed typos
  * Updated docs (Unix Sockets and Transport API in doxygen)
  * Clarified Unix Sockets support in docs
  * Further cleanup of SDP when stripping for plugin usage (should fix issue #509)
  * new JANUS_CHECK_SECRET() and JANUS_CHECK_PIN() helper macros for plugins
  * use JANUS_VALIDATE_JSON_OBJECT() and related helpers in all plugins
  * consolidate JANUS_CHECK_PIN() into JANUS_CHECK_SECRET()
  * Add new listforwarders request
  * Check out_stats.video_packets when dealing with video.
  * Fixed typo in streaming API validation
  * In videoroom, protect recorders with a mutex to avoid race conditions.
  * mutex and name fixes
  * fix port name
  * In SIP, protect recorders with a mutex to avoid race conditions.
  * Allow configuration of HTTP method to use to contact TURN REST API, if enabled
  * Fixed other typo in streaming API validation
  * Fixed incorrect casting in listforwarders
  * sip: add ability to customize the display name
  * Autodetect media from payload type if SSRC wasn't advertized ('Not audio and not video' warning)
  * Fixed typos
  * Allow websocket transports to only bind to a single interface and not all
  * Don't warn in response to a "detached" event because that situation happens when detaching from JavaScript.
  * Include fcntl.h to fix build error on Alpine Linux
  * Add calls to janus_videoroom_message_free
  * Increase version to 0.1.1, due to recorder changes
  * Increase version to 0.1.1, due to recorder changes
  * Max number queue in seconds instead of packets, plus some other RTCP related tweaks
  * Helper method to create MHD daemon in HTTP transport
  * Allow HTTP transports to only bind to a single interface and not all
  * New mutexes to protect recorders in plugins from race conditions (see #531 and #533)
  * Changed granularity of new Max NACK queue to milliseconds instead of seconds (min is 200ms)
  * Don't buffer packets if max_nack_queue is 0
  * Don't notify about a new publisher until its WebRTC setup has been completed (should avoid issues of people subscribing to ghost publishers that failed to get a working PeerConnection)
  * Reduce code duplication in videoroom plugin with several new functions.
  * New function janus_videoroom_recorder_create. Set the rejected mline at the end.
  * Fixed postprocessor compile error when FFmpeg version doesn't support VP9
  * Fixed old NACK check time
  * Add package.json
  * Add files array to package.json to only install client side scripts
  * Add janus.nojquery.js to files array at the package.json
  * Fixed typo in docs
  * Validate request parameters in janus.c with new macro
  * Remove AudioBridge rooms lazily in the watchdog, to avoid race conditions after a destroy
  * Better management of incoming RR
  * Make naming of new attributes in Admin API less ambiguous
  * janus_videoroom_access_room returns error_cause. New functions janus_videoroom_sdp_a_format, janus_videoroom_sdp_v_format.
  * Fixed broken SS/RR/NACK transmission, due to incorrect filtering
  * Combine log messages for codec mismatch.
  * Support for recording data channel text messages, and post process them to .srt files
  * Improved locking in AudioBridge rooms and participants management
  * Added new plugin (and demo) for datachannel based text broadcasting
  * Fixed broken automatic REMB in VideoRoom
  * Allow binding HTTP transports to a specific IP
  * Make HTTP trasports dual stack, if no interface/IP is specifiec
  * Allow binding WebSockets transports to a specific IP, and fixed some typos
  * Handle larger buffers of text when post-processing
  * Reduce verbosity of processing
  * sip: add ability to override User Agent per account
  * sip: style fixes
  * Initialize variables
  * dtls: simplify key loading code
  * misc: style fixes (editorconfig)
  * dtls: refactor loading certificate and key files
  * dtls: automatically generate a key and cert if they were not specified
  * doc: remove trailing spaces from README
  * doc: command line options -c and -k apply to DTLS only
  * dtls: add warning when autogenerating key/cert
  * Clarified in the docs that the Admin API over WebSockets needs a different subprotocol
  * Started v0.1.2
  * Don't ignore alerts for DataChannel only (or non-muxed) components
  * Added AG Projects' repo to the resources
  * Add time to outgoing messages in TextRoom plugin
  * Remove alphanumeric constraint on username for TextRoom
  * Use MHD_create_response_from_buffer instead of deprecated MHD_create_response_from_data (issue #565)
  * Fixed ACL for HTTP transport (issue #564)
  * Fix detection of lost incoming packets
  * Fixed creation of live/ondemand file-based streams
  * Only validate RTSP parameters if libcurl is available
  * Added 'autoack' parameter to 'call' in SIP plugin to drive NUTAG_AUTOACK
  * Added optional admin key to selected plugins to protect 'create' methods
  * Fix VideoRoom SDP compose error
  * Conditional support of DTLSv1_set_initial_timeout_duration Note: DTLSv1_set_initial_timeout_duration is a method only supported in recent versione of BoringSSL, and allows for setting a different value for the retransmission timeout used by DTLS (which by default is 1s, very high for WebRTC: this commit sets 100ms). You need to explicitly enable the flag when doing the configure as we can't check for its availability in the existing BoringSSL setup, which is typically a static library. Update BoringSSL to the latest version (removed the revision line from the README) and when configuring add --enable-boringssl --enable-dtls-settimeout, then make clean and make again.
  * Fixed check of updated BoringSSL
  * Don't add ongoing recordings to the list
  * Fixed typo
  * Resolves #569 On MacOS X, libraries can be in /opt/local/lib
  * Fixed duplicate pcma in VideoRoom

-------------------------------------------------------------------
Fri Mar 18 09:05:07 UTC 2016 - opensuse-packaging@opensuse.org

- Update to version 0.1.0+git20160318.a1feedf:
  + Fixed management of incoming fragmented WebSockets messages
  + Reduce unneeded verbosity from latest fix
  + Fixed broken support for non-trickling endpoints
  + Add an inactive SDP attribute for rejected/inactive media streams Note: fixes the 'Answer tried to set send when offer did not set recv' exception in Firefox
  + Configurable video codec to force in VideoRoom plugin
  + Send BYE after a POLLERR on RTP file descriptors in SIP plugin
  + Fix typo in voice mail demo
  + Added last_received timestamps to rtp streams and provide info in 'list' message
  + Configurable audio codecs supports in VideoRoom plugin. We should now be able to decide which audio codec ( OPUS, ISAC 32K, ISAC 16K, PCMU ) to use as publisher during creating a room.
  + Removed verbosity of line in SIP plugin
  + Allow users to provide custom headers to add to a SIP INVITE
  + Fixing errors, and suggested improvements by lminiero
  + make last_received_* rtp members part of struct janus_streaming_rtp_source
  + clean up rtp list message response to show age in ms and get rid of 'now'
  + only output video or audio stats if enabled, initialize last_received_* with current monotonic time
  + PCMA_PT was missing
  + Fixing indentation bug, and adding missing code convention practise.
  + Added atomic check to avoid creating ICE thread twice (see #481)
  + Use json_object_iter instead of json_object_foreach (for older jansson versions)
  + Minor fix for coding convention
  + Added request to get info on a specific mountpoint
  + Fixed indentation bug, and added missing code convention practise, and PCMA audio codec
  + Fixed indentation bug, and added missing code convention practise

-------------------------------------------------------------------
Sat Mar 05 10:43:37 UTC 2016 - opensuse-packaging@opensuse.org

- Update to version 0.1.0+git20160304.2ef5bb8:
  + Make DTLS alert and related events more asynchronous
  + Fixed typo
  + fix for janus_videoroom_listener leak for janus_videoroom_listener_muxed
  + Add support for partial writes in websockets transport
  + fix leak of component (timeout) source
  + Update janus_videoroom.c
  + local_ip private network check: if nat_1_1_mapping set, check it instead
  + re-do valgrind suppressions file
  + Renamed valgrind suppression file (see #429)
  + Use a shared buffer for outgoing websockets messages
  + Fixed size of data to write when offset is set
  + janus_ice_send_thread(): use g_async_queue_timeout_pop() instead of g_usleep()
  + Set the right amount of outgoing data to resume after a partial write (websockets)
  + Always free original response in websockets module
  + Fixed broken RabbitMQ transport queues (issue #435)
  + Updated README to use the right tagged version of libwebsockets
  + Fixed check of when to load adapter.js (issue with Firefox 46)
  + Fixed outdated demo description
  + Fix check when hanging up WebRTC peerconnection
  + Ability to configure virtual host, username, and password for RabbitMQ
  + Have the parent wait for an exit code from the child during startup, when daemonizing
  + Wrap write in a do/while to catch EINTR
  + Shorter do/while code for EINTR management
  + Further recommendations on AWS deployment in janus.cfg
  + free allocated memory and move up credentials to be used by either admin or janus api
  + bloody semicolon
  + fix indents
  + Fixed a couple of indent typos, and added info for new RabbitMQ config values
  + Check SIP stack before using it in Sofia callback (issue #447)
  + Use authuser, when provided, for REGISTER as well and not only for INVITE
  + added missing var statements
  + Send DTMF tones using SIP INFO messages
  + Send DTMF tones using SIP INFO messages - configurable duration
  + Added note on upstart in documentation (see issue #455)
  + Send DTMF tones using SIP INFO messages - use inband in the demo
  + Fixed a couple of nits after merging #453
  + Handle 'unpublished' event even in case no DTLS alert was received
  + Reset the hangingup flag in plugin when a new negotiation occurs (to account for cases when hangup_media arrives without a prior setup_media)
  + Better management of poll in streaming plugin
  + Removed unneeded double check
  + Check POLLERR and POLLHUP when waiting for child to start (daemon mode)
  + Don't fail if libmicrohttpd is not found and --disable-rest was provided (issue #461)
  + Avoid ambiguity on number of params for send in janus.js (it's always one, an object)
  + Avoid ambiguity on number of params for send in janus.js (it's always one, an object)
  + Better management of poll in SIP plugin too, and fixed default values for sockets
  + Make fd check more explicit
  + Simplified and clarified poll checks
  + Clarified in the README that Janus will require some configuration files, and that make configs installs a default set of them
  + Fix use of jQuery method before jQuery is loaded (selective logging)
  + Sequential loading of required JS scripts
  + Autodetect libwebsockets version and use the right API
  + Move initial declaration outside of the loop
  + Buffer the latest received keyframe in streaming plugin for new viewers
  + Fixed indentation
  + Fix to race conditions when shuttind down SIP stack
  + Restored missing su_home_init
  + Fixed a couple of memory leaks
  + Fixed video recording for remote packets in SIP plugin
  + Added optional SDES-SRTP support to SIP plugin
  + Added alternative version of janus.js without jQuery dependency (see #464)
  + Allow for optional/mandatory SDES support in SIP plugin
  + Fixed incoming SIP calls with mandatory SDES, and better SDP generation
  + Fix for ID parsing precision in several plugins
  + Fixed compilation errors when detected version of libwebsockets is >= 1.6
  + Try handling more than one timestamp reset when post-processing recordings
  + New hangup request in core, and updated docs
  + Make getUserMedia errors more explicit (due to JSON.stringify failures)
  + Removed exceedingly verbose debug line
  + Removed extra unneeded file
  + Optional docs and updated README Made docs building optional with --enable-docs (issue #474) and added dependency to epel-release in README for CentOS 7 (issue #475)
  + Fixed typo in janus.js when using API secret and WebSockets
  + Fixed typo (wrong prefix in 1.6 branch)
  + Fix check for 1.7 version of libwebsockets
  + Restore, although commented, the README line on the libwebsockets 1.5 stable branch
  + Fixed typos
  + Fixed configure.ac check of websockets
  + Documented additional modes of janus-pp-rec
  + Add number of packets sent/received per medium to Admin API
  + Fix EchoTest demo for Chrome 50
  + Use right RTP profile when answering

-------------------------------------------------------------------
Wed Dec 23 10:38:52 UTC 2015 - lnussel@suse.de

- change certificate location to /etc/janus/cert.pem

-------------------------------------------------------------------
Thu Dec 17 06:58:16 UTC 2015 - opensuse-packaging@opensuse.org

- Update to version 0.1.0+git20151215.b8c852d:
  + First attempt at using conditions (wait/signal) instead of sleeps for some of the workers we have (at the moment, echotest plugin only for testing)
  + Use g_async_queue_pop to implement conditions automatically
  + Use g_async_queue_pop for handler threads in other plugins as well
  + Use static exit_message for plugin handler threads Use conditions to handle/break the main Janus loop
  + Use single GAsyncQueue for incoming/outgoing dat channel messages
  + Reverted unsafe usage of condition in signal handler
  + Removed frequent sleeps in HTTP transport module
  + Use g_async_queue_pop instead of g_async_queue_try_pop in RabbitMQ transport
  + Set got_response when mutex is locked
  + Removed accidentally added video file
  + Make sure we don't free the static exit message
  + Restored sleep-based approach for HTTP transport, and added some fixes as to RabbitMQ
  + Fix message response condition wait in HTTP transport
  + Fixed message response condition wait in HTTP transport for admin too
  + fix structs janus_request and janus_ice_trickle being typedef'ed twice
  + Destroy libwebsockets contexts at shutdown
  + config comment stripping was off-by-one, fix and simplify
  + log msg typo fix "Transpor plugins folder:"
  + Don't free the static exit_message message when shutting down plugins
  + Added third-party PHP stack to the resources page in the docs
  + Correctly skip candidates when using bundle.
  + Fixed a couple of memory leaks in the SIP plugin
  + Initialize timeout value before calling DTLSv1_get_timeout (issue #419)

-------------------------------------------------------------------
Thu Dec 10 07:26:11 UTC 2015 - opensuse-packaging@opensuse.org

- Update to version 0.1.0+git20151209.5e5f9e4:
  + logging: simplify buffer sizing
  + Remove ini_config from configure.ac, since it's not required.
  + Only modify the ice-udp and ice-tcp libnice attributes if the library supports them
  + Improve sample configuration
  + Added option to create/destroy/check PID file
  + Doxygen documentation for new utils methods
  + Set default logging level to info
  + Use atexit to always remove the PID file (if any) before leaving
  + Moved janus_log_destroy to the atexit function
  + sip: add display name to missed_call event
  + Check the result of fscanf wne reading a PID file
  + build: clean all generated sample files
  + Redirect stdin/stdout/stderr to /dev/null (#407), move the related code to log.c (otherwise log init errors when daemonizing may be lost) and don't enable libnice debugging unless explicitly stated (not even if debug level is 7)
  + Don't close standard file descriptors, let freopen do that
  + Updated (and prettified) resources page in documentation
  + configure.ac: ssl_version and glib_version should be shell variables
  + Use the bundled adapter.js instead of an external dependency
  + Added LWS_WITH_OLD_API_WRAPPERS=1 in README for building libwebsockets, to account for the change in their API (issue #410)

-------------------------------------------------------------------
Sun Nov 29 21:54:54 UTC 2015 - opensuse-packaging@opensuse.org

- Update to version 0.1.0+git20151127.0437cad:
  + Attempt to fix the infamous DTLS decrypt alert error (issue #316)

-------------------------------------------------------------------
Sun Nov 29 21:47:49 UTC 2015 - opensuse-packaging@opensuse.org

- Update to version 0.1.0+git20151129.f9e498a:
  + Removed dependency from libini_config, changed the way categories are accessed, and added permanent save of configurations in some plugins
  + Increase plugin API version, although it's the INI stuff that changed
  + Update janus.cfg by removing now useless transport related settings
  + Properly remove WebSocket event listeners
  + Fix wsHandlers misspelling
  + Handle websocket error during session destruction
  + Clear keepalive timeout at session destruction
  + New methods to mute/unmute audio and video in janus.js
  + First code to allow Janus to run as a daemon (no logging yet)
  + Updated version in configure.ac
  + Fixed typo when handling plugin-originated answer
  + Fixed problem of VideoCall plugin not working anymore due to always failing check
  + Fixed problem of SIP calls not getting working RTP after the first time
  + Optional SIPS when registering
  + Use TAG_IF for NUTAG_SIPS_URL
  + Don't gather TCP candidates if ICE-TCP support is disabled
  + init buffered logging
  + formatting
  + tabs are from the devil
  + remove timed wait, reduce locking, tabs
  + remove more glib
  + free buffers and synchronization fixes
  + Use new audio mute functions in videoroom demo
  + Allow admin UI to show either raw or prettified handle info
  + ditch vasprintf from glib printf routines
  + Allow for console and/or logfile output (to hook to config/cmd line) Fix undefined reference in post processing due to new log code A few changes to align the code style to the code base
  + Update janus_log_console when initializing
  + janus_process_error(): use buf on stack, avoid leaking allocated error string buf
  + Configurable logging and daemonization
  + Fixed docs typo
  + Make sure the session is valid and not being destroyed when notifying events (issue #378)
  + More details when something in OpenSSL fails
  + Remove ini_config from configure.ac, since it's not required.

-------------------------------------------------------------------
Tue Nov 10 07:33:12 UTC 2015 - opensuse-packaging@opensuse.org

- Update to version 0.0.9+git20151109.b5865bd:
  + Make sure trickle candidates are not passed to the stack until we have both offer and answer ready
  + Fixed occasional failure to start ICE when answering from a plugin
  + Verbosity change for trickle queueing message
  + Force dummy candidate for unneeded RTCP components when rtcp-mux has been negotiated
  + Changed IP for dummy candidate to 127.0.0.1
  + Added an UDP server (random port) to act as blackhole for keepalives from unneeded RTCP components
  + msg->handle->plugin_handle may not exist when message is handled
  + do not let stun public ip override nat_1_1_mapping ip
  + BUGFIX : opus fill silence packet
  + Added note about better logging when launching Janus via systemd
  + Fixed typo in docs
  + Fixed blackhold fd initialization
  + More conservative suggestions for systemd based logging
  + Fixed access to invalid component when forcing rtcp-mux (issue #370)
  + Fixed silence packet size written when postprocessing audio
  + Removed usage of SO_REUSEADDR for UDP sockets Fixed autogeneration of IDs in streaming plugin Increased size of some sources in debugging code
  + Added fix from #366 and #367 to other plugins as well

-------------------------------------------------------------------
Wed Oct 28 06:57:11 UTC 2015 - opensuse-packaging@opensuse.org

- Update to version 0.0.9+git20151026.a61dd85:
  + Check if adapter is already loaded
  + Fix the duplicate call to the server
  + A couple more memory leak fixes concerning SIP re-invites. Also fixed some per-session leaks where memory associated with the janus_sip_session structure was not begin freed.
  + Added new ICE 'enforce' list, to specify the only interfaces to use for gathering candidates
  + Updated janus.cfg sample to address the new ICE enforce list
  + Removed unused public_ip setting from janus.cfg sample
  + Allow IP addresses to be passes to the ICE enforce list
  + A couple more memory leak fixes concerning SIP re-invites. Also fixed some per-session leaks where memory associated with the janus_sip_session structure was not begin freed.
  + Use different handlers for ws and sws (issue #340)
  + Use different handlers for ws and sws (issue #340)
  + Don't add prflx candidates to the SDP offer/answer
  + Restored the old public_ip setting as a new nat_1_1_mapping setting (-1 on the command line), to clarify what it is for and when it should be used
  + Fixed crash caused by extra g_free on session->stack->session.
  + fix read cert_pem for REST https
  + Add optional BoringSSL support via configure
  + Fixed occasional inability to remove RTP forwarders in videoroom plugins
  + Require a valid certificate key when staring Janus
  + Use 'checkout' instead of 'fetch origin' for BoringSSL
  + Switched inet_ntoa to inet_ntop (new resolving method in utils)
  + Changed debugging for skipped candidates from warning to verbose
  + Free addrinfo after it's been used
  + Fixed echo test data channels forwarding (last character cut away)
  + Converted memory allocations to GLib ones, and fixed a couple of leaks
  + Converted memory allocations to GLib ones, and fixed a couple of leaks
  + Added the possibility to specify an optional PIN to access streaming mountpoints and audiobridge/videoroom conference rooms
  + If both API secret and token auth mechanism are enabled at the same time, either one that is provided and valid is fine
  + If both API secret and token auth mechanism are enabled at the same time, either one that is provided and valid is fine
  + sip: fixed reporting re-INVITEs as missed calls
  + Added method to save a configuration object to file
  + sip: manually handle re-INVITEs and reject them with 488
  + Add .editorconfig file
  + sip: fixup style
  + Fixed a couple of compilation warnings
  + Added a comment header with time for saved configuration files
  + Use minimum FPS as the info to put in WebM header when postprocessing
  + Add info on when the handle was created to the admin API
  + sip: fix handling subsequent incoming calls
  + sip: fix SDP parser leak when handling reinvites
  + off by one buffer overflow
  + protect access to freed janus_websockets_client with old_wss_mutex
  + Further check before pushing plugin session event
  + Fixed typo in writing recording header
  + Added possibility to limit scope of auth tokens to specific plugins
  + Fixed token/plugin check when API secret is involved
  + fixup patch according to janus coding style
  + Add a new helper method to get the system real time, besides the monotonic one
  + Use janus_get_real_time instead of janus_get_monotonic_time for a few things
  + Added admin API methods to dynamically toggle log colors and timestamps
  + Added admin API methods to dynamically toggle log colors and timestamps
  + Return whether API secret and token mechanism are enabled in the server info
  + Return whether API secret and token mechanism are enabled in the server info
  + New UI and features for the admin API web demo
  + Add transports to the new admin API web demo
  + Show docs creation/update time in html pages
  + Allow enter to be used in admin web UI for new tokens
  + Allow for a separate authentication username.
  + Decreased verbosity for some lines (info to verb), and added call to nice_agent_remove_stream when enforcing bundle/rtcp-mux (see #154)
  + Bug-fix: use the correct 'authuser' fields and some indenting cleanup.
  + Enhancement: also report display-name of caller when present.
  + Extra check on "from" field.
  + Allow applications to provide their own MediaStream to janus.js
  + Updated documentation
  + Skip packets that are too large to be RTP in the post processor
  + Fix postprocessing when last packet is broken
  + Send a FIR to the new RTP forward publisher
  + Remove the extra space
  + Send FIR only if forward video
  + Make sure rec_dir is honored even when providing a filename in a videoroom configure request (issue #357)
  + Fixed missing CR in SDP generation
  + Added new console wrappers to janus.js, and bound them to debug level in init (see #292)
  + Fill gaps in audio recordings with silence, when postprocessing
  + Fixed detection of Opus and VP8 payload types in some cases
  + Fixed detection of Opus and VP8 payload types in some cases
  + Removed unneeded extra debugging
  + First attempt at getting Edge and Janus to talk to each other
  + Use code 480 in case a SIP decline is caused by a denied permission on WebRTC
  + Don't start data thread until ICE connectivity has been established
  + Prettier admin UI for handle info
  + List discovered (prflx) remote candidates when querying the admin API
  + Use MediaStreamTrack.stop() (see #363)
  + Updated references to videoroom in the demos, and clarified it's an SFU and not MCU
  + Added autorefresh checkbox for handle info in admin API web demo
  + Fixed parsing of fingerprints so that they can be different per each stream
  + Fixed missing stream/component IDs in janus_ice_component

-------------------------------------------------------------------
Tue Sep 22 16:21:31 UTC 2015 - opensuse-packaging@opensuse.org

- Switched to the modular-transports git branch
  (version 0.0.9+git20150921.e617df8)

-------------------------------------------------------------------
Wed Sep 09 09:59:07 UTC 2015 - opensuse-packaging@opensuse.org

- Update to version 0.0.9+git20150909.f1b79cd:
  + plugins rtsp streaming : manage multicast stream
  + plugins rtsp streaming : fix multicast checking (wrong byte order using IN_MULTICAST macro)
  + streaming plugins rtsp : fix double free + add timeout for RTSP requests
  + streaming plugins : initialize ip_mreq
  + streaming plugins : rtsp : send TEARDOWN before closing connexion and send multicast transport when SDP signal a multicast stream
  + rtsp streaming plugins : check RTSP DESCRIBE return code and enable cURL output depending on log level
  + fix usage of audio_port instead of video_port
  + fix compilation due to renaming log_level into janus_log_level
  + First attempt at getting rid of the increasing delay in audiobridge rooms when network is shaky for a few users
  + config: fix typo, 'apisecret' -> 'api_secret'
  + Attempt to fix occasional issue with websockets and session timeouts (see issue #307)
  + Changed recordings header to contain more info (as of now, mostly codecs and created/first written times), using a JSON format so that it can be extended in the future (old recordings can still be read/played) Added recording capability to EchoTest, VideoCall and SIP plugins Fixed a few nits here and there
  + Only unlock the audiobridge peek buffer after mixing has been done (may help issue #319)
  + Fixed occasional multiple events in reply to the same request

-------------------------------------------------------------------
Fri Aug 28 16:15:42 UTC 2015 - opensuse-packaging@opensuse.org

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

-------------------------------------------------------------------
Wed Aug 26 15:01:37 UTC 2015 - opensuse-packaging@opensuse.org

- Update to version 0.0.9+git20150826.3c5413c:
  + Prevent bower to use a too recent adapter.js
  + Additional checks to avoid using old plugin sessions
  + Changed names of external logging variables to avoid conflicts with libwebsockets
  + ICE Lite fix (conflicting roles)
  + sip-demo: add ability to use HA1 hashed passwords
  + Fixed ICE not starting when all trickles received before processing remote answer
  + Fixed update request in RecordPlay plugin so that deleted recordings are removed from the list (see issue #278)
  + Fixed regression in Record&Play demo (issue #278)
  + Clarified documentation on local, file-based, deployment (issue #291)
  + Suggest version 1.5 of libsrtp in documentation
  + Added AC_CONFIG_AUX_DIR macro to configure.ac (issue #290)
  + sip: fix setting the correct caller for the incomingcall event
  + Fixed typo in sample configuration file, and updated favico
  + sip: send a 'registration_failed' event when SIP registration fails
  + Print timestamp of first detected keyframe when postprocessing videos
  + Added alternative git repo for libwebsockets, in case the first one is unreachable
  + Fixed deadlock in videocall plugin Fixed hangup_media not being invoked in some plugins when preceded by destroy_session (see issue #297 and #298)
  + Made hangingup checks in plugins atomic (see issue #297)
  + Added options to force BUNDLE and/or rtcp-mux (forcing both will always only allocate a single port for media, instead of 2/4)
  + Better management of hangingup flag in plugins (issue #297)
  + doc: update usrsctp repository location
  + build: clean doxygen generated sqlite files
  + build: don't build static versions of the modules by default
  + Fixed issue in janus_dtls_bio_filter_ctrl (issue #308)
  + Fix in management of HTTP URL splitting (issue #309)
  + Fixed wrong verbosity level added in previous commit
  + First take at a daemon/service documentation page (see #306)
  + Added option to disable colors in logging (issue #304)
  + Fix management of new UDP/TLS/RTP/SAVPF rewriting in SIP plugin
  + Fixed issue of sending busy that also hanged up the current call in SIP plugin (see issue #312)
  + Better management of issue #312, new missed_call event in SIP plugin, and fixed missing registration_failed event handler in SIP demo
  + Added upstart sample to the documentation
  + Documentation on how to effectively debug Janus
  + Parse SSRC used for retransmissions by Chrome

-------------------------------------------------------------------
Tue Jul 14 07:53:30 UTC 2015 - opensuse-packaging@opensuse.org

- Update to version 0.0.9+git20150713.bb495e7:
  + echo: return error if unrecognizable message is received
  + Added resetdecoder request (synchronous) and queues length in audit to the audiobridge plugin (issue #242)
  + Added a getVolume() method to janus.js to get the current peer volume, and made both getBitrate() and getVolume() a by request property (don't start timers if they weren't asked for)
  + Made janus.js getBitrate() work with Firefox too (note: does it break Firefox pre-38?)
  + Added JANUS_PPREC_DEBUG environment variable to increase debug in post processor
  + Some more debugging in post processor
  + sip: add ability to choose the response code for 'decline'
  + sip: refactor emitting the 'hangup' event
  + sip-demo: print code and reason for hangup event
  + sip-demo: allow outgoing calls to be rejected
  + Minor nits
  + Selective listeners of media in videoroom, and related fix in core
  + Fixed media constraints for Firefox
  + sip: simplify code for handling SIP authentication
  + sip: add ability to specify a prehashed secret (ha1)
  + sip: separate registration and call states
  + sip: remove redundant check
  + sip: add ability to skip SIP registration
  + A few changes and typo fixes; improvements in janus.js
  + Better checking of invalid configuration object in janus.js
  + Better handling of invalid handle object in janus.js
  + Fixed occasional problems with double detaches (as evidenced in #260)
  + Dropdown menu for registration approach in SIP demo
  + Simple helper request to verify if Janus can write on the RabbitMQ
  + Fixed typos in documentation
  + Fixed a potential problem with incoming RTP streams, and removed a useless parameter in janus_process_success that did nothing (probably a leftover)
  + Added subscriber configure, to dynamically choose what to receive (issue #277)

-------------------------------------------------------------------
Wed Jul  1 07:57:53 UTC 2015 - cwh@suse.com

- Enable janus_postprocessing for oS newer than 13.2

-------------------------------------------------------------------
Thu Jun 18 14:55:18 UTC 2015 - opensuse-packaging@opensuse.org

- Update to version 0.0.9+git20150617.31ac9d1:
  + Fixed missing callback on handle send in janus.js (see #244)
  + Fixed typo in docs (candidate->candidates
  + Fixed typo that caused the wrong pointer to be checked (WS/RMQ), see issue #245
  + demo: Add checkbox for using video in the SIP demo
  + demo: Simplify checking for checkbox state in SIP demo
  + Created 1024 bits certificate (see #251), and added small documentation file
  + Implemented new OpenSSL BIO filter to fix fragmentation issue in DTLS on large certificates (see #252)
  + Made starting MTU value for the BIO filter configurable
  + Restored markdown file describing the certificates folder
  + Added checks to avoid negative integers in API requests (issue #241)
  + rtp_listener feature added for videoroom plugin
  + corrected indention, moved to create sockaddr_in structures for individual streams, both media types are now not mandatory, and changed to rtp_forward
  + Forgot to change OPUS back to actually being opus
  + Fixed detection of incoming RTCP packets (audio vs video) when remote SSRC is unknown (issue #258)
  + Fixed occasional issue when processing video recordings
  + fix a problem in wav header
  + janus: reject incoming WS connections if sub-protocol is not set
  + Remove session from the RabbitMQ manager if it timed out or was destroyed

-------------------------------------------------------------------
Thu May 21 14:28:02 UTC 2015 - ancor@suse.com

- added websockets support 

-------------------------------------------------------------------
Wed May 20 18:05:20 UTC 2015 - mrueckert@suse.de

- prepared post processing support atm guarded with
  janus_postprocessing:
  new BR: libavformat libavcodec libavutil via pkgconfig
- added BR for a few more features:
  ogg and opus via pkgconfig and curl-devel

-------------------------------------------------------------------
Wed May 20 17:14:49 UTC 2015 - mrueckert@suse.de

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

-------------------------------------------------------------------
Wed May 20 14:22:35 UTC 2015 - opensuse-packaging@opensuse.org

- Update to version 0.0.9+git20150520.3f16cb8:
  + Added way for videoroom plugin to just relay FIR/PLI coming from viewers to publishers, for faster video recovery

-------------------------------------------------------------------
Tue May 19 15:26:13 UTC 2015 - opensuse-packaging@opensuse.org

- Run the service as root (until we found the culprit of the errors)
- Update to version 0.0.9+git20150519.0ac1398:
  + streaming : rtsp
  + rtsp : enable rtsp only if libcurl is available
  + rtsp : fix rtp port + keep open RTSP connection
  + rtsp: use dynamic port
  + rtsp : fix crash when media is not supported
  + remove modification of log
  + rtsp: manage create message
  + rtsp update comment
  + rtsp: fix build without libcurl
  + rtsp: fix build without libcurl
  + rtsp: fix build without libcurl
  + rtsp : merge rtp & rtsp structure to reduce copy of code
  + rtsp : merge rtp & rtsp structure to reduce copy of code
  + rtsp: rename method
  + rtsp : fix memory leak + useless duplicate line
  + Trickle error log messages lacked trailing newlines
  + convert double trinary-operator to single trinary for event->payload
  + clean up "adding remote candidate" code, mainly logging
  + Updated bibtek for Janus performances paper
  + fix extra newline when logging ice candidate buffer
  + fix ice log message spelling "credendials"
  + webserver request logging quieter
  + quiet cleaning up session / destroying session log messages
  + log only when starting to wait for webrtc state to change
  + combine multiple feature-state logs into one, quiet redundant feature-state logs
  + quieter logging of final "ice candidate added" message
  + quiet log "Looping ICE"
  + log number of recent retransmits once per 5 seconds at INFO level
  + log retransmitted packets summary at VERB instead of INFO
  + log "Looping ICE" at DBG instead of HUGE
  + slow_link callback refactor: count NACKs over full second
  + Updated bibtek for IPTComm 2014 paper on Janus (in proceedings now)
  + Reduced debug level of REMB transmission in videoroom (VERB, was INFO)
  + only log once when Still cleaning up from previous media session
  + remove check for g_strdup() failing to allocate memory
  + A few changes to pull #217: Added related info to the sample config file; Fixed an htons that was using a pointer instead of the short int; A few cosmetic changes to align to the code style
  + Just a couple of cosmetic changes to pull #230 (capitalize first letter of comments)
  + Fixed indentation (#222)
  + count retransmits, instead of received NACKs, for slow_link
  + avoid starting more requests while janus is stopping
  + Added further check to verify validity of SRTP stack
  + Fixed missing bracket in conditional code in sdp.c
  + in_stats and out_stats: add total new nacks
  + Disabled MHD_quiesce_daemon as per discussion in #235
  + re-write NACK generation for missing rtp sequence numbers
  + Cosmetic changes to #238 (comments) and renamed seq_in_range to janus_seq_in_range

-------------------------------------------------------------------
Wed May  6 07:58:35 UTC 2015 - ancor@suse.com

- added systemd service file
- added janus user
- Update to version 0.0.9+git20150501.35a37ba:
  + make JANUS_LOG macro less redundant
  + Fixed invalid addresses in Via and Contact headers in SIP plugin (issue #213)
  + Handle recent change in libwebsockets build that adds a _shared to the so builds
  + Better management of watchers in case a mountpoint is destroyed (issue #215)
  + sip: definitively remove TPTAG_SERVER tag
  + sip: simplify handling of allocation failures
  + sip: fix potential double-free
  + sip: fix using the duplicated sdp

-------------------------------------------------------------------
Tue Apr 21 15:00:07 UTC 2015 - lnussel@suse.de

- switch to disabled service file for factory package
- change versioning format to 0.0.9+git...

-------------------------------------------------------------------
Mon Apr 13 09:19:31 UTC 2015 - ancor@suse.com

- update to 0.0.8 version
- enable data channels

-------------------------------------------------------------------
Mon Oct 27 09:54:08 UTC 2014 - mlin@suse.com

- switch back to upstream master

-------------------------------------------------------------------
Thu Oct 23 09:58:14 UTC 2014 - mlin@suse.com

- switch the source to ancorgs's break_mcu tree 

-------------------------------------------------------------------
Thu Oct 23 09:23:12 UTC 2014 - mlin@suse.com

- add usrsctp for data channel support 

-------------------------------------------------------------------
Wed Oct 22 14:45:43 UTC 2014 - mlin@suse.com

- let the samples being default configuration if no config file found

-------------------------------------------------------------------
Wed Oct 22 11:26:21 UTC 2014 - maxlin@localhost

- initial import from git