Features
- H265 Transcoding from RTMP to WebRTC #2058
- WebM Recording #2144
- Force WebRTC Player to play at specified resolution #2155
- Create a websocket message that returns the available streams in the conference room #2227
- Create a websocket message that notifies client that if bandwidth is less than the video/audio bitrate #2103
- Check broadcast start and end time before accepting the WebRTC Stream #2181
- Update video.js to the latest version for HLS and MP4 playback #2231
- Create a REST method that can send message to the viewers through Data channel #2026
- Provide the ability to choose audio input in WebRTC publishing #2164
- Implement switch in front and back camera in JS SDK for mobile platforms #2022
- Fetching streams in the origin cluster #1406
- Support Unified Plan or PlanB in WebRTC #2226
- New REST method to get VoD Id by Stream Id #2244
Fixes and Improvements
- Upgrade Tensorflow Library to 1.15.0 #2025
- Adding Facebook RTMP Endpoint is not working #1981
- Fix 10 NAL Units in libx264 && freeze and quick play #2037
- Show total available memory in the web panel #2136
- The sound stops after 20 seconds on the edge server #2198
- MP4 Files cannot be downloaded because of the wrong absolute path #2070
- Unexpected number of HLS viewers increase #2015
- Decrease number of threads in WebRTC signaling #2265
- Fix for EncoderBlocked Warning #2273
- Micro freeze in some RTMP streams #2095
- Stream fetcher does not start again after restart period #2241
- Edit stream source does not work if it's not fetching #2251
- MP4 files uploaded in S3 have public_read permission issue #1965
- Completing MP4 record while server is stopping #2030
- phtread_create exception in some instances #2254
- Add second to the date-time value in mp4 recording #2232
- Fix external SSL certificate #2301
- Add listenerHookURL in updateSettings #2230
- SFU Mode sometimes does not work in H264 & VP8 Enabled #2175