[Added]
- Added the
<TcpForce>
option to enforce WebRTC over TCP. - Added simulator for WebRTC performance measurement.
- Added segmentation rule setting option in file recording.
[Improved]
- AES GCM has been applied to SRTP.
- Improved Socket stability.
- Reduced the number of unnecessary threads in the transcoding filter.
[Changed]
- Changed
libsrtp
version to v2.4.0
[Fixed]