janus-gateway
https://github.com/meetecho/janus-gateway
C
Janus WebRTC Server
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- Issues
- [1.x] janus.js default simulcast scaleResolutionDownBy (1-2-4) force per-layer encoders in libwebrtc
- Videoroom: Don't link subscriber streams after the handle is destroyed
- SIP plugin: don't leave a helper with the call handle of a destroyed stack
- [1.2] Adding and remove streams using NOSIP Plugin, SRTP context fix
- Allow separate audio/video negotiation of SRTP in NoSIP plugin (fixes #3717)
- [1.x] Crash in NO-SIP Plugin
- [1.x] SIP extension stuck in invited and returns 486 Busy Here after caller disconnects during ringing
- AudioBridge: fix jitter buffer seq wrap, and PLC errors killing participant thread
- Save frames to plugin recorders holding a reference to them
- sip: keep REFER handles valid for as long as the transfer is tracked
- Docs
- C not yet supported