Server functions¶
Streaming video functions¶
IN - streams incoming to the server (publishers)
OUT - outgoing streams (spectators)
IN / OUT
WebRTC browser
Flash Player
MSE
WSPlayer
HLS
RTSP
Android app, WebRTC
iOS app, WebRTC
WebRTC Browser
Webcam
+
+
+
+
+
+
+
+
Canvas
+
+
+
+
+
+
+
+
Screen
+
+
+
+
+
+
+
+
Flash Player
+
+
+
+
+
+
+
+
RTMP encoder
+
+
+
+
+
+
+
+
RTSP IP cam
+
+
+
+
+
+
+
+
RTMP server
+
+
+
+
+
+
+
+
WCS server
+
+
+
+
+
+
+
+
SIP call
+
+
+
+
+
-
+
+
Android app
+
+
+
+
+
+
+
+
iOS app
+
+
+
+
+
+
+
+
VOD
+
+
+
+
+
+
+
+
Supported codecs¶
Audio |
Hz |
Video |
Hz |
---|---|---|---|
Opus Speex G.711 AAC AAC G.729 |
48000 16000 8000 48000 44100 8000 |
H.264 VP8 |
90000 90000 |
Incoming streams operations¶
- Management of camera, microphone, bitrate, resolution etc.
- Mixing streams
- Taking a preview snapshot of a stream as PNG
- Recording streams
- Forced stopping of streams on the server
- Searching for current streams on the server
Video streams republishing functions¶
- To another RTMP server
- To another WCS server via WebRTC
- To a SIP call
Complex functions¶
- Working with chat rooms
- CDN 1.0, static
- CDN 2.0, dynamic
- CDN 2.1 with transcoding nodes
WebRTC-SIP gateway functions¶
From - the caller
To - the callee
From / To | WebRTC browser | Android App, WebRTC | iOS App, WebRTC | SIP |
---|---|---|---|---|
WebRTC browser | + | + | + | + |
Android App, WebRTC | + | + | + | + |
iOS App, WebRTC | + | + | + | + |
SIP | + | + | + | + |
Call management functions¶
- DTMF
- Hold
- Transfer
- Call recording
Supported codecs¶
Audio |
Hz |
Video |
Hz |
---|---|---|---|
Opus Speex G.711 G.729 |
48000 16000 8000 8000 |
H.264 VP8 |
90000 90000 |