WCS can work as a WebRTC-SIP gateway. In this case, audio and video stream of a SIP call made through WCS can be captured and played in a browser or republished to another server.
Typical usage scenario
- A video call is established between WCS and a SIP device (SIP MCU, conference server or a SIP softphone)
- WCS receives audio and video data from this SIP device
- The WCS server redirects the received audio and video traffic to an RTMP server or another device capable of receiving and processing an RTMP stream
Supported protocols:
- RTMP
- SIP
Supported SIP codecs:
- Video: H.264, VP8
- Audio: G.711, Speex
Supported RTMP codecs:
- Video: H.264
- Audio: AAC, G.711, Speex
Capturing and republishing of SIP calls is managed using REST API queries.
- Configuration
- Redirecting a SIP call to a stream (SIP as Stream function)
- Publishing a SIP call as an RTMP stream (SIP as RTMP function)
- Publishing of a SIP call stream to an RTMP stream with the /push/startup REST function
- Known issues