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

1. A video call is established between WCS and a SIP device (SIP MCU, conference server or a SIP softphone)
2. WCS receives audio and video data from this SIP device
3. 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