Capturing video and republishing of SIP video call
Functions SIP as Stream and SIP as RTMP
A SIP call made through a WCS server can be captured to the stream on the server when making the call. Then this stream can be played in a browser or republished to an RTMP server for further playback.
Capturing and republishing SIP phone calls is controlled by the REST API calls
Specifications
| Protocols and technologies |
SIP codecs | RTMP codecs | SIP devices |
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Protocols and technologies
- SIP
- WebRTC
- Flash
- RTMP
SIP codecs
- H.264
- VP8
- G.711
- Speex
- Opus
RTMP codecs
- H.264
- AAC
- G.711
- Speex
SIP devices
- Polycom DMA
- Cisco Media Server
- Asterisk
- OpenSIPs
- Bria, etc
Scheme for capturing and republishing a stream from a SIP call
- A video call is established between the WCS and the SIP device (SIP MCU, conference server or SIP softphone)
- The 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 other device that can receive and process an RTMP stream, or plays in the player
Step-by-step diagram of capturing and republishing a stream from a SIP call
- REST client starts a call using REST call /call/startup;
- WCS server connects to SIP server;
- The SIP server transfers audio-video stream to the WCS;
- The browser requests playback of the stream from the WCS server;
- The WCS server broadcasts the audio-video stream that was received from the SIP server.
Step-by-step diagram of capturing and republishing a stream from a SIP call to RTMP
- REST client starts a call using REST call /call/startup;
- WCS server connects to SIP server;
- The SIP server transfers audio-video stream to the WCS server;
- The WCS server republishes the received audio-video stream to the RTMP server.
