Broadcast_SIP_call_to_streamRTMP

Capture video and republishing of SIP video call

Functions SIP as Stream and SIP as RTMP

A SIP call made through the WCS server can be captured in 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.

Specifications

Capturing and republishing SIP-phone calls is controlled by the REST API calls

Protocols and Technologies SIP codecs RTMP codecs SIP endpoints
  • SIP
  • WebRTC
  • Flash
  • RTMP
  • H.264
  • VP8
  • G.711
  • Speex
  • Opus
  • H.264
  • AAC
  • G.711
  • Speex
  • Polycom DMA
  • Cisco Media Server
  • Asterisk
  • OpenSIPs
  • Bria, etc

Scheme for capturing and republishing a stream from a SIP call

  1. A video call is established between the WCS and the SIP device (SIP MCU, conference server or 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 other device that can receive and process an RTMP stream, or plays in the player

 

sip to stream rtmp SIP as RTMP WCS stream VLC browser Flash

 

Step-by-step diagram of capture and republishing of a stream from a SIP call

  1. REST client starts a call using a REST call /call/startup;
  2. WCS server connects to SIP server;
  3. SIP server transfers audio-video stream to WCS server;
  4. The browser requests playback of the stream from the WCS server;
  5. The WCS server broadcasts the audio-video stream that was received from the SIP server.

 

sip as stream call flow SIP as RTMP WCS stream VLC browser Flash

 

Step-by-step diagram of capturing and republishing a stream from a SIP call to RTMP

  1. REST client starts a call using a REST call /call/startup;
  2. WCS server connects to SIP server;
  3. SIP server transfers audio-video stream to WCS server;
  4. The WCS server republish the received audio-video stream to the RTMP server.

 

sip as rtmp-call flow SIP as RTMP WCS stream VLC browser Flash