sip as stream

  1. L

    REST - Starting call with hasAudio false still uses sendrecv in SDP causing call to timeout

    Using Web Call Server Version 5.2.755 I'm passing the following to /rest-api/call/startup { "callId": "1234567", "callee": "1234567", "hasVideo": false, "hasAudio": false, "sipLogin": "[login]", "sipAuthenticationName": "[authName]", "sipPassword": "[password]"...
  2. B

    RTMP to SIP/RTP

    Hi, After reading the documentation I can't find a way to use WCS as RTMP gw and convert RTMP (input) to SIP/RTP (output) , only SIP/RTP to RTMP. Is it possible? Is there an exemple? Thx in advance
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