sip as stream

  1. T

    Stream is not playing

    While i am checking the stream I am getting the error like session is not initialized or terminated on play ordinary stream.
  2. A

    Mixing of SIP calls and WebRTC calls in Conference

    Hi there, Is there a possibility of Mixing of SIP calls and WebRTC calls in Conference? kindly assist... Regards AB
  3. R

    sip в микшер

    Как можно входящий sip звонок добавить в микшер? чтобы как с миксером говорим в user1#b, а слушаем b-user1b а то я добавляю и слышу сам себя, и еще вопрос как убрать видео "hasAudio": true, "hasVideo": true,
  4. 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]"...
  5. B


    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