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1. For the test we use:

• two SIP accounts;
• a softphone to answer to the call;
• the REST client of the Chrome browser;
• the program to broadcast an RTMP stream, OBS Studio.

2. Run the RTMP stream broadcast program. Configure and start broadcasting to your WCS server using the following parameters:
URL of the server: http://demo.flashphoner.com:1935/live/
Stream key: obsStream

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3. Open the REST client. Send the /call/startup query to the WCS server, and in the query parameters specify the following:

• parameters of your SIP account the call is made from;
• the name of your second SIP account the call is made to.

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4. Receive the call in the softphone:

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5. From the REST client send the /call/inject_stream query to the WCS server and specify the following parameters:

• the identifier of the call;
• the name of the broadcast stream (obsStream).

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Make sure the audio track of the broadcast can be heard in the softphone.

6. To terminate the call, click the corresponding button in the softphone.