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1. For the test we use:

• two SIP accounts;
• a softphone to answer the call;
• the REST client of the Chrome browser.

2. On the WCS server create a directory: /usr/local/FlashphonerWebCallServer/media. Put a file in the RIFF WAV format there, for example test.wav.

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

• parameters of your SIP account the call is made from;
• the name of your second SIP account the call is made to;
• the name of the stream for the following /push/startup query, for example call_stream1:

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4. Receive the call in the softphone:

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5. From the REST client send the /push/startup query to the WCS server and specify in the query's parameters:

• the URL of the WCS server to accept the RTMP stream;
• the name of the call (call_stream1);
• audio file to stream redirection options "options": {"action": "sound_on", "soundFile": "sound.wav", "loop": true}

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6. From the REST-client send the /call/inject_stream query to the WCS server and specify in the query's parameters:

• the identifier of the call;
• the name of the stream with the applied audio file (rtmp_call_stream1).

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Make sure the softphone plays the test file.

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