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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

Web Call Server 5.1 - EN > Redirecting a stream to a SIP call using /call/inject_stream > republish-sip-obs.jpg

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.

Web Call Server 5.1 - EN > Redirecting a stream to a SIP call using /call/inject_stream > republish-sip-call.jpg

4. Receive the call in the softphone:

Web Call Server 5.1 - EN > Redirecting a stream to a SIP call using /call/inject_stream > republish-sip-answered.jpg

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).

Web Call Server 5.1 - EN > Redirecting a stream to a SIP call using /call/inject_stream > republish-sip-inject.jpg

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.