SIP-RTMP gateway demonstration

This is a demonstration video of siprtmp software. I will start the siprtmp gateway on a terminal. Then I will start a SIP server on another terminal. The SIP server is running on port 5060 on local host and siprtmp on port 1935. I will load a web page containing the web client. It shows Flash Player security prompt where you approve device access. Configure your gateway address as rtmp://localhost/sip then your local SIP address of record as kunweb then set your authentication name, password and display name.

Select to remember the config if you want. It register your SIP address with the SIP server via gateway. You can see the server and gateway logs to verify this. Now I will start another SIP client, X-lite. It registers as user kunxlite to same server. Verified via server logs. Let me show you x-lite config to register. Also the audio codec list configured. It must include Speex and Speex wideband for compatibility with Flash Player audio. Lets call from kunweb to kunxlite as an example of Flash to SIP voice call.

Once x-lite rings, I answer the call. The call is successfully connected. Also verified the server and gateway logs. You can terminate the call from web or SIP. Let us call from kunxlite to kunweb (SIP to Flash). Accept from web and call is connected. Again, terminate from web or SIP side. Visit this URL for more details. Use this code to include in your web site..