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copied!<p>Firstly I implemented something similar for my company a few months back.</p> <p>Lessons learnt:</p> <pre><code>1. you can't just pass IPs around and expect the users to like that over skype. Solution: a. You will need your own server with the necessary ports forwarded. You will have to use some sort of firewall hole punching algorithm(take a look at UDP hole punching). 2. Using existing VoIP library is always better. Downside? You can't write proprietary code using opensource library. Hence you will need to learn H.323 and RTCP/RTP protocol. 3. You will need to write echo reduction algorithms for voice. 4. COMPRESS your audio data before sending it to another computer. PCM data can and will clog your network, delaying sound and fuzzing up everything in the process. Use aLaw and uLaw compression schemes. 5. Make sure you take care of all the error conditions. Multimedia over network can be tricky if not really hard to implement. 6. DONT USE QT. Use a platform specific framework like .NET and libraries that deal with sound (NAudio). </code></pre> <p>I think this will sum up the problems you will need to solve first, before delving into the art of VoIP programming. </p> <p>For your question, your problem is much smaller.</p> <pre><code>1. You don't need echo reduction algorithms IF you use headsets. 2. You don't need to write hole punching algorithms if you're OK with passing IPs around. Take a look at NAT traversal(UPnP?) if the data is suppose to go on a network and to a computer that isn't on your LAN. FLOW: COMPUTER1->DATABUFFER->COMPRESSuLaw/aLaw->NETWORK->DECOMPRESSuLaw/aLaw->OTHERCOMPUTER and vice versa. </code></pre> <p>Good luck :)</p>

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