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    copied!<p>This is a lot of work as there is no "built in" sdk features or third party off the self components that will help you do it without a lot of work on your end. Your options (as far as I know) are:</p> <ul> <li>Build your own solution.</li> <li>Look for a third party solution.</li> <li>Look for a open source solution.</li> </ul> <p>Voice "chat" could be one of many things like:</p> <ul> <li>using the voice features of a cellular network to a conference call.</li> <li>using the voice features of a cellular network with a PBX server that supports conference calls.</li> <li>using a <a href="http://en.wikipedia.org/wiki/Voice_over_Internet_Protocol" rel="noreferrer">VOIP</a> solution using a <a href="http://en.wikipedia.org/wiki/Session_Initiation_Protocol" rel="noreferrer">SIP</a> stack with a SIP server</li> <li>using a <a href="http://xmpp.org/" rel="noreferrer">XMPP</a> <a href="http://en.wikipedia.org/wiki/Jingle_%28protocol%29" rel="noreferrer">Jingle</a> solution (I believe the google voice service uses this) </li> <li>using your own SIP setup solution</li> <li>custom solution</li> </ul> <p>None of these options are easy. </p> <p>Open Source SIP implementations that have iphone ports (that I know of):</p> <ul> <li><a href="http://www.resiprocate.org/Main_Page" rel="noreferrer">reSIProcate</a></li> <li><a href="http://www.pjsip.org/" rel="noreferrer">pjsip</a></li> </ul> <p>Update:</p> <p>SIP &amp; Jingle both use <a href="http://en.wikipedia.org/wiki/Real-time_Transport_Protocol" rel="noreferrer">RTP</a> for the actual transport protocol between the parties. RTP is a UDP point-to-point protocol. The ports which form a session are negotiated using other protocols such as RTSP (using SDP in the setup method) and SIP. RTP and RTCP typically use unprivileged UDP ports (1024 to 65535).</p>
 

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