<br><br><div class="gmail_quote">2013/2/20 Nguyễn Công <span dir="ltr"><<a href="mailto:nguyencong.1210@gmail.com" target="_blank">nguyencong.1210@gmail.com</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div lang="EN-US" link="blue" vlink="purple"><div><p class="MsoNormal">Hello everyone, I’m new to Asterisk and I have a question. There is a phone call between two users, then they are talking to each other directly or by the server. I mean all packets from the user A to user B will be send directly to each other or will those packets from user A must be send to server and server will send to user B.<u></u><u></u></p>
<p class="MsoNormal">Thanks.<u></u><u></u></p></div></div><br>--<br></blockquote><div><br></div><div>Both cases can happens. In a VoIP call we have two connections, one is used for signaling, usually port 5060 for SIP protocol, UDP transport and one is used for media (voice), usually random port. When the call starts the asterisk server sits in the middle of the media path, meaning all voice packets from phone A go to asterisk server and they are rerouted to phone B. After few milliseconds, if configured this way, asterisk server instructs the phone A to send the media directly to phone B to save bandwidth. It is named "reinvite"</div>
<div><br></div><div>Leandro</div></div>