<div>Or Put another Way Asterisk acts in SIP terms as a Back2Back User Agent, to terminate one side of the call let and originate a new call leg with a different codec profile in the SIP/SDP. Asterisk then terminates the inbound media, transcodes it an originates a new media stream on a completely different call leg.</div>
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<div>Neill....;o)<br><br></div>
<div class="gmail_quote">2009/1/23 Iņaki Baz Castillo <span dir="ltr"><<a href="mailto:ibc@aliax.net">ibc@aliax.net</a>></span><br>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">2009/1/23 Rawshan Iajdani <<a href="mailto:iajdani@provati.com">iajdani@provati.com</a>>:<br>><br>
> UA----->OpenSer(Outbound Proxy)---------Register Server<br>> | |<br>> |<br>> Asterisk(codec converion)----------------------<br>
><br>> The UA will register to Register server through outbound proxy OpenSer. When<br>> UA makes call it first comes to Openser, OpenSer should route the media to<br>> Register server through Asterisk for codec conversion. OpenSer will not hold<br>
> any User account rather it will act as a proxy.<br><br>Asterisk cannot receive *just* the media, it needs to receive the SIP<br>signalling so then it can handle the media (and do the codec<br>conversion).<br><br>--<br>
Iņaki Baz Castillo<br><<a href="mailto:ibc@aliax.net">ibc@aliax.net</a>><br><br>_______________________________________________<br>Kamailio (OpenSER) - Users mailing list<br><a href="mailto:Users@lists.kamailio.org">Users@lists.kamailio.org</a><br>
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