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<TITLE>Re: [Kamailio-Users] [OpenSIPS-Users] Dimensioning a telephony system based on openser!</TITLE>
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On 10/30/08 5:49 AM, "Nuno Marques" <<a href="yangsengopenser@gmail.com">yangsengopenser@gmail.com</a>> wrote:<BR>
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</SPAN></FONT><BLOCKQUOTE><FONT FACE="Calibri, Verdana, Helvetica, Arial"><SPAN STYLE='font-size:11pt'>...” i don't need to use mediaproxy in all calls, only in the NATed ones” ....<BR>
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</SPAN></FONT></BLOCKQUOTE><FONT FACE="Calibri, Verdana, Helvetica, Arial"><SPAN STYLE='font-size:11pt'>I using 1.4.0 with the new NAT Transversal module, and it so far it handles <B><U>all </U></B>my NATed clients; even folks that have devices that don’t support STUN (like the older Polycom IP Soundpoint phones). So in this case, the above statement is not true with me as I am not proxing their audio.<BR>
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I only proxy media under certain circumstances, like a court-ordered subpoena (CALEA), call re-direction support (which I haven't got fully working yet), or virtual fax and other media services (voicemail, conf calls, etc) from which the audio goes straight to my asterisk machines. And even with those, those are on a per-caller basis.<BR>
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With each g711u call leg, taking around 85kbps - that’s 170 for each handled call ... 85 in, 85 out ... you can really start eating away at bandwidth. <BR>
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Plus, I am finding that the call quality is a bit better when the audio goes directly from the NAT client straight to the PSTN provider. While we do operate our own network (AS / BGP, with two Tier1 and Tier2 providers), if I don’t have to proxy the audio, the better.<BR>
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And for the record, I have yet to come across a billing issue. I have clients that pay for unlimited service and I have ones that pay on bundled minutes plan.<BR>
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Hope this helps and regards,<BR>
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-graham</SPAN></FONT>
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