I have fount the Sipx relay to have little impact on call quality, other than your typical QOS issues on a low bandwidth internet connection. But thats not Sipx's fault.
The difference with Asterisk, is that Asterisk transcodes everything to IAX, where as Sipx is just relaying the traffic. So a SIP call in Asterisk goes from SIP Endpoint -->> IAX Transcoded --->> Other SIP Endpoint -M >>> "Yakout Esmat" <[email protected]> 05/22/09 7:45 AM >>> <o:shapedefaults v:ext="edit" spidmax="1026" /> <![endif]--> <o:shapelayout v:ext="edit"> <o:idmap v:ext="edit" data="1" /> </o:shapelayout><![endif]-->Hi All,<o:p></o:p> <o:p> </o:p> So far we have been avid users of Asterisk-based PBXs withall our clients running distributions like Trixbox and PIAF.<o:p></o:p> <o:p> </o:p> But now with the release of the new sipX version 4.0, Ibelieve that we have to revisit our strategies.<o:p></o:p> <o:p> </o:p> I have many questions but letÂ’s focus on quality issuein this post, my question is:<o:p></o:p> <o:p> </o:p> Is there any difference in the way sipX handles RTP mediacompared to Asterisk? And whether or not that difference affects the quality ofVoIP calls given that all other parameters are equal?<o:p></o:p> <o:p> </o:p> I know that with sipX, media goes directly between endpoints bypassing the server all together, but this is only true if there wereno NAT between these end points.<o:p></o:p> <o:p> </o:p> If we had 2 endpoints separated by NAT, hence media has totravel through the Media Relay agent (the sipX server) would the call qualitybe better than if the same endpoints were using an Asterisk server? All elseare equal.<o:p></o:p> <o:p> </o:p> Thanks for your feedback.<o:p></o:p> <o:p> </o:p> Yakout<o:p></o:p> <o:p> </o:p> <o:p> </o:p> </[email protected]>
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