Hi All,
So far we have been avid users of Asterisk-based PBXs with all our clients running distributions like Trixbox and PIAF. But now with the release of the new sipX version 4.0, I believe that we have to revisit our strategies. I have many questions but let's focus on quality issue in this post, my question is: Is there any difference in the way sipX handles RTP media compared to Asterisk? And whether or not that difference affects the quality of VoIP calls given that all other parameters are equal? I know that with sipX, media goes directly between end points bypassing the server all together, but this is only true if there were no NAT between these end points. If we had 2 endpoints separated by NAT, hence media has to travel through the Media Relay agent (the sipX server) would the call quality be better than if the same endpoints were using an Asterisk server? All else are equal. Thanks for your feedback. Yakout
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