Thanks for this interesting info.
I googled this link myself:
http://mnis.fr/ocera_support/rtos/c1393.htmlActually the situation when I think it is a quite hearable difference in sound quality, is not when it is a steady flow of datas the one way or the other, it is mainly when two persons try to speak in two ends of "the line" simustinely.
When both persons try to speak at the same time its allmost like it generates a series of some "pauses" of a leght of a fraction of a second. Thats what its sonds like, and thise make the whole connection to sond a bit like "ip telephony".
When using Astlinux that is made for IP telephony only, it is reasonably to believe that this contains a kernel that is optimized for IP telephony and realtime processing of datas.
When using Astlinux there is no problems with two persons speaking at the same time at all. It still sounds completely "smooth" like a analogue telephony line. There is no "ip telephony sound" at all.
But what about Asterisk at the SME server, does this give "ip telephony sound" or will it typically give "a analogue telephony sound" ? (I have runned Asterisk at SME server before, but I did not compare with other installations.)
If this eventually should be a problem what should then be the solution for the problem ?
1. To compile a modified kernel, optimized for ip telephony ?
2. To keep gateway, web server, file server etc in one box and the ip telephony server in one other box, because those server functions does not perform well together ?
Actually I have not really checked out if Asterisk at SME server realy gives a "ip telephony sound".
Anybody who has any toughts about or experiences with that ?
(One day I will set up an Astlinux box and a SME server box with Asterisk to compare, but I have not done that yet. It would be nice if it should apear that the sound quality is the same.)
Best reg Arne.