I am not worried about bandwidth so i use ulaw the entire way. then there is no transcoding and soundquality will be good (if you have no bandwidth issues) with those specs I dont think 30 simultaneous calls will be a problem. Why are you not interested in using ulaw? as far as “amplifying the sound” thats all codecs pretty much. Skype has a propreitary codec. there has been work in the asterisk branch on “HD” voice, but at this point ulaw is the best for quality. The reason to use ULAW or THE SAME CODEC the entire way is to minimize transcoding formats, this takes up cpu and messes with quality.
I dont use ztdummy because all of my servers have pri cards in them (for one reason or another).
[quote=“LG”]Yes, how many simultaneous calls from softphone - asterisk - outbound voip provider - enduser?
Did you use ilbc or speex for minimiseing the bandwidth? If I have 3 Gh Xeon dual core, 2 Gb DDR2 and Ultra320SCSI HDD’s the transcoding for ~30 simultaneous calls do you think is a problem? how much CPU usage I should have in this situation? Considering that I’m using this server only for asterisk.
Do you think is possible to amplify the sound from asterisk? for example if I switch to Skype, the sound is a lot louder, the other person seems to be much “closer”,
Do you use a Digium card? or just a ztdummy setup?[/quote]