[Asterisk-Users] Skipping problems on outgoing calls (using uLaw with an internal * server through Voxee)

[Robert Geller]

Hello all,

I am using a headset and the X-lite softphone (sometimes I use IAXComm, 
but I'm having difficulties using OSS emulation with it) to connect via 
uLaw to my internal Asterisk server, which is a Pentium II 400 with 128 
megs of RAM. After getting this headset, most or all of the echo people 
on the other line were complaining about is now gone, according to them. 
However, every five to ten seconds, I get quick "skipping" or lag on the 
other side, so that the person whom I'm talking to's voice sounds like 
it "skips a beat," analogous to when a CD you're listening to skips quickly.

I don't think -- but am not positive -- that it is a question of 
insufficient bandwith, as I am on a Cox 5mbps/2mbps cable line that is 
very reliable and pretty stable. I believe I am using uLaw both to the 
Asterisk server /and/ from the Asterisk server to Voxee, my outgoing SIP 
provider/PSTN terminator.

Is this a common problem? It doesn't seem like it should be, as it is a 
major detriment to having enjoyable, good-quality VoIP conversations and 
doesn't seem like it would be the "standard" for such conversations. 
Perhaps I shouldn't be using uLaw, but this really bugs me because I do 
have the bandwith to use uLaw, and its quality is unsurpassed.

Could this be an insufficient RAM problem with my * server? As I have 
128 megs of RAM on my PII, about 122 megs of it are constantly in use, 
and the CPU is, for the most part, pretty idle during single 
conversations. I'm not sure about incoming calls when music, etc., is 
played, but I'm not talking about that right now -- just the skipping 
I'm getting when I make outgoing SIP calls to Voxee (and, ultimately, to 
the PSTN). Would the problem be resolved with more RAM? This is an old 
Compaq Deskpro that I'm using as an Asterisk server (not much else is 
running, but I haven't specifically optimized it) on Debian, and I don't 
even know what type of RAM it takes and can find no documentation to 
tell me.

The problem exists even when I make internal SIP calls, i.e. to 
voicemail (Comedienne mail, is it?) and other test extensions. Allison, 
the voice of Asterisk, asks for your mailbox, but it isn't a continuous 
flow; instead, it skips: "Maaa-aa-aailbox".

Something is definitely wrong, and I eagerly await advice and the key to 
making crisp, clear VoIP/PSTN calls -- free of this extremely annoying 
skipping!

Regards,
Robert Geller