[Asterisk-Users] Compare to Skype

Gabriel Afana asterisk at gafana.com
Sat Apr 29 21:04:57 MST 2006


> One of my user is praising Skype!!!
>
> I cannot figure out anymore what I can improve!
>
> This users sip show peers  is jumping from 65 msec to 1800 all the time. 
> Of course his voice quality is like a morse code with dashes or dots of 
> connection time.
> The next minute he calls me via Skype and it works fine !!!! What 
> indicates that there is no fault on his Internet connection!!!
>
> He is using his notebook and Xlite, but also tried the snom 360.
>
> Any hints?

Is he calling you on another VoIP phone or calling you on a 
landline/cellphone (through the PSTN)?  If he is calling a 
landline/cellphone, then it is probably your upstream termination provider 
that is having jitter problems (this is my exact problem).  If I check my 
voicemails on my IP phone (which connects directly to my asterisk box 60 
miles away), everything is great.  HOWEVER, if I *dial* my telephone number 
and check my voicemails (as if I was calling in to check my voicemails), I 
get loads of jitter.  So between my IP phone and my * box, the connection is 
great, but its what is after my * box that is causing the problem.

Who is providing you termination?

- Gabe
 




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