[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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