[asterisk-users] High delay and some echo

Antony Stone Antony.Stone at asterisk.open.source.it
Tue Jun 11 14:10:27 CDT 2019

On Tuesday 11 June 2019 at 20:53:09, Luca Bertoncello wrote:

> Am 11.06.2019 um 20:42 schrieb Antony Stone:
> Hi Antony,
> > I think the main question here is: how are you connecting Asterisk to the
> > telephone system?
> Via VoIP...
> > You mention that you're on DSL from Deutsche Telekom, but is the call
> > going over this DSL link to soem SIP provider, who then connects you to
> > the PSTN, or are you connecting Asterisk locally to the phone line via
> > some ATA device?
> Deutsche Telekom uses since years just VoIP. No ISDN, PSTN, and so on... :(

Well, same as Net Cologne here where I am, but the cable modem I have still 
has PSTN sockets on it so you can connect analogue phones to it as well as 
speaking SIP to it.  I wasn't sure which you might be doing with your 

> I'm connecting to the VoIP-Server of Deutsche Telekom via DSL (50Mbps
> down, 10Mbps up).

So, you have a SIP phone, connected to an Asterisk server on your local 
network, which then connects to D Telekom's SIP server over the DSL line?

> The other party use VoIP, too, since they are in Germany (and Italy) and
> here there are just VoIP... Sigh!

Are they also using a SIP phone?

Do they also have an Asterisk server on their local network?

> Now I disabled the jitter (jbenable = no), and I called my father in
> law. He sayd me, the quality is really better, but I hear sometimes
> little noises...
> Any other suggestion?

Have you considered trying some tool such as http://sipcapture.org/#about to 
see if you can identify where the latency comes in?


Schrödinger's rule of data integrity: the condition of any backup is unknown 
until a restore is attempted.

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