[Asterisk-Users] MTU and Voice Delay (latency??)

Rusty Dekema rdekema at gmail.com
Mon Jan 9 17:02:36 MST 2006


How far (physically) is the Asterisk server location from the location of
the phones? Have you tried pinging the Asterisk server from the network to
which the phones are connected?

As a rule of thumb, If the two sites are within 2500 miles of each other and
the network connection between them is working properly, the round trip time
for a 64 byte ping should be less than 100 ms, the round trip time should
not vary from one ping to another by more than 2-5 ms (typical), and there
should be virtually no dropped packets (well under 0.1%).

If your network does not meet these standards, then it may well be the cause
of your problems. In that case, if you e-mail me a traceroute from the phone
location to the Asterisk location as well as the output of a ping from the
phone location to the Asterisk location (preferably including at least 100
repetitions), I will take a look at it and let you know what I think.

If your network seems fine by the above standards, then you/we will have to
pursue other Asterisk/SIP/RTP-related avenues of troubleshooting.

Regards,
Rusty



On 1/9/06, Geoff Manning <gmanning at zoom.com> wrote:
>
> Our users are experiencing some unacceptable delay when trying to have a
> conversation. The delay is so noticeable that they keep stepping on each
> others words and resort to calling the customers via cell phone.
>
> Here is the setup
>
> SDSL Connection (PPPoA)
> Speedtouch 610 SDSL Modem
> 3Com 2224PWR Plus Switch (phones on separate VLAN)
> 8 Cisco 796 Phones
>
> All connecting to a remote Asterisk Server.
>
> We found that the MTU for the SDSL modem was set to 1500 and I have since
> changed it to 1458 which is the ISP's recommended setting.
>
> Can this MTU difference cause the delay my users are experiencing? All the
> voice packets would become fragmented so it sounds logical.
>
> And simply changing the MTU on the modem, will that fix it, I can't find a
> way to change it at the Cisco phone level.
>
> Thanks,
> Geoff Manning
>
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