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 <[EMAIL PROTECTED]> 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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