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

Geoff Manning gmanning at zoom.com
Wed Jan 11 05:56:19 MST 2006


Rich Adamson wrote:
> No. The reason is that "if" the phones are the only thing on this, the
> size of the sip packets will never be greater then 214 bytes.  

> Given your table below, there "are" other devices on your network and
>   6% of those are sending packets of in the 512 to 1023 byte range.

Actually these are the only devices, honestly. Looking at a packet capture
from the SDSL network shows plenty of larger packets. The SIP Invite packets
are 769 bytes, SIP Notify at 516 bytes, SIP Option packets at 481, Register
packets between 430-609 bytes, Status 200 at 725 packets. They are minimal
in number compared to the RTP packets though.

> 
> Have you tried the previous suggestion relative to two simultaneous
> ftp sessions?

Unfortunately not, I have no access to the remote site inside the LAN. The
onsite tech is out of the office and it is difficult to walk others through
this process.

> 
> What city/state are you located in?
> 

The phones and the asterisk server are in London.



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