[Asterisk-Users] SIP with Delay

[M.N.A.Smadi]

no this is not normal and the codec has very slight effect on the delay.
Delay is a function of two things:
1) Transcoding
2) Routing (dialplan routing + network latency)

Network latency >> Transcoding+dialplan routing.
If you are using two sip client which are using the same codec then no 
transcoding is done (if sip.conf is setup properly)

Now let's have a closer look at the network latency:
1) audio source (sound card, microphone)
2) if  the two end terminals are across the public internet then you may 
be looking at 400ms delay (or may be even more) since you are most like 
not owning the link (some form of ATM) not running QoS (won't help too 
much on the public net).
3) Sampling rate: increase this (i.e. decreas the sample interval, so 
choose something like 10ms rather than 20ms) will help in giving a 
feeling of better quality

hope this gives you some tips to diagnose
mohammed

Giovanni Miano wrote:

>I use codec g711u or g711a but comuncation between two sip client
>(XTen lite) have bastard dalay of 0,5 -  1 second
>
>Is it normal ? 
>
>Are there any configuration to solve problem ?
>
>Thanks all
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