[Asterisk-Users] Problem with Voice Quality

[Leo Ann Boon]

mkumar at mantragroup.com wrote:

>Hi All,
>
>	We are making a VOIP application for Mobiles (PDA's) and we are using Asterisk
>for it. We have a setup consisting of both SER and Asterisk. SER acts as a SIP
>router and routes everything to Asterisk. We also have rtpproxy for SER. Our
>packet delivery from clients (Mobiles, PDA's) is inconsistent and ranges
>between 10 to 60 ms delay but the average is near to 20 ms. We use SIP. How can
>we solve this problem, is there any setting at the server end to handle this,
>as clients have very limited resources we have to manage this at the server
>end, please tell me how can I do this?
>  
>
Sanity check:
a) What kind of connectivity - WiFi or GPRS, 3G etc?
b) What's the ping time to your clients?

 From your ping values, I think you're running over WiFi? Always bear in 
mind, anything going over the air will have delay. It's pretty much out 
of your control. Some things that you can do to smooth it out:
a. Use a low bit-rate codec that does PLC
b. Use a large jitterbuffer
c. Send more than 1 frame per packet. I don't think stock Asterisk can 
do this, but I remember there was mention of a patch for it.
d. If you're using WiFi, you might want to check your cell planning. 
WiFi handover is a bitch for VoIP.

a. and b. will pretty much depend on what your PDA softphone is capable of.

Leo