Hi,

I am running asterisk 1.2.12 on Buffalo WHR-G54S 4MB flash device. I have two inbound voip provider registered. If I get a call from freedigits (asterix pbx voip provider), my call is very clear and I can see that the cpu is usage is low. But I get a call from voicestick then my cpu bumps up to 98% and my call quality is very choppy.

I only setup the device to use ulaw. So I am guessing both provider has to use this codec. Does any one knwo what is the difference and how can I make the voicestick connection to behave like the freedigits ?.

Thanks in advance.

Sam.

Hello,
I can't answer your question, but I have the same problem with VoIP provider Eutelia.
When I make sip calls between two internal phones CPU usage in around 40%, when I get external call, CPU raise to 100%.

(Last edited by vinx on 10 May 2007, 09:47)

probably voipstick and eutelia uses diffrent codec that you use beetwen your ip phone and router, so there is need to transcode it. Check the codesc used by both providers.

Do U know how to check provider's codec?

hmmm you can easy check if it's g.711a/u (uncompressed) or any other by the transfer rate - the g.711 (64kbit/s) +  SIP signalization should get approx. 10kB/s on both direction. If it is less (half or more) that mean there is other codec.
AFAIK you can define what codec can be used with every trunk so you can force to g.711 and then it shouldn't transcode anything.

I know Eutelia support the following codecs:

G711alaw, G711ulaw, G723ar53, G723ar63, G723r53, G723r63, G726r16, G726r24, G726r32, G728, G729br8,  G729r8, GSM-EFR, GSM-FR.

How to understand what's used during a call, and eventually, is possible to force a particular codec?

I tried enabling SIP DEBUG, but I cannot understand what's the codec used.

as far as I remember you can define allowed codecs in trunk definition:

disallow=all
allow=ulaw # for g.711u
allow=alaw # for g.711a

but you should look into the asterisk documentation for details.

This is what I have right now in the trunk, but the problem still persist.

Check to make sure you've disabled DTMF dialing.

thankU, I solved it by adding "dtmfmode=rfc2833" on incoming trunk.