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We have a dedicated Asterisk 1.8 server running as a front end SIP server for authenticated and internal users only, using SDSL 2MB/2MB, in the UK. The provider of our internet also is the provider of our outgoing SIP(so there is no internal hop), and intotal there are a few hops which are mostly LAN routers with a total of 30ms to the sip server.

  • Everything works fine in the office. No delays.
  • Everything works fine using extension dialling from two outside lines. Ie bot extensions are at home using home br= broadband of 512mb.. there is roughly 100-200ms delay on both channels- which is fine.

The problem occurs when any extension, office, or home user dials out using a Trunk. Out Main one or even voipcheap.

The outgoing audio get to the "mobile" phone for test sake in not more than 250ms- that's perfect. But when the person on the "mobile" replies, the audio returns after 500-750ms.

I have not timed anything exactly but that's what it feels like. I tried running channel stats in CLI - jitter is 0, lost is 0

I mainly use aLaw codec, but i started forcing other like g729, g726 - Jitter is off, but even when its on it makes no difference (to my ear) the returning delay is still there.All these codecs are supported by my SIP servers, and all SIP clients connect using aLaw codec, and the aLaw has the lest translation time.. but I am not sure if it actually translating it somewhere along the line, at should just pass through.

I do not know how to debug/trace the returning channel issue. Where is the delay, what can i do to try and resolve this.

The problem is that people start talking over each other because the delay out and in is different and confuses people, it would be better if the delay was the same. but i think 500ms is too much for the time and effort we spent setting up dedicated internet and server.

--Edit my sip settings

host=[voiproip]
username=[myusernme]
secret=[secret]
type=peer
context=from-trunk-sip-GREY

extension setting

[16]
deny=0.0.0.0/0.0.0.0
secret=[secret]
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=udp
encryption=no
callgroup=
pickupgroup=
dial=SIP/16
mailbox=16@default
permit=0.0.0.0/0.0.0.0
callerid=device <16>
callcounter=yes
faxdetect=no
cc_monitor_policy=generic

resolve nothing special.

Does anybody have some advice?

enter image description here

share|improve this question
    
Please post relevant contents of /etc/asterisk/sip.conf, as well as /etc/resolv.conf. –  lorsungcu Jan 20 '12 at 5:34
    
Hi I am not sure what you want to see from resolv and sip conf? But i put my sip entry there and resolve just has nameserver of my router. Thats it –  ppumkin Jan 20 '12 at 10:33
    
just wanted to add that i am using the freepbx distro. –  ppumkin Jan 20 '12 at 10:43
    
In future please add requests for additional information as comments on the question. –  sam Jan 20 '12 at 10:55
1  
I did all sorts of test of everykind. End of the day the IP Phone it self was the culpret! It was a more expensive one- when he hooked up el-cheopo there is no delay(or acceptable enough to pass for normal talking) Not sure what exatcly caused the issue but we dumped those phone (grandstream something with lcd screen) - even sip over 3g work a blooming treat! –  ppumkin Feb 17 '13 at 20:27

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