Very poor sound quality

Hi Everyone,

Ive got a wierd problem with my asterisk system that Im hoping someone can help with.

In a nutshell when caling from one SIP extension to another with one being on the same LAN as the asterisk box and the other being remote, I get very poor sound quality - drop outs, crackling etc basically making the conversation inaudible for about 50% of the time.

I have an OpenVox card with one FXO and 3 FXS ports. My home DECT phone is connected to one of hte FXS ports. I have a BT phone line connected to the FXO port and a sipgate line registered with Asterisk.

Ive done various testing of the following scenarios and still cant see the problem.

  1. Calls between FXS (DECT phone) and BT line both inbound/outbound are fine
  2. Calls between FXS DECT phone and Sipgate are fine both in and out are fine. A friend called my Sipgate number from his Sipgate account (which Asterisk directed to the FXS port (DECT Phone) and the quality was perfect.
  3. The same friend (and therefore the same internet connection) registered an account on my Asterisk box and called my DECT phone extension (1000). Whilst he could hear me perfectly, the conversation at my end was very broken and barely intelligible.

At first I wondered if this was due to problems with hte internet connections. However, since using both my friends and my own connection directly and via Sipgate cause different results - this seems impossible to my mind.

Im wondering if this is a CODEC issue and intend to look and see what CODECs are being used for the calls - but can anyone offer any advice?



It certainly sounds like a codec issue. If you are running G.711 then I would switch to a lower bandwidth codec like G.729 or GSM.

You should be able to see the codec user by issuing the SIP SHOW CHANNELS command during the call.