[asterisk-users] SIP trunk gives fuzzy / distorted audio on mobiles, OK on fixed lines

[A J Stiles]

I'm having a problem with a new SIP trunk.

Calls within the UK to fixed lines are fine, but calls to mobiles have 
noticeably poorer audio quality.

I thought it might have been a codec issue; we have used G.726 for internal 
and external calls  (over primary ISDN and GSM).  So I tried allowing "alaw",  
(G.711 A-law)  which is the native codec used within the PSTN in this country, 
but this made no improvement.

We had
  disallow=all
  allow=g726

in the [general] section of sip.conf.  In the section for one of the phones, I 
added
  allow=alaw
and then inserted
  Set(SIP_CODEC=alaw)
in the relevant part of extensions.conf.  For good measure, I also added
  NoOp(Codec was ${SIP_CODEC})
in the "h" extension.  The messages in the Asterisk CLI appeared to show that 
the audio codec was correctly being set to "alaw", and on hangup I got "Codec 
was alaw", but there was no improvement to the sound quality.

Is there something I am doing wrong, or do I need to get in touch with our SIP 
trunk provider?

-- 
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .