asterisk users - Jul 2014 - SIP trunk gives fuzzy / distorted audio on mobiles, OK on fixed lines

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 .

Is the quality the same incoming from mobile as outgoing to mobile?


On Wed, Jul 30, 2014 at 4:51 AM, A J Stiles <asterisk_list at
earthshod.co.uk>
wrote:
> 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 .
>
> --
> _____________________________________________________________________
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I have now fixed this issue, and am posting this for the benefit of anyone else 
who may be suffering with a similar problem.

It was, as I suspected all along, a subtle misconfiguration at this end.

The fix was to give the SIP trunk its own configuration stanza in sip.conf as 
follows;

[sip_trunk_outbound]
type=peer
host=provider.sld.cc
disallow=all
allow=alaw

and replace all instances of

Dial(SIP/provider.sld.cc/44${EXTEN:1})
with
Dial(SIP/sip_trunk_outbound/44${EXTEN:1})

In the absence of that important little stanza, the [general] settings were 
applying to the ad-hoc SIP endpoint; meaning that even in spite of explicitly 
setting the outbound SIP codec, Asterisk was insisting to use G726.


No sooner had I worked this out, than the SIP trunk provider e-mailed 
basically to confirm my thinking.


The moral of this story:  Never trust a configuration file written by someone 
else, no matter how close it was to working  ;)


-- 
AJS

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