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


 
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asterisk_list at earth...
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PostPosted: Wed Jul 30, 2014 4:52 am    Post subject: [asterisk-users] SIP trunk gives fuzzy / distorted audio on Reply with quote

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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jthomasdpu at gmail.com
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PostPosted: Thu Jul 31, 2014 10:06 am    Post subject: [asterisk-users] SIP trunk gives fuzzy / distorted audio on Reply with quote

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@earthshod.co.uk (asterisk_list@earthshod.co.uk)> wrote:
Quote:
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 .

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
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asterisk_list at earth...
Guest





PostPosted: Thu Jul 31, 2014 10:09 am    Post subject: [asterisk-users] SIP trunk gives fuzzy / distorted audio on Reply with quote

On Thursday 31 Jul 2014, James Thomas wrote:
Quote:
Is the quality the same incoming from mobile as outgoing to mobile?

It's a one-way trunk (outgoing only).

Anyway, I've now fixed it, with help from the trunk provider. Details to
follow in a separate message.

--
AJS

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

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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