[Asterisk-Users] Re: IAX Inbound Sound Quality

[Gene Willingham]

This may not be the issue, but it is definitely worth looking into.  

You should verify that your equipment is running in FULL Duplex mode.  I had
a problem were the equipment was not Auto Sensing FULL duplex, and it
created some weird call quality issues.  I had to force the equipment
(manually) to full duplex.

Gene


-----Original Message-----

   3. Re: IAX Inbound Sound Quality (Mark Eissler)
   6. Re: Re: IAX Inbound Sound Quality (Mark Eissler)

Message: 3
Date: Mon, 24 Jan 2005 09:49:06 -0500
From: Mark Eissler <mark at mixtur.com>
Subject: Re: [Asterisk-Users] IAX Inbound Sound Quality
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at lists.digium.com>, Brian Dingman
<bdingman at gmail.com>
Message-ID: <18B1C660-6E17-11D9-9EE8-0030654681D6 at mixtur.com>
Content-Type: text/plain; charset=US-ASCII; format=flowed

Try changing to a less-bandwidth intensive codec (like GSM) and see 
what happens.

-mark

On Jan 21, 2005, at 7:08 PM, Brian Dingman wrote:

> I have a couple of DID's through VP Connect and have been having sound
> quality issues on incoming calls. During the call, the calling parties
> voice sometimes sound like it is crackling, in other words it is not
> very crisp. I would liken it to listening to a radio with a blown
> speaker. This sound defect comes and goes throughout the call. The
> other person is always audible but it just isn't as crisp and clear as
> when I make outgoing calls over IAX. The other party does not hear any
> audio defects.
>
> Anybody have any suggestions on tweaking this? Or has anyone
> experienced the like?
>
> Running * 1.0.3 on an AMD 1700 with 512 MB of RAM (Red Hat 9). I am
> the only user currently on the system. I am connecting with their IAX
> server using ULAW and my SIP phone is also using ULAW (Sipura 2000).
>
> Thanks,
> Brian
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--
Mark Eissler, mark at mixtur.com
Mixtur Interactive, Inc. - at - http://www.mixtur.com


------------------------------

Message: 6
Date: Mon, 24 Jan 2005 09:57:51 -0500
From: Mark Eissler <mark at mixtur.com>
Subject: Re: [Asterisk-Users] Re: IAX Inbound Sound Quality
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users at lists.digium.com>
Message-ID: <51B6D5FB-6E18-11D9-9EE8-0030654681D6 at mixtur.com>
Content-Type: text/plain; charset=US-ASCII; format=flowed


On Jan 22, 2005, at 10:49 PM, Michael Graves wrote:
>
> I notice that all four of my IAX2 based termination providers send
> incomming calls in trunking mode. You can tells since the command IAX2
> Show Registry reports all the connections to port 8617. This is
> something that is determined at their end. In trunk mode I beleive that
> the jitter buffer is not effective.
>

IIRC the jitter buffer is currently broken in trunk mode and should be 
turned off.

http://www.voip-info.org/wiki-Asterisk+bandwidth+iax2

An alternative for testing is to set trunk=no in iax.conf. I've had to 
do that for my VPC trunks because I've also found that outbound faxing 
seems to be broken with trunking turned on (at least to VPC).

> FWIW, I had similar problems with VPC so I switched to Sixtel.net. No
> such problems anymore.
>

VPC must still be using quite a lot of custom code or routing their 
calls in some weird way because I've found two problems with them so 
far while using IAX2:

1) The fax problem mentioned above.

2) Inbound DTMF is quite broken. (They are working on a fix and said it 
would be at least 30 days...but then in December they said it would 
take 2 weeks...). What a drag.

-mark

--
Mark Eissler, mark at mixtur.com
Mixtur Interactive, Inc. - at - http://www.mixtur.com