asterisk users - Sep 2004 - CODECs and sip.conf and voice quality

Group,

	Just want to share with the group my recent findings regarding
CODECs/Vocoders and the effect it has had on voice quality and the
intermittent noise and breakup problem I have which I mentioned in a
previous emailing with the u-law CODEC. Calls again are placed through a
SIP phone to a TDM400P to the PSTN. A good reference on the reasoning
behind the selection of a CODEC was found in the archive and was of
great help. See link: 
 http://lists.digium.com/pipermail/asterisk-users/2004-May/045957.html
 
	After some difficulty understanding how they are defined in sip.conf
and in the phone itself. I have come to the following conclusions:

	The settings in the sip.conf override anything set on the phones
themselves. I expect the SIP phone CODEC settings are only for direct
phone to phone conversations without and asterisk server getting
involved.

	Here's my configuration. I have Grandstream and SNOM phones and would
like to have the GS and * negotiate through a list as follows:

So based on the descriptions for sip.conf, I put the order of precedence
in the [general] area of sip.conf as follows

ilbc
gsm
g729
u-law

In this config, the GS phones communicate using ILBC and SNOM phones
communicate with gsm. SNOM's do not support ILBC therefore GSM was the
result of the negotiation. The settings in the phone themselves had no
impact as mentioned.

Now if I set the CODEC in the user section for the phone, the general
settings still control the CODEC selection. I had to go and individually
set the CODEC in the each phone definition in sip.conf and comment out
those that I placed in the [general] section. In this way, I could
choose which phone used u-law, gsm and ilbc, and it worked! However, the
order can't be forced as when defining in the general section.

I am currently configured with a 7/29 CVS version of Asterisk. Perhaps
things have changed with the latest release. If anyone knows, please
share. I may be updating our version sometime soon.

I could not get g729 working. Asterisk complains about it. I suspect
that the I may need to specifically install it. I was hoping to use it
in a non-pass through mode (asterisk translates it) for testing purposes
only since it is a patented and liscensed CODEC.

Another Caveat:
Transfer does not work using the # key with the ILBC CODEC on the GS
phones. I can transfer only with the transfer button. I have asterisk in
the loop doing call supervision since I have the tT option set in the
dial command and canreinvite=yes for the SIP phones. Anyone else have
this problem?

How can I force a general CODEC precedence and a user precedence for
those phones that I want to override the general order? Can it be done?

As for the voice quality and the intermittent noise.

        GSM still had the problem with intermittent noise as tested on
        the SNOM 200. Voice quality itself is good. I felt it was
        clearer, perhaps better than u-law
        
        ILBC intermittent noise remains as tested with GS BT100. Not
        much MF signalling heard, but break up seems to be more
        persistent than with u-law. Voice quality is good. I felt it was
        also clear as if not better than u-law
        
Next step is to upgrade our * server to a dual processor platform with
more RAM to see if that helps. The current system has 1G RAM and 1Ghz
clock.

	Any comments on my approach are welcome. Thanks in advance.

ILBCin' You,
Mike Meyer

> -----Original Message-----
> From: Mike Meyer [mailto:mjmeyer@gendesign.com] 
> Sent: Tuesday, September 28, 2004 1:07 PM
> To: Asterisk Users Group
> Subject: [Asterisk-Users] CODECs and sip.conf and voice quality
> 
> 
> Another Caveat:
> Transfer does not work using the # key with the ILBC CODEC on 
> the GS phones. I can transfer only with the transfer button. 
> I have asterisk in the loop doing call supervision since I 
> have the tT option set in the dial command and 
> canreinvite=yes for the SIP phones. Anyone else have this problem?
> 
Shouldn't canreinvite be set to no to keep the phones from 
reinviting? I agree that with the tT flags * should still be
in the path, but I was just curious.

Robert,

	RE: The reinvite=yes option. 

	My answer is that I am not sure, but I think you are right. Based on
the descriptions I have found ...

a)http://www.voip-info.org/tiki-index.php?page=Asterisk:%20Letting%20SIP%20clients%20connect%20directly
b)http://www.voip-info.org/wiki-Asterisk+sip+canreinvite
 
it sounds like the Asterisk is still in the signalling path with it set
to yes. From previous investigation to support call parking,

c)http://www.voip-info.org/tiki-index.php?page=Asterisk%20call%20parking

documentation indicates it is to be set to yes for that to work.
Probably for the same reason though, so that call supervision can detect
the # to transfer and must be kept in the media path.

In my case, I am probably getting away with it set to yes since the dial
has tT option set and I am connected a call between a SIP phone and a
TDM card so it won't reinvite in either case.

I can't remember the reason that I had now for setting it to yes. I may
have just gotten totally confused. Easy to do. All these options and
dependencies keep my head spinning!

In final; I tested ilbc with canreinvite=no just for kicks. Transfer
still does not work with the #. Had to use the transfer button on the GS
phone.

Thanks again for your comment,
Mike


>Date: Tue, 28 Sep 2004 13:24:53 -0400
>From: "Robert Jackson" <RobertJ@promedicalinc.com>
>Subject: RE: [Asterisk-Users] CODECs and sip.conf and voice quality
>To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>        <asterisk-users@lists.digium.com>
>Message-ID:
>       
><2BE8436A70E4AF4C9663B9E98CD52B763880C8@promed_2.promedicalinc.com>
>Content-Type: text/plain;       charset="US-ASCII"
>
>> -----Original Message-----
>> From: Mike Meyer [mailto:mjmeyer@gendesign.com] 
>> Sent: Tuesday, September 28, 2004 1:07 PM
>> To: Asterisk Users Group
>> Subject: [Asterisk-Users] CODECs and sip.conf and voice quality
>> 
>> 
>> Another Caveat:
>> Transfer does not work using the # key with the ILBC CODEC on 
>> the GS phones. I can transfer only with the transfer button. 
>> I have asterisk in the loop doing call supervision since I 
>> have the tT option set in the dial command and 
>> canreinvite=yes for the SIP phones. Anyone else have this problem?
>> 
>
>Shouldn't canreinvite be set to no to keep the phones from 
>reinviting? I agree that with the tT flags * should still be
>in the path, but I was just curious.

Hello,

On Tue, 28 Sep 2004 16:43:09 -0500, Mike Meyer <mjmeyer@gendesign.com>
wrote:
> Robert,
[snip]
> In final; I tested ilbc with canreinvite=no just for kicks. Transfer
> still does not work with the #. Had to use the transfer button on the GS
> phone.
Try changing dtmfmode in the phone and in sip.conf. Inband DTMF will
not work using compressed codecs. Try INFO or RFC2833.

If you use tT in Dial, asterisk will stay in the path: canreinvite
will be ignored.

-- 
Nicol?s Gudi?o
Buenos Aires - Argentina