Hi,

In one side, I have a 512 Kbps link. In the other a 128 Kbps link.
With IPTRAF I see the amount of rate the port 4569 consumes. In ANY codec, this is not less than 30 Kbps (30 Kbps for up and 30 Kbps for down).


The voice is perfect when the use of internet in company that have 128 K is low...

I have a couple of PLANETs VIP 400 (4 x FXO H323 gateway) in other companys and, using g729 (without asterisk), they consumes 20 kbps with a very good voice quality (ps: they have silence supression and voice audio detection).

Best regards,

Alexandre


Miguel Ruiz Velasco Sobrino wrote:
Well, I don't know how to tune it more, it connects at about that rate in a 
mediocre
rural landline.

ILBC uses samples of 30ms, so if you set the trunkfreq set to 20 you will be 
using more
of the necesary scarce bandwidth AND dropping sample info in each frame, thus 
making
audio choppy and unclear.

Make shure to disallow all codecs and then allow only ILBC or lpc10 (search for 
it in
iax.conf), use the one that gives better results in your case.

lpc10 uses 20ms samples, so you can not allow at the same time both!!.

if you cannot connect at least at about 20kbps, your audio will suffer, and 
from mi
experience, at 16kbps the audio becomes so distored that is very difficult to 
understand
the other party.



--- [EMAIL PROTECTED] wrote:
Message: 1
Date: Wed, 19 Jan 2005 15:05:53 -0200
From: "alexandre::aldeia digital" <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Re: Asterisk bandwidth tuning?
To: Asterisk Users Mailing List - Non-Commercial Discussion
        <asterisk-users@lists.digium.com>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

18 - 22 Kbps .... my dream!

I have asterisk -> INTERNET -> asterisk connection with IAX2 and I try iLBC, gsm, g729 and speex and the minimun bandwidth was 38 Kbps for 1 channel.

What the parameters do you set to have this rate ???

Thank you.


Miguel Ruiz Velasco Sobrino wrote:

I have an installation that connects in a [very] good day at 22kbps, but the normal is
about 18kbps.
I use de ILBC codec, and also change in iax.conf the
trunkfreq = 20
to trunkfreq = 30


It works, you can understand well the other person, but don't expect miracles 
or an
outstanding sound quality.




Dear Dan;

Thanks alot for your kindly reply.

Well, what u advise us to use if the bandwidth is about 22kbps (dial up
connection in very old countries)?


GSM Codec is 13k plus overhead. That may work?


No way. GSM is 13.2kbps, and with Asterisk's hardcoded 20ms packetization, this gives 29200bps with RTP-based protocols or 26000bps with IAX, and as long as Asterisk doesn't support silence suppression, this needs to be full duplex, and I doubt you don't get that from a modem. If we could add silence suppression, we could do with half duplex, effectively saving half the bandwidth. in addition, if we could increase the packetization to something like 50 or 100ms for extreme use (like dial-up), we could end up with a lot less. Say, using 100ms slicing with IAX and speex at 6.3kbps, we'll end up with effective bandwidth use of 8860bps plus ethernet. now that's something you can brag about....


roy


Miguel Ruiz Velasco



=====
Miguel Ruiz Velasco

Version: OpenKeyServer v1.2
Comment: Extracted from belgium.keyserver.net
Signature: 0x59831109




__________________________________ Do you Yahoo!? Yahoo! Mail - You care about security. So do we. http://promotions.yahoo.com/new_mail
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to