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
<[email protected]>
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
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