asterisk users - Nov 2005 - asterisk sample size adjustment

Is there any way to adjust the sample size asterisk uses for VoIP
codecs?  From what I have gathered it uses a fixed 20ms sample size for
all codecs.  While some require at least this, some can be configured
for less.  This results in more overhead, but can be tweaked to provide
more efficient transfer on the backbone links due to ATM framing
properties.

If anyone has any information on how to change the sample size I would
appreciate hearing about it, because I cant find anything with google.
Asterisk is a particularly bad google term since it is used as a
footnote market, wildcard, etc :P


-- 
Trixter http://www.0xdecafbad.com     Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group
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hi,

723 needs 30ms + 5ms forward. 729 needs 10ms + 5ms forward. The 
'standard' for 711 is actually 6ms (48 bytes). This would have to be 
done per channel (or per codec), but I am not sure wherever Asterisk 
allow per codec size or run's with one static size???

Jan

trixter aka Bret McDanel wrote:
>Is there any way to adjust the sample size asterisk uses for VoIP
>codecs?  From what I have gathered it uses a fixed 20ms sample size for
>all codecs.  While some require at least this, some can be configured
>for less.  This results in more overhead, but can be tweaked to provide
>more efficient transfer on the backbone links due to ATM framing
>properties.
>
>If anyone has any information on how to change the sample size I would
>appreciate hearing about it, because I cant find anything with google.
>Asterisk is a particularly bad google term since it is used as a
>footnote market, wildcard, etc :P
>
>
>  
>
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>> hi,
>> 
>> 723 needs 30ms + 5ms forward. 729 needs 10ms + 5ms forward. The
>> 'standard' for 711 is actually 6ms (48 bytes). This would have
to be
>> done per channel (or per codec), but I am not sure wherever Asterisk
>> allow per codec size or run's with one static size???
There is a patch in Mantis, bugid 5162, I think, that allows for
changing
the packetization size.  It surrently works on  global/users/peers, but 
is not per codec.

Bug 5162 updates the  core RTP code and adds the packetization options
to chan_sip.  There is a seperate patch to add packetization to the 
ooH323c version of chan_h323.
> Yeah and the gsm one usually uses 20ms.  A per codec way would be
ideal,
> I implied that in my original post, or something.  I just thought that
> for network tuning purposes it might be nice to actually have that
> ability.  Less padding more payload on the ATM cells makes for a more
> efficient network :)
> AFAIK all the sample sizes are hardcoded, but figured I would ask and
> see if anyone knew of a way short of altering the code to adjust this.
> While altering the code is usually not a problem, it makes updating a
> little more work and stuff..
Dan

>> >> hi,
>> >> 
>> >> 723 needs 30ms + 5ms forward. 729 needs 10ms + 5ms forward.
The
>> >> 'standard' for 711 is actually 6ms (48 bytes). This
would have to
be
>> >> done per channel (or per codec), but I am not sure wherever
Asterisk
>> >> allow per codec size or run's with one static size???
>> 
>> There is a patch in Mantis, bugid 5162, I think, that allows for
>> changing the packetization size.  It surrently works on 
>> global/users/peers, but is not per codec.
> 
> eeps that can break stuff or at least cause performance problems with
> mixed codecs :(
Six in one hand, half a dozen in the other.  We had a number of devices
with less than ideal packetization defaults.  The patch allowed
us to force a common denominator and works well.
> At least its a start..  personal preference I dont like stuff
hardcoded
> unless it has to be, but that is just me.
Indeed it is.  At the time it was posted someone suggested that adding
an option to the codec selection process would be better, but no one
added the code (allow=ulaw:30,g729:40 type of thing).

Dan

On Mon, 2005-11-14 at 10:17 -0800, Dan Austin wrote:
> Indeed it is.  At the time it was posted someone suggested that adding
> an option to the codec selection process would be better, but no one
> added the code (allow=ulaw:30,g729:40 type of thing).
> 
I think that is a better way to go so that you can tweak things specific
for your system.  Someone on DSL with PPPoE has to contend not only with
ATM framing but also with PPPoE framing (6 bytes per packet iirc) and so
on.  Typically though if you are on DSL, especially with PPPoE you are a
home user, and the calling volume should be low enough that you can
tolerate a little inefficiency, if they are tuned for straight ATM that
would prolly work best for the majority of people the majority of the
time.


-- 
Trixter http://www.0xdecafbad.com     Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group
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I have a 4 port brooktrout PCI E1/T1 blade card (MPAC 1200) that was 
used for some Sun carrier server. Will Asterisk support this card? Has 
anyone used this successfully before?

Thanks! Stephen
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Stephen Arulraj wrote:
> I have a 4 port brooktrout PCI E1/T1 blade card (*MPAC 1200*) that was 
> used for some Sun carrier server. Will Asterisk support this card? Has 
> anyone used this successfully before?
The only hardware supported is the list here: 
http://www.asterisk.org/hardware

Sangoma (some cards) are also supported by Sangoma with their Zaptel 
compatable drivers.