asterisk users - Jan 2005 - Enhancing performance and utility of an Asterisk machine

Hi, some questions/comments about performance/utility of * and * hardware

I've been reading this list for a few weeks and I think I have
compiled the better feelings of the users.
please correct me if I'm wrong, still learning * ....
Will be nice to see something like this in a wiki.
After being flamed and corrected I will repost "clean" data.

1- Transcoding is the process of converting from one codec to another.
Example from G.723.1 to G.729.
2- It does not matter if you're doing voicemail or a call. If one of
the ends does not "speak" the same codec, transcoding is involved.
3- Transcoding is very CPU intensive and should be avoided when possible.
4- DSP based cards will improve * performance by offloading work from the CPU.
5- If you configure the SIP phones to use the same codec (G.729) .
then no transcoding is involved when they talk to each other.
6- If you're doing VoIP to POTS/T1/E1 you're doing transcoding.
6a-If youre using G.711 as the codec and doing VoIP to POTS/T1/E1
you're NOT doing transcoding?
6b- More than 50 calls VoIP to POTS/T1/E1 will kill an * box due to
excesive transcoding??????
6bb- unless using quad machines, plenty of RAM and DSP cards?????

File Codecs
What codec should I use to save my voicemail and IVR prompts?

Hardware
DTMF generation and cut-through detection are features you must get on a card
Integrated DSP Echo Cancellation is a must.
any other features that I should go out and buy? * compatible hardware
of course.

Another off-this-topic question
i read that the TDM cards from Digium are having some problems. Im
just saying what i read. I have no intention to discuss the problems.
But if this is true, then * VoIP players like Nufone with their 80+ *
server farms.....are using what E1/T1/DS3/etc * compatible hardware?
are Digium and VoiceTronix the only * compatible hardware? I googled a
lot and found "will work with opensource apps" but do not explicitly
say Asterisk.
(Shido can you shed some light on this?)

Since I am an end user/Unix sysadmin/VoIP learner I think I will ask
the forum to give me the task of doing a wiki or update existing wikis
and voip-info.org (when we all agree of course).

I bought the * book from signate. It's very good but lack some
things...I'm waiting for the next revision (if any).
-- 

-------------------------------------------
Erick Perez
Linux User 376588
http://counter.li.org/  (Get counted!!!)

Erick Perez wrote:
> Hi, some questions/comments about performance/utility of * and * hardware
> 
> I've been reading this list for a few weeks and I think I have
> compiled the better feelings of the users.
> please correct me if I'm wrong, still learning * ....
> Will be nice to see something like this in a wiki.
> After being flamed and corrected I will repost "clean" data.
> 
> 1- Transcoding is the process of converting from one codec to another.
> Example from G.723.1 to G.729.
Yes.
> 2- It does not matter if you're doing voicemail or a call. If one of
> the ends does not "speak" the same codec, transcoding is
involved.
Yes. It will depend on the codecs you've allowed for the two endpoints.
> 3- Transcoding is very CPU intensive and should be avoided when possible.
Yes.
> 4- DSP based cards will improve * performance by offloading work from the
CPU.
Yes, I suppose. But none of the Digium boards have DSP. Some others like 
Dialogic boards do have DSPs, but they aren't supported directly -- for
Dialogic
  stuff you have to purchase the drivers from Digium, IIRC.
> 5- If you configure the SIP phones to use the same codec (G.729) .
> then no transcoding is involved when they talk to each other.
Yes.
> 6- If you're doing VoIP to POTS/T1/E1 you're doing transcoding.
Not necessarily -- if your SIP/IAX connection uses the alaw/ulaw codec then none
is involved.

> 6a-If youre using G.711 as the codec and doing VoIP to POTS/T1/E1
> you're NOT doing transcoding?
As above. See 
http://lists.digium.com/pipermail/asterisk-users/2004-November/070998.html for 
some more info.
> 6b- More than 50 calls VoIP to POTS/T1/E1 will kill an * box due to
> excesive transcoding??????
> 6bb- unless using quad machines, plenty of RAM and DSP cards?????
> 
The max number of calls is something that's not really a hard-and-fast
number.
There's been numerous discussions regarding *'s "upper limit"
of connections,
check the previous postings to this list for more info.

Again, it depends on a lot of different variables, like what the * box is doing,
how many AGIs it's running, whether or not you're doing extensive DB
stuff.

Generally speaking, you won't go wrong with a box that's beefy in terms
of RAM.
> File Codecs
> What codec should I use to save my voicemail and IVR prompts?
> 
IVR prompts usually are done in GSM format, although I've used WAV files
without
any problems.

For voicemails, in my opinion it's more of an end-user question -- if
you're
emailing the voicemails, the recipient may not have an application that can play
it back. Most Windows users will be using the media player or sound recorder to 
play back audio files, so if a majority of them are on the windows platform it 
may be easier to record your VMs in WAV format.

Although .WAV files do take up more HD space than .GSM files.
> Hardware
> DTMF generation and cut-through detection are features you must get on a
card
> Integrated DSP Echo Cancellation is a must.
> any other features that I should go out and buy? * compatible hardware
> of course.
> 
Not sure about that one ;)
> Another off-this-topic question
> i read that the TDM cards from Digium are having some problems. Im
> just saying what i read. I have no intention to discuss the problems.
> But if this is true, then * VoIP players like Nufone with their 80+ *
> server farms.....are using what E1/T1/DS3/etc * compatible hardware?
Perhaps you could send them an email to ask?
> are Digium and VoiceTronix the only * compatible hardware? I googled a
> lot and found "will work with opensource apps" but do not
explicitly
> say Asterisk.
> (Shido can you shed some light on this?)
For the most part, yes. However there are some ISDN BRI cards that are 
supported, read http://www.voip-info.org/wiki-Asterisk+CAPI+channels for a 
start. There's also a wiki page about using ISDN4Linux and modems compatible
with it at http://www.voip-info.org/wiki-Asterisk+ISDN4Linux

Flynn