[Asterisk-Users] Zap channel(s), meetme and codecs/licences

[Rich Adamson]

> > > What codec does the Zap channel use by default?
> >
> > None/all. Look in /usr/src/asterisk/configs/zapata.conf.sample and
> > you won't find any reference to codecs.
> 
> I did better; I googled, searched the wiki, and the list archives.
> I find this hint from "zap show channel #":
> Default law: ulaw
> 
> I didn't know if this was just a documentation thing, or if it was
> something that was configurable.
> 
> > > Finally ... if I have a 3way call going, between 1 g729 caller and two
> > > other callers, do I need one or two available licences? (I'm guessing that
> > > zap doesn't do g729, and am wondering if I have an FXO caller and a local
> > > FXS person talking to a VoIP caller using g729, how it would work)
> >
> > Someone else might want to chime in here, but it seems to me (as a non-
> > programmer) that internal handling of voice packets (within *) were
> > primarily slinear or something like that. If two end points can communicate
> > with the same codec, the voice data is simply passed through (no
> > conversion). If one user is g729 only communicating with other users,
> > that g729 user consumes one license instance to convert to whatever
> > the other users might be using. Two g729 users and one g711 user, likely
> > uses two g729 licenses.
> 
> This is my thinking too, but that is what I'm a bit unclear about.
> 
> > Without a better understanding of what devices you are truly trying to
> > use, I don't believe anyone is going to be able to answer your questions
> > relative to zap channels and codec selection/conversion.
> 
> I have a customer that wants to try using a local IAX provider for origination.
> The provider is using g729, and right now I have some digium FXS ports (along
> with an FXO for outbound access). We have some SIP hardphones, and they all
> have the ability to do g729. We will need a licence to do voicemail unless we
> convert files to g729 format. However, to get a phonecall to be handled by
> the Zap phones, I wasn't sure if something was needed or not, and you've
> cleared that up.

Okay, without reading code (which would not be very reliable), the best
reference I can think of for the FXS ports is the chipset doc for the
TDM card. The fxo and fxs modules use the Silicon Labs chipsets, and
that doc indicates ulaw, alaw and 16-bit linear are supported. 

The zaptel.c code has "lineartoalaw" and "lineartoulaw", so best guess
is the TDM card is likely config'ed to linear. Not much help, but at
least a hint.

> I asked about meetme internal format just so I could see if there was a
> possibility to "work around" the need for converters. My worry is that we
> might run out of licences during the test, and that Asterisk would simply
> drop a call if it can't complete one leg of the call, leaving both the caller
> and my customer frustrated. Ideally we could buy n+1 licences, but I am
> trying to get some sort of a feel for what "n" will look like, and how
> practical/viable that is on a much larger scale rollout.

In the asterisk/apps/app_meetme.c, I see stuff like this:
  /* Set it into linear mode (write) */               
  if (ast_set_write_format(chan, AST_FORMAT_SLINEAR) < 0) {
         ast_log(LOG_WARNING, "Unable to set '%s' to write linear mode\n", chan->name);  

So that would suggest the internal structure of meetme is slinear.

For the low cost of testing, I'd just buy two/three licenses or download
the illegal g729 (in US) and do some serious testing. Then you'll know
without relying on opinions, etc. ;)