[Asterisk-Users] iaxtel and jitterbuffer

[Michael George]

On Sat, Aug 28, 2004 at 11:31:48PM -0400, Andrew Kohlsmith wrote:
> On Saturday 28 August 2004 23:01, Michael George wrote:
> > It's a PII 266 (okay, not the fatest system) with 192MB RAM.  X is not
> > running and the Framebuffer has been turned off in /boot/grum/menu.lst.  I
> > have disabled all the servers except for sshd.  I have the latest source
> > from CVS HEAD as of about 30min ago.
> 
> Should be fine.  I ran * on a P90 for a while; it did everything I needed 
> except iLBC.  :-)

Okay, that's a good assurance.  Unfortunately, I have discovered that either
the HDD or the ide controller in that system is bad because it cannot stay up
overnight.  When I stress it with a YaST update, it will die much more
reliably.

Until I can address that issue, I will have to work on my main system.  I'll
just have to take it down to init 3 and stop many of the other server
processes that will still be running.

> > There is no Zap card in this sytem.  The only phone on it is a SIP phone.
> > With it I dial in to digium's 1-700 number.  The audio is better, but still
> > choppy and unacceptible.
> 
> Is your SIP phone doing any kind of silence suppression?  It must be turned 
> off because asterisk takes its timing from the RTP stream and if the phone's 
> not transmitting frames continuously you'll get shitty audio.

Good suggestion and I have double checked it.  I am and was not doing that.  I
think I'd read about it in a Granstream-* page

> Note that latest CVS HEAD looks like they're making provisions for self-timing
> but without a stable clock source it's unlikely to help you.  There are 
> ztdummy modules which use the RTC or certain brands of USB controller to 
> provide adequate timing but ideally you want some kind of Zaptel hardware in 
> there providing a 1000Hz interrupt.

Hmm, I thought that the timing source was only needed for trunking.  I don't
have on on the little box, but I do have a TDM400 (which seems to have faults,
also, but Digium suggested moving the FXO to socket 4, we'll see if that
helps) in the main box, so that should be all set for a timing source.

> Also -- make sure your uplink is acceptable.  First test: make sure there is 
> nothing plugged into your upstream except for your asterisk box and the 
> phone.  Some routers are known to play silly bugger with your packets which 
> naturally wreaks havoc with asterisk.  :-)

The only things on the net when I run the next test will be my main server.
Since I have to test on that with X turned off, I don't even need the SIP
phone active.  In case it might be relevant (there are SO many pieces to this
puzzle that I want to mention all I can think of in case they ring a
trouble-bell in someone's mind...) my router is a Netgear FVS318 acting as a
NAT to my ISP.

> > So even with X11 eliminated the sound is still bad to Digium.  I tried
> > another's 1700 number, and it sounded the same, so it's not something
> > unique to digium and me.
> 
> Perhaps something to do with your upstream or connection to IAXtel.  That's 
> why I'm recommending having nothing but asterisk and the phone on the 
> connection, at least until we nail down what the poor audio's being caused 
> by.

That's possible.  I've checked with my ISP and he said that the connection is
surely half-duplex, but you say that you have 1/2 also and it works fine for
you, so that's not it.  I'm also inquiring about other filters they might have
in place.  I've heard them mention before that they had some cool router
software that could detect traffic patterns usually associated with software
and music piracy and then throttle that traffic into a small part of The Pipe.

I haven't yet heard back, and I'm hoping that isn't the case.  However, if it
*is*, a VPN between offices might help.  IAXtel would be shot, though.
Hoever, if that *is* the case, I can probably convince them to tell their
software to leave me alone on a couple specified ports.

> > Would IAX/GSM be so sensitive to half-duplex that I cannot expect it to
> > work with my ISP only giving me 1/2 duplex service?
> 
> It has nothing to do with IAX or GSM. Stop blaming them.  My upstream is half 
> duplex as well (pretty much anyone on DSL or cable is on a half duplex 
> connection whether they realize it or not).  
> 
> There are many, many people using asterisk every day for long distance and in 
> environments where audio quality is crucial.  Let's stop blaming asterisk and 
> take a good hard look at what's happenning, shall we?

My apologies.  I'm not trying to blame anyone, I love * and except for a
couple glitches that we're working on (with all your gracious help), I'm very
impressed.  My one glitch may be with the hardware, so that's a separate
issue, but the other is trying to figure out this issue with IAX/GSM.

When I ask about sensitivity, I don't mean to be accusatory.  IAX is open and
freely available and GSM is freely usable, and I'm glad.  Sometimes OSS has
its limitations and I am willing to work with them.  So I do not intend any
condescention(sp?), and I'm sorry if it sounds like that.

So, I thank you all for your help and confirmation on thigs which may or may
not be the problem.  Expecially you, Andrew, for I seem to have stepped on a
nerve and yet you still gave me a long and useful reply.  I see there's
another reply further down with some logging recommendations in it.  I'll get
to that one next...

-- 
-M

There are 10 kinds of people in this world:
	Those who can count in binary and those who cannot.