[Asterisk-Users] RFD: With echo and other distortion, can ulaw/alaw line quality ever be good enough for faxing?

Wed May 5 08:35:04 MST 2004

we use cisco ata186 units using sip to talk to asterisk in ulaw.  there
is only one machine plugged into it but they fax @ 14400 with no
problems.

On Wed, 2004-05-05 at 19:35, Darren Nickerson wrote:
> Folks,
> 
> The silence was deafening ... I had a few private replies but overall I'd
> have to conclude that most people on this list aren't interested in faxing
> thru Asterisk. You're all probably jazzed about VoIP and fax is forgotten
> for now ;-)
> 
> We have learned the answer to _one_ of the questions I asked below - the
> Adit channel bank's gain controls can be usefully applied to adjust each of
> the incoming telephone lines independently, and by combining the Adit's gain
> setting with Asterisk's, fun and good things can be made to happen ;-)
> 
> Another interesting observation is that echo can be seen visually when using
> ztmonitor. Muting one side of the call (the RX side), and speaking from my
> handset I can see my voice appear stongly on the TX side, and then I can see
> the echo appearing on the RX side at an attenuated level, but it follows my
> voice perfectly ;-)  I spend half a day talking to myself, and I have to
> say, for the first time in ages I was running out of things to say! :-)
> 
> In other interesting news, updating zapel from CVS seems to have made things
> significantly better. I had CVS from about 4 weeks ago previously, so
> perhaps something was amiss. We can actually get faxes transmitting now, and
> although they fail about 30% of the time, this is definitely progress.
> 
> Another data point, is that MARK3 echo cancellation is audibly worse under
> our particular conditions. We're using MARK2 with the aggressive option
> enabled, and while occasional flares of distorted (but attenuated) echo can
> be heard, it's much easier to have conversations with customers! I'm
> disappointed nobody commented on the echo cancelation options in the
> source... I can only assume nobody actually knows the origins of any of them
> any more, and the other ones are mostly anachronisms that are no longer
> useful, and MARK2 is the clear winner.
> 
> Unfortunately, the news from Digium's technical support is not good. Here's
> what they had to say in response to our inquiry about faxing:
> 
> "Faxing in asterisk is only at varied levels of support and is
> definitely not at a production level.  However, if you wish to work on
> this we wouldn't mind at all :-).  The source code is of course free
> to your disposal."
> 
> I'm not sure I understand that this support rep is saying to me ... we've
> had much success faxing from one PRI to another with them both connected to
> Asterisk via a TE400P. Asterisk doesn't even break a sweat. We're rarely
> getting the theoretical maximum of 33,600 that we can get with a T1 loopback
> though, so something's slightly amiss, but it's only when we try to go out
> to the PSTN that things start to fall apart badly.
> 
> -Darren
> 
> --
> Darren Nickerson
> Senior Sales & Support Engineer
> iFax Solutions, Inc. www.ifax.com
> darren.nickerson at ifax.com
> +1.215.438.4638
> +1.215.243.8335 (fax)
> 
> ----- Original Message ----- 
> From: "Darren Nickerson" <darren.nickerson at ifax.com>
> To: <asterisk-users at lists.digium.com>
> Sent: Tuesday, May 04, 2004 12:35 AM
> Subject: [Asterisk-Users] RFD: With echo and other distortion, can ulaw/alaw
> line quality ever be good enough for faxing?
> 
> 
> > Folks,
> >
> > I've been following recent discussions regarding echo and echo training
> with
> > much interest, since it's a problem we've never been able to eliminate
> here.
> >
> > We're facing two challenges presently, and they may be related (or not):
> >
> > a) Cisco 7960s in the office here echo back to our staff, but the customer
> > hears decent sound.
> > b) We've yet to be able to pass faxes through asterisk, despite trying a
> > number of approaches
> >
> > A bit about our setup:
> >
> > Two analog lines come into the office, into the FXO card of an CAC ADIT
> 600.
> > These are 'connected' to the TDM interface of the ADIT, and go out to
> > Asterisk via 2 channels of a a 24-channel fxs_ks line into a port of a
> > TE405P. We have two other ports on the TE405P occupied by digital fax
> > boards, connected to asterisk via AT&T 5ESS PRI. And of course there's the
> > 7960 handsets, connected directly to a switch at 100MBps (fdx).
> >
> > Challenge a)
> > =========
> > In terms of looking into a), we started with:
> >
> > http://asterisk.sohoskyway.net/Asterisk_Doc/current/docs-html/x939.html
> >
> > Clearly some of the details are outdated, since the echo canceller options
> > live in zconfig.h now, not a Makefile, but it's a decent starting point.
> > Which gives rise to my first question - which echo canceler should we be
> > using? We have Steve, Steve 2, and Mark 1-3, with an aggressive option
> with
> > 3 that could make calls scratchy ;-)
> >
> > Is there any conventional wisdom here?
> >
> > Next question ... we have control over gain at the zaptel level, and also
> at
> > the ADIT level apparently. Which should we use when trying to adjust
> things
> > with ztmonitor?
> >
> > Next question ... it's not clear to me what the target is with ztmonitor.
> > Are we shooting for tx and rx levels that are balanced. and about 50% of
> the
> > scale at their max energy? I found that when I called a voicemail service
> > and listened to the auto-attendant that the RX meter of ztmonitor was
> about
> > 50%, but I found the TX meter fluctuated wildly when I spoke. I could
> reduce
> > my voice level slightly and it would be very low energy, but with a slight
> > increase in volume it was pegged. The settings I made seemed to do nothing
> > to change this (I stop asterisk, unload and reload all zaptel modules, and
> > restart astertisk between each test).
> >
> > Is this really supposed to work? I've managed to get the connection
> > extremely distorted with some settings, but have yet to make a change that
> > improved the quality or removed the echo.
> >
> > Challenge b)
> > ========
> >
> > In some ways, this is the most worrying problem for us, since we would
> love
> > to be able to pass faxes through asterisk reliably, and from the traffic
> on
> > this list and the fine efforts of Mr. Underwood, it seems many others
> would
> > too. People have spoken here of the nearly immediate echo cancelation on
> > pure TDM circuits, and so I would have thought we'd escape the echo
> problems
> > above in either of the following setups:
> >
> > i) Plain ole' HP fax machine plugged into TDM400 FXS card
> > ii) Brooktrout/Eicon fax board connected to TE405P
> >
> > In practice we're rarely able to train with the remote fax machine when
> > dialing to an outside line through asterisk and the ADIT 600. If we do
> > manage to get connected and the connection supports ECM error correction,
> we
> > can see lots of retransmitted frames which again points to very poor line
> > quality. In contrast, by connecting directly to the analog lines we can
> send
> > faxes all day with the trusty little HP ... ie: the lines aren't
> inherently
> > bad.
> >
> > I guess what I'm looking for here is a sanity check ... are we trying to
> > push the limits here, or should this stuff be working much better than
> this?
> > We're willing to invest some time getting faxing to work, but I'd hate to
> > ask people here to dedicate their time and energy to something that's
> never
> > really going to work well due to limitations in Asterisk/zaptel
> technology.
> >
> > -Darren
> >
> > --
> > Darren Nickerson
> > Senior Sales & Support Engineer
> > iFax Solutions, Inc. www.ifax.com
> > darren.nickerson at ifax.com
> > +1.215.438.4638
> > +1.215.243.8335 (fax
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
> 