Re: [Ekiga-list] Bad sound quality because of missing low pass filter

2010-03-02 Thread Dave Higton
On 2010-03-02, Christian Jaeger wrote:

 Hello
 
 I'm using Ekiga and Twinkle with the Diamondcard.us service as my main
 phone service. People are telling me every now and then that the sound
 quality is pretty poor, and in fact when I listen to my voice on the
 echo service it's always been quite much worse than when using a
 normal phone I think. It's regardless of the codec used, all that
 are offered by Diamondcard are using 8000 Hz sampling rate. And I'm
 using a headset and have checked sound quality locally, which is just
 fine, as I can verify for example with (the bfr command is to delay
 playback, so you can hear the echo better):
 
 arecord -t raw -f dat -r 22050 -c 1 | bfr -m 33075 | aplay -t raw -f
 dat -r 22050 -c 1 -
 
 Now 8000 Hz is of course lower, and in fact when using that rate:
 
 arecord -t raw -f dat -r 8000 -c 1 | bfr -m 12000 | aplay -t raw -f
 dat -r 8000 -c 1 -
 
 it sounds exactly as in the echo test. Now one thing I know is that if
 you sample a signal, you need a low pass before the quantization, at
 half the sample rate or lower, to remove frequencies that are too high
 to be represented in the digitalized data. If the low pass is missing,
 the high frequencies are turned to noise instead. So what happens when
 a program samples data at a low frequency, will it somehow make the
 sound card tune a low pass to the correct cut-off frequency when the
 sample rate is changed? Sounds rather unlikely, and in fact the echo
 test at 8000 Hz sounds exactly like suffering from that problem: you
 can't distinguish s from f anymore, which may just be unavoidable
 with no frequency spectrum above 4000 Hz, but worse, they sound very
 loud and noisy, and for example an combined s and vowel, like they
 often appear in french (e.g. bourgeoisie I think, I'm not a native
 french speaker), sound really awful. Now there's a way to remedie
 this: low-pass filtering can also be done computationally, so we can
 digitize at say DAT quality (48000 Hz) and resample down to 8000
 samples/second programmatically, which is what the following command
 line is doing:
 
 arecord -t raw -f dat -c 1 | bfr -m 36000 | sox -t raw -r 48000 -s -2
 -c 1 -  -t raw -r 8000 - resample -ql |bfr -m 6000 | aplay -t raw -f
 dat -r 8000 -c 1 -
 
 It sounds more hollow/muffled now, but much cleaner. No loud noises
 anymore for sharp consonants.
 
 Would doing this in VoIP programs improve perceived sound quality for
 the listener? I would expect so, but haven't been able to test. (All I
 know is that more often than not I'm using Skype to call people now
 (Skype-out service), because people are telling me they hear me
 better.)
 
 Note that you can very well say you're wrong, we don't need a
 low-pass, just use a higher sampling rate then it sounds just fine,
 which is of course partially true, with higher sampling rates the
 relevance of a low-pass decreases quickly as the spectrum of
 frequencies that are higher than half the sample rate becomes narrower
 (and an increasing part of the consonant frequency range is digitized
 correctly instead of as noise). But given that at least Diamondcard
 only offers 8000 Hz, and there may be valid reasons for this (like
 maybe the phone system is still using that frequency), what's your
 position on this?

You've discovered aliasing.  It sounds pretty bad at 8 kHz sampling
rate, doesn't it?

I suspect you have a poorly engineered sound card.  It /should/, as
you suggest, have proper low pass filtering, but apparently it
doesn't.  You can often get away with gentle anti-alias filtering
for a 48 kHz sampling rate, but not 8 kHz.

Ideally, the sound card would sample everything at 48 kHz and then
decimate by a factor of 6 using proper low-pass digital filtering.
It sounds as if yours simply sub-samples everything instead, which
aliases almost everything between 4 kHz and 48 kHz and folds it
into the up-to-4 kHz band.

Maybe a better sound card would do what you want.  (I'm sorry, I
don't have any recommendation.)

Dave


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Re: [Ekiga-list] Bad sound quality because of missing low passfilter

2010-03-02 Thread Dave Higton
 -Original Message-
 From: ekiga-list-boun...@gnome.org 
 [mailto:ekiga-list-boun...@gnome.org] On Behalf Of Christian Jaeger
 Sent: 2010 March 02 14:19
 To: Ekiga mailing list
 Subject: Re: [Ekiga-list] Bad sound quality because of 
 missing low passfilter
 
 Well, I wouldn't say my hardware is cheap; it's a ThinkPad T61 laptop,
 worth around 1800 USD when bought new about 2.5 years ago.

I didn't say cheap; I said poorly engineered :-)

I was a designer of professional audio equipment, later of audio
test equipment, for many years.  More recently I commissioned two
ASICs to do decimation and interpolation.  I've always thought of
audio as being easy; but I keep being surprised by the number of
engineers who are completely clueless about audio.  Some of them
design audio equipment. :-(

Dave


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[Ekiga-list] What would cause this?

2010-01-04 Thread Dave Higton
I'm trying to make Ekiga (3.0.2 on Fedora 10) work with an Asterisk box.
Note: this is an ENTIRELY INTERNAL switch; there is NO connection to the
outside world; there is no Ekiga.net or any other trunk account.

I'm running Wireshark on the Ekiga/F10 box, and I see this first line in
the registration SIP packets:

REGISTER sip:sip SIP/2.0

That can't be right.

The account I have created has these settings:

Name: 221
Registrar: sip:192.168.46.178
User: 221
Authentication User: 221
Password: secret
Timeout: 3600

The Asterisk box is 192.168.46.178; the Ekiga/F10 box is 192.168.44.221.

What do I have to change in order to get it to register with a sensible
name or IP address?

Dave


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Re: [Ekiga-list] What would cause this?

2010-01-04 Thread Dave Higton
Then no registration packets are sent at all.
 
Dave




From: ekiga-list-boun...@gnome.org 
[mailto:ekiga-list-boun...@gnome.org] On Behalf Of Damien Sandras
Sent: 2010 January 04 16:52
To: Ekiga mailing list
Subject: Re: [Ekiga-list] What would cause this?


Le lundi 04 janvier 2010 à 16:45 +, Dave Higton a écrit : 

I'm trying to make Ekiga (3.0.2 on Fedora 10) work with an 
Asterisk box.
Note: this is an ENTIRELY INTERNAL switch; there is NO 
connection to the
outside world; there is no Ekiga.net or any other trunk account.

I'm running Wireshark on the Ekiga/F10 box, and I see this 
first line in
the registration SIP packets:

REGISTER sip:sip SIP/2.0

That can't be right.

The account I have created has these settings:

Name: 221
Registrar: sip:192.168.46.178

Use 192.168.46.178, not sip:192.168.46.178 as registrar.



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Re: [Ekiga-list] What would cause this?

2010-01-04 Thread Dave Higton
I don't think so - but where would an outbound proxy be specified in Ekiga?
 
Dave




From: ekiga-list-boun...@gnome.org 
[mailto:ekiga-list-boun...@gnome.org] On Behalf Of Damien Sandras
Sent: 2010 January 04 17:01
To: Ekiga mailing list
Subject: Re: [Ekiga-list] What would cause this?


Le lundi 04 janvier 2010 à 16:55 +, Dave Higton a écrit : 

Then no registration packets are sent at all. 


Probably you specified an outbound proxy ?
No idea, you should post a -d 4 output somewhere. 


-- 
 _ Damien Sandras
(o-  
//\Ekiga Softphone : http://www.ekiga.org/
v_/_   Be IP   : http://www.beip.be/
   FOSDEM  : http://www.fosdem.org/
   SIP Phone   : sip:dsand...@ekiga.net
   



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Re: [Ekiga-list] What would cause this?

2010-01-04 Thread Dave Higton
OK... let me see, there isn't a newer packaged version for Fedora 10, is there? 
 So it's a build-it-myself job?
 
I suspect that, at least with the version I'm currently running, it wasn't 
designed (or tested) to run without an ekiga.net account.  There is no way I 
can connect my system to the outside world, so an Ekiga.net account is out of 
the question for me.
 
Dave




From: ekiga-list-boun...@gnome.org 
[mailto:ekiga-list-boun...@gnome.org] On Behalf Of Damien Sandras
Sent: 2010 January 04 18:00
To: Ekiga mailing list
Subject: Re: [Ekiga-list] What would cause this?


Le lundi 04 janvier 2010 à 17:24 +, Dave Higton a écrit : 

Only 4 KB when zipped, so here it is. 


I don't see anything wrong in the file.

Perhaps try upgrading to a newer Ekiga version ? I have never heard of 
such a problem before.



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Re: [Ekiga-list] ekiga.net down again

2009-04-30 Thread Dave Higton
On 2009 April 29, Damien Sandras wrote:
 
 Le mercredi 29 avril 2009 à 23:25 -0700, Allen T Parker a écrit :
  Yes.  I've been unable to connect for nearly 24 hours.
  
 
 I restarted it. However, I'll be offline for several days. If 
 it crashes
 again, nobody will be able to restart it.

But /why/ does it fall over so frequently?

Dave


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Re: [Ekiga-list] direct IP to IP SIP call without SIP registrar, not working in 3.x

2009-02-26 Thread Dave Higton
Patrick Lee wrote:

 Not sure any one has asked this question before.
 
 I was able to make call without registering any SIP registrar in Ekiga
 ver 2.x.  No need to create SIP account.
 Just type sip:192.168.1.34  (the remote IP address) and the 
 call was setup.
 And the remote client could be x-lite.
 
 But NOT with ver 3.x, the same syntax gave me no response.  It seems
 like I have to create a SIP account first.
 
 My environment is a complete isolated network so I simply can't
 register in ekiga.net for SIP account.
 And there is some other SIP clients so I have to stick with 
 SIP protocol.
 
 I know SIP registration is a proper SIP operation.  But I love ekiga
 as it allows pt-to-pt call without setting up a SIP server.  But not
 true for ver 3.x.  Any solution ?

It may be worth mentioning that I have the same situation (no SIP
access to the outside world, therefore Ekiga cannot possibly
register to anything), the same requirement to do VoIP across our
own LAN (only), and the same problem.

Dave


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Re: [Ekiga-list] What is 'ekiga's internal roster'?

2009-01-07 Thread Dave Higton
 2009/1/7 Peter Saint-Andre stpe...@stpeter.im:
  Julien Puydt wrote:
  Peter Saint-Andre a écrit :
  FWIW, we use the term roster in XMPP to refer to a contact list.
 
  That's for that very reason it's named like that in ekiga ;-)
 Hmm, but Ekiga is not written just for XMPP developers is it? I think
 for consistency the working should be reconsidered.

I have to add my agreement.  XMPP and Ekiga have added a new definition
for the word roster that didn't previously exist.  One would have to
have used XMPP and/or Ekiga, and done some research, in order to understand
what they mean by roster.

It's an inappropriate choice of word.

Dave


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Re: [Ekiga-list] Cannot make SIP calls with Ekiga 3.0.1

2009-01-07 Thread Dave Higton
 Le mercredi 07 janvier 2009 à 14:11 +0100, Eugen Dedu a écrit :
  Clayton wrote:
   Try to call:
   sip:yournum...@sip.voiparound.com
   instead of sip:yournumber
  
   There is this error in your log:
  
   SIP Connecting to sip:[...] via [...]
   Socket  Illegal RFC952 characters in DNS name [...]
   OpalCould not find host [...]
   
   OK, tried that (I thought I had tried that several times in the
   troubleshooting I had done before sending a mail to this 
 list), and I
   can make a call now... the call quality is shockingly 
 bad, but I do
   get through using the full 
 sip:yournum...@sip.voiparound.com syntax.
   
   Is this a new requirement?  In 2.0.12 I could simply type 
 in the phone
  
  Yes.
  
  I propose (maybe I will fix too):
  - either print Malformed number or something like this
  - or desactivate the Call button until the number has a @ inside
 
 In this particular case, there is only 1 account, thus ekiga could
 automatically add the full address by itself...

That's more helpful - and, I suspect, a common case.

Dave


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Re: [Ekiga-list] Infos

2008-09-03 Thread Dave Higton
I feel I have to add a me too: direct IP calls absolutely do make
sense.  I could call someone else on my LAN (which could be quite
large), or on my VPN (which could of course be even larger).  I don't
want to have to install an IPBX just to avoid direct IP calls.

SIP itself supports direct calls, and I believe that all SIP UAs should
too.

Dave

 emanuele bottegoni wrote:
 
 Hi,
 
 Why direct SIP calls have don't make sense?
 For example I'm involving in Home automation and in this kind of
 application not always is possible to put a PBX server even 
 if the place
 is ethernet cabled.
 So,as intercoms SIP based services,We can consider the 
 opportunity tou SIP
 direct calls.
 I've noticed that SJphone can support this feature.I'd like 
 to know how
 with Ekiga is possible to realize this kind of aaudiuo video sessions.
 
 Thanks
 Emanuele
 
 
  Le mercredi 03 septembre 2008 à 15:18 +0200, emanuele bottegoni a
  écrit :
   Thanks,
   
   I'll try to use the beta 3 concerning the use of H264.
   What about the SIP direct Call in which version is available?
   Furthermore,which is the correct format of SIP address I have to
  insert?
   sip:@(IP of the host):5060 ?
   
  
  Direct ip calls do not make sense.
  -- 
  Damien Sandras



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Re: [Ekiga-list] For or against ?

2008-08-07 Thread Dave Higton
The link is wrapped.  Add the +Dialup and it works.

Dave

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of Paul Magarey
 Sent: 2008 August 07 11:02
 To: Ekiga mailing list
 Subject: Re: [Ekiga-list] For or against ?
 
 Link doesn't work.
 
 2008/8/7 Damien Sandras [EMAIL PROTECTED]:
  Hello,
 
 
  Would you be for or against such a scheme ?
 
  http://www.voip-info.org/wiki/view/Fee+Announcement+from+Free+World
  +Dialup
 
  --
   _ Damien Sandras
  (o-
  //\Ekiga Softphone : http://www.ekiga.org/
  v_/_   NOVACOM : http://www.novacom.be/
FOSDEM  : http://www.fosdem.org/
SIP Phone   : sip:[EMAIL PROTECTED]
 
 
  ___
  ekiga-list mailing list
  ekiga-list@gnome.org
  http://mail.gnome.org/mailman/listinfo/ekiga-list
 
 ___
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 ekiga-list@gnome.org
 http://mail.gnome.org/mailman/listinfo/ekiga-list
 



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[Ekiga-list] Ekiga for Eee?

2007-12-14 Thread Dave Higton
Is it possible to create a special distribution of Ekiga
for the Asus Eee?

For those who're not (yet) familiar with it, the Eee has
only 4 GB of flash memory for all its applications and
data; no hard drive.  It does come equipped with sound
in and out, and a webcam.  One of the supplied apps is a
Skype client.  I thought it would be nice to have proper
VoIP.

The critical qualities are: (1) minimal use of file space,
(2) ease of installation.

If not a special distribution or package, the other
alternative is to take a standard distribution of Ekiga
and its required libraries, then prune out all files that
are not necessary to running Ekiga.  In which case I would
appreciate advice on what can go.

Dave


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Re: [Ekiga-list] trace of problem getting Ekiga to workin Kubuntu gutsy

2007-12-04 Thread Dave Higton
 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of Julien Puydt
 Sent: 2007 December 04 10:16
 To: Ekiga mailing list
 Subject: Re: [Ekiga-list] trace of problem getting Ekiga to 
 workin Kubuntu gutsy
 
 Sergei Steshenko a écrit :
  Looks like a bug (midding feature) in Ekiga. OpenWengo has 
 a configuration file
  which has port numbers in it; the file can be changed from the GUI.
 
 In gconf, there is : /schemas/apps/ekiga/protocols/sip/listen_port
 
 Nothing's missing.

I can't look at my home installation atm since I'm at work, but I also
have Ekiga installed on an FC5 box next to me.  This has directories
/root/.gconf/apps/ekiga/protocols and
/home/dhigton/.gconf/apps/ekiga/protocols, neither of which contains
a sip subdirectory or a listen_port file.  Both installations are 2.0.11.

Is sip/listen_port created only under earlier/later versions of Ekiga,
or only under some different circumstance, or what?

Dave


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Re: [Ekiga-list] trace of problem getting Ekiga to workin Kubuntu gutsy

2007-12-03 Thread Dave Higton
 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of jim ruxton
 Sent: 2007 December 04 00:37
 To: Ekiga mailing list
 Subject: Re: [Ekiga-list] trace of problem getting Ekiga to 
 workin Kubuntu gutsy
 
 
Hi I still can't get Ekiga to connect to the echo test 
 (or anything
else) . Below is a trace of starting Ekiga and 
 attempting a call to
sip:[EMAIL PROTECTED] . Can anyone see why this doesn't work?
http://www.pastebin.ca/795453

   
   You are not registered to ekiga.net, so the echo test rejects you.
   
   Moreover, you are behind NAT and STUN is not enabled.
  Thanks Damien, strange I thought I was registered, anyway I
  re-registered as [EMAIL PROTECTED] and enabled STUN and I 
 still get the
  same result of secutity check failed . Here is the most 
 recent trace:
  http://www.pastebin.ca/796302
 
 I finally figured out what was going on. Another program was 
 using port
 5060 so I changed the port ekiga uses to 5061 using gconf-editor.
 Everything is working well now. Thanks everyone for helping out.

What file did you have to edit to do this?  I have (sort of) the
opposite problem: my Ekiga installation (SuSE 10.2) uses port 5061
and I'd prefer it to use 5060.  I searched through the installation
at the weekend and couldn't see any reference to port numbers
anywhere.

Dave


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Re: [Ekiga-list] No echo with sip [EMAIL PROTECTED]

2007-11-20 Thread Dave Higton
 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of Damien Sandras
 Sent: 2007 November 19 08:53
 To: Ekiga mailing list
 Subject: Re: [Ekiga-list] No echo with sip [EMAIL PROTECTED]
 
 
 Le dimanche 18 novembre 2007 à 16:30 -0200, Emmanuel Favre-Nicolin a
 écrit :
  Le dimanche 18 novembre 2007, yannick a écrit :
   Le dimanche 18 novembre 2007 à 11:04 +0100, Damien 
 Sandras a écrit :

 http://emmanuelfavrenicolin.free.fr/Public/Divers/20071117_eki
 gaoutput.tx
   t
   
Unfortunately this is not a valid -d 4 output. It looks 
 like a -d 1
output.
  
   Are you using Ubuntu Gutsy or Debian (sid I guess... not 
 sure)? There is
   a bug there which prevent to get a valid -d 4 output. 
 Here is the bug
   report:
   https://bugs.launchpad.net/ubuntu/+source/opal/+bug/155302
  
  In fact, I'm using gentoo. I didn't found any similar bug 
 in the gentoo bug 
  system.
  I reinstalled ekiga with debug flag, but guess it didn't 
 change too much the 
  ekiga -d 4 output:
  
 http://emmanuelfavrenicolin.free.fr/Public/Divers/20071118_eki
 gaoutput.txt
  
  Oh, better I reinstalled pwlib, opal and ekiga with debug 
 flag and here is the 
  ekiga -d 4 output for an entire session with one attempte call to 
  [EMAIL PROTECTED] (I manually stopped the call) :
  
 http://emmanuelfavrenicolin.free.fr/Public/Divers/20071118b_ek
 igaoutput.txt
  
 
 You seem to be sending and receiving audio, so I do not understand why
 you hear nothing... 
 
 While being in a call, make sure the mic volume and such are correct.
 Disable sound events in the preferences.
 
 And if you connect to someone else, does he hear you ?

If you're happy using Wireshark, you can catch the entire session and
see whether you're transmitting audio.  Configure Ekiga so that the mu
law codec is at the top (or the only one ticked).  Then you'd normally
expect all your audio transmissions to have 172 bytes of payload, of
which the last 160 are the audio; there should be one of these packets
from you every 20 milliseconds.  What you DON'T want to see in the audio
is 160 identical bytes, as this means silence (usually 7F or FF).  A
random-looking mix of values means you're sending non-silence.  You want
to see a fairly wide range of values.  You should see the same sort of
thing coming back too.

Dave


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Re: [Ekiga-list] Video Problem with Ekiga on Windows XP Pro

2007-11-20 Thread Dave Higton
In fact I've received video on the Windows 2.0.11 beta too.  The only
thing I haven't had working is simultaneous transmission and reception
of video, since I only have one webcam handy.
 
Windows XP Pro.  This is on the company's LAN where there is no SIP
server of any kind and no SIP access to external providers.
 
Dave




From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave Higton
Sent: 2007 November 19 08:06
To: Ekiga mailing list
Subject: Re: [Ekiga-list] Video Problem with Ekiga on Windows XP
Pro


Yes; I can see my video; the far end can see my video.  I've not
had a call with anyone who has a camera at their end so I don't yet know
whether incoming video works.  This is the Windows 2.0.11 beta.
 
Dave




From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: 2007 November 16 20:58
To: ekiga-list@gnome.org
Subject: [Ekiga-list] Video Problem with Ekiga on
Windows XP Pro


Anyone have video working  on the Windows Beta of EKIGA
?

-Original Message-
From: Komara, Kevin 
Sent: Thursday, November 15, 2007 8:28 PM
To: 'ekiga-list@gnome.org'
Subject: Video Problem with Ekiga on Windows XP
Pro 


Hi,
I am a new Ekiga user.  I have to compliment
your efforts - Ekiga is a very nice piece of software !!
 
I have successfully installed Ekiga on 2 PCs
(PC1 and PC2) on my home network running Windows XP Pro on both PCs.
The software installed beautifully on both PCs.  I was able to configure
SIP through my Asterisk Win32 Server (On a third Windows XP PC - PC3).
I can make full duplex voice calls through Asterisk WIn32 using Ekiga
between PC1 and PC2 with no issues.  What I really want to do is video
conference between PC1 and PC2 through Asterisk.  
 
I have not set up a SIP account on your web site
- I just want to do SIP calls on my private network.
 
PC1 has an older analog video capture card -
Pinnacle DC10+ installed.  I have an old VHS video camera plugged into
it.  It works very well under Windows XP and is recognized by Ekiga.  I
can see the video from the DC10+ on the local screen under Ekiga.
 
PC2 has a newer USB video camera installed.  It
also works very well under WindowsXP - although it is slower and with
much poorer resolution than the DC10+.  The USB camera is also
recognized by Ekiga on PC2 and I can see its video on the local Ekiga
window.
 
When I make a call from PC1 to PC2 I expected
the video from PC1 to be visible in a window on PC2, and of course the
video from PC1 on a window on PC2.  This does not happen.  I only see
the local video on both PC1 and PC2.  Under the view on both Ekiga
installations I see the options for Remote Video,
Both(Picture-in-Picture), Both (Side-by-Side), and Both (Both in
New Windows) - but NONE of these options can be selected - they are
grayed out.  Should this be working ???  Do I have something configured
incorrectly ?  Do I have to purchase an upgrade ?
 
I have successfully configured another piece of
Video conference software (Zoomcall Pro) and the video from these same
PCs works fine in video conference mode.  Unfortunately Zoomcall
REQUIRES you to communicate through their server on the Internet.  I
don't want to do that - I want to stay on my private network.
 
Please help any way you can.
 
Also - as you might have guessed - I have quite
an elaborate set up in my lab area so I would be willing to test any new
software you might need a third party to test.
 
Thanks,
Kevin Komara
 


*
Kevin J. Komara P.E.
Senior Engineer
PJM Interconnection L.L.C.
 
Phone:   (610) 666-4751
Fax:   (610) 666-4282
Email:[EMAIL PROTECTED

Re: [Ekiga-list] No echo with sip [EMAIL PROTECTED]

2007-11-20 Thread Dave Higton
 [mailto:[EMAIL PROTECTED] On Behalf Of Emmanuel 
 Favre-Nicolin
 Sent: 2007 November 20 10:13
 To: Ekiga mailing list
 Subject: Re: [Ekiga-list] No echo with sip [EMAIL PROTECTED]
 
 Le mardi 20 novembre 2007, Dave Higton a écrit :
   -Original Message-
   From: [EMAIL PROTECTED]
   [mailto:[EMAIL PROTECTED] On Behalf Of Damien Sandras
   Sent: 2007 November 19 08:53
   To: Ekiga mailing list
   Subject: Re: [Ekiga-list] No echo with sip [EMAIL PROTECTED]
  
  If you're happy using Wireshark, you can catch the entire 
 session and
  see whether you're transmitting audio.  Configure Ekiga so 
 that the mu
  law codec is at the top (or the only one ticked).  Then 
 you'd normally
  expect all your audio transmissions to have 172 bytes of payload, of
  which the last 160 are the audio; there should be one of 
 these packets
  from you every 20 milliseconds.  What you DON'T want to see 
 in the audio
  is 160 identical bytes, as this means silence (usually 7F or FF).  A
  random-looking mix of values means you're sending 
 non-silence.  You want
  to see a fairly wide range of values.  You should see the 
 same sort of
  thing coming back too.
 
  Dave
 
 mulaw is named PCMU in ekiga?

Yes.

 Dave, I did that, but 
 http://emmanuelfavrenicolin.free.fr/Public/Divers/Wireshark/20
 071120_wireshack.libcap
 
 first transmitted RTP packet

[snip] This is quiet but not silent.

 5th

[snip] Similar.

 last transmitted

[snip] This looks like convincing audio, definitely
not silent, not even quiet.

 last received

[snip] This is quiet but not silent.

 Is that make sense

Yes.

  There is another problem. The flux (payload) on eth0 is at 
 the beginning not 
 zero, but then it rapidly becomes zero byte/second so that I 
 expect that 
 absolutly nothing else is transmitted.
 
 I observed 2 cases. 
 1) the transmitted bytes (meekiga.net) becomes zero before 
 the end of the 
 woman blablabla and then the received bytes/ (ekigame) 
 becomes zero too
 2) Both  transmitted and received bytes fluxes becomes zero 
 when the woman 
 ends up her blablabla

You mean that there are no more packets being received and/or
transmitted?  That would be wrong, but it would explain why
you can't hear the echo.

You should be transmitting continuously, 50 packets per
second, for the entire duration of the call.

I can't be so sure about what you should receive.  If the
echo is of (say) 4 seconds of speech with a 4 second gap
between them (for you to record in), it's possible that
the packets will only flow from ekiga.net to you during
the times that you should be hearing something.

Dave


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Re: [Ekiga-list] Ekiga paralysed if no registration server

2007-11-06 Thread Dave Higton
 Le mardi 06 novembre 2007 à 10:42 +, Dave Higton a écrit :
  Why is Ekiga paralysed if no registration server is specified or
  enabled?  I get this on SuSE Linux 10.2 with Ekiga version 
 2.0.11 but it
  was no different on the previous version (2.0.3 I think).
  
 
 What does that mean ?

It takes several tens of seconds (at least) to respond, for
example to an incoming call.

Dave


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Re: [Ekiga-list] Ekiga paralysed if no registration server

2007-11-06 Thread Dave Higton
 Le mardi 06 novembre 2007 à 11:45 +, Dave Higton a écrit :
   Le mardi 06 novembre 2007 à 10:42 +, Dave Higton a écrit :
Why is Ekiga paralysed if no registration server is specified or
enabled?  I get this on SuSE Linux 10.2 with Ekiga version 
   2.0.11 but it
was no different on the previous version (2.0.3 I think).

   
   What does that mean ?
  
  It takes several tens of seconds (at least) to respond, for
  example to an incoming call.
  
 
 There is a limitation in Ekiga 2.0: the fact that it does DNS queries
 even when it is not useful. On WIN32, the code has been accidentally
 removed. It would explain why you do not encounter the 
 problem on WIN32.
 
 It indicates a problem on your DNS server... (it is slow).

In that configuration, it's non-existent.

Can I disable Ekiga's DNS requests?

Dave


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Re: [Ekiga-list] RTP/RTCP port ranges

2007-10-29 Thread Dave Higton
 Dave, can you send to me the ports windows use? Thanks in advance.

This is the relevant excerpt from my ekiga.conf file:

applyto/apps/ekiga/protocols/ports/udp_port_range/applyto
typestring/type
default5060:5100/default
/schema
schema
applyto/apps/ekiga/protocols/ports/rtp_port_range/applyto
typestring/type
default5000:5059/default
/schema
schema
applyto/apps/ekiga/protocols/ports/tcp_port_range/applyto
typestring/type
default3:30010/default
/schema
schema
applyto/apps/ekiga/protocols/h323/listen_port/applyto
typeint/type
default1720/default
/schema

Dave
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Re: [Ekiga-list] Bad sound from mic in Windows Ekiga

2007-10-24 Thread Dave Higton
 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of Damien Sandras
 Sent: 2007 October 23 17:07
 To: Ekiga mailing list
 Subject: Re: [Ekiga-list] Bad sound from mic in Windows Ekiga
 
 Hello,
 
 Le mardi 23 octobre 2007 à 16:12 +0100, Dave Higton a écrit :
  I've just deleted my Ekiga installation (which came from
  ekiga-setup-2.0.9-BETA.exe, and on which I never got more 
 than a buzz
  from the microphone) and replaced it from 
 ekiga-setup-2.0.11-BETA.exe.
  This is an improvement, i.e. it's not quite as disastrous; 
 I can hear my
  voice, though with a bad buzz added, and probably even 
 modulated by the
  buzz.  I'm referring to what I hear when I run the audio test in the
  installation process.
  
  I can hear other sounds in the headphones OK, so I know 
 that audio out
  is working OK.
  
  I have the impression that there's a mismatch in the 
 sampling rates.  I
  opted to install with a sample rate of 8000, which wasn't 
 offered in the
  list, it had to be a custom sample rate.  I need 8000 as I 
 need to be
  compatible with common telecom sample rates.
  
  I think my next step will have to be to feed in a signal 
 from a signal
  generator and look at the output with an oscilloscope while 
 running the
  test.
  
  OS is Windows XP with a full set of updates.  The sound system is
  Realtek AC97 Audio for microphone and playback.
  
  Can anyone suggest what might be done to fix the problem?  
 Have others
  experienced the same?
 
 The sound audio test runs at 16 kHz. But when making calls, 
 it will use
 8 kHz or 16 kHz depending on the codec being used.
 
 I can not really help for your audio problem though, it seems 
 to be some
 kind of hardware driver problem.

I don't think so.  I just used Cool Edit 96 to record a few
seconds of my voice and play it back.  This uses the same
drivers as does Ekiga - but Cool Edit records and replays
fine.  I set Cool Edit to record in 8 bit mono at 8000 Hz.

It has to be something about the way Ekiga uses the drivers,
doesn't it?

Dave


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[Ekiga-list] Ekiga rpm for Fedora Core 5?

2007-10-24 Thread Dave Higton
Does an rpm exist for any recent version of Ekiga on Fedora Core 5?  If
so, where can I find it?

I can't upgrade from FC5 because the product I'm working on uses FC5 and
therefore all our dev tools and boxes also have to remain at FC5.  We do
plan to upgrade, but we haven't got firm dates to do so yet.

Dave


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[Ekiga-list] RTP/RTCP port ranges

2007-10-24 Thread Dave Higton
Is it possible to configure the range of ports that Ekiga uses for RTP
and RTCP?

Dave


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Re: [Ekiga-list] RTP/RTCP port ranges

2007-10-24 Thread Dave Higton
 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of Damien Sandras
 Sent: 2007 October 24 15:21
 To: Ekiga mailing list
 Subject: Re: [Ekiga-list] RTP/RTCP port ranges
 
 
 Le mercredi 24 octobre 2007 à 10:48 +0100, Dave Higton a écrit :
  Is it possible to configure the range of ports that Ekiga 
 uses for RTP
  and RTCP?
  
 http://wiki.ekiga.org/index.php/Internet_ports_used_by_Ekiga

Ah.  I take it this facility isn't available in the Windows
version, then?

Dave


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Re: [Ekiga-list] RTP/RTCP port ranges

2007-10-24 Thread Dave Higton
 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of Damien Sandras
 Sent: 2007 October 24 15:44
 To: Ekiga mailing list
 Subject: Re: [Ekiga-list] RTP/RTCP port ranges
 
 
 Le mercredi 24 octobre 2007 à 15:29 +0100, Dave Higton a écrit :
   -Original Message-
   From: [EMAIL PROTECTED] 
   [mailto:[EMAIL PROTECTED] On Behalf Of Damien Sandras
   Sent: 2007 October 24 15:21
   To: Ekiga mailing list
   Subject: Re: [Ekiga-list] RTP/RTCP port ranges
   
   
   Le mercredi 24 octobre 2007 à 10:48 +0100, Dave Higton a écrit :
Is it possible to configure the range of ports that Ekiga 
   uses for RTP
and RTCP?

   http://wiki.ekiga.org/index.php/Internet_ports_used_by_Ekiga
  
  Ah.  I take it this facility isn't available in the Windows
  version, then?
  
 
 You can edit the configuration file by hand. You will have to discover
 by yourself where it is being put, because I ignore it.

OK, got it.  Thanks.

Dave


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[Ekiga-list] Bad sound from mic in Windows Ekiga

2007-10-23 Thread Dave Higton
I've just deleted my Ekiga installation (which came from
ekiga-setup-2.0.9-BETA.exe, and on which I never got more than a buzz
from the microphone) and replaced it from ekiga-setup-2.0.11-BETA.exe.
This is an improvement, i.e. it's not quite as disastrous; I can hear my
voice, though with a bad buzz added, and probably even modulated by the
buzz.  I'm referring to what I hear when I run the audio test in the
installation process.

I can hear other sounds in the headphones OK, so I know that audio out
is working OK.

I have the impression that there's a mismatch in the sampling rates.  I
opted to install with a sample rate of 8000, which wasn't offered in the
list, it had to be a custom sample rate.  I need 8000 as I need to be
compatible with common telecom sample rates.

I think my next step will have to be to feed in a signal from a signal
generator and look at the output with an oscilloscope while running the
test.

OS is Windows XP with a full set of updates.  The sound system is
Realtek AC97 Audio for microphone and playback.

Can anyone suggest what might be done to fix the problem?  Have others
experienced the same?

Dave


*

NICE CTI Systems UK Limited (NICE) is registered in England under company 
number, 3403044.  The registered office of NICE is at Tollbar Way, Hedge End, 
Southampton, Hampshire SO30 2ZP.

Confidentiality: This communication and any attachments are intended for the 
above-named persons only and may be confidential and/or legally privileged. Any 
opinions expressed in this communication are not necessarily those of NICE. If 
this communication has come to you in error you must take no action based on 
it, nor must you copy or show it to anyone; please delete/destroy and inform 
the sender by e-mail immediately.

Monitoring: NICE may monitor incoming and outgoing e-mails.

Viruses:  Although we have taken steps toward ensuring that this e-mail and 
attachments are free from any virus, we advise that in keeping with good 
computing practice the recipient should ensure they are actually virus free.



 

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