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Unread 04-14-2007, 01:49 AM   #1
handy9
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Question Caller voice distorted 'donald duck or Dalek'

I am new to Voxalot, got reasonable understanding of IP telephony but terrible time to get it to work. Hope this is the right place to post.

However I have recently registered to test with voxalot because a friend is with voxalot. Registration successful but calling *600 does not work get half a ringtone and then dead.

Calling my voxalot friend all appears fine, the call is set up and he sounds just great but my sound is only just understandable.

I am testing with SJphone 1.65 on win2000, from private LAN via Netcomm NB5 modem/router connected to Exetel

Added: Broadband link is 256/64 ie upload is 64 kb/s and my free sjphone does not have g729, however I was hoping it would try GSM to keep the bandwidt requirement down for uploading, hmmm?

Can anyone enlighten me.

Last edited by handy9; 04-14-2007 at 02:24 AM. Reason: Adding ADSL speed
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Unread 04-14-2007, 03:09 AM   #2
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Quote:
Originally Posted by handy9 View Post
Added: Broadband link is 256/64
This is more than likely due to your upstream speed (64K).

You should most definitely try changing your SJPhone Codecs and remove the g711 ones from the list.

*600 works from SJPhone so I am unsure what this problem could be.
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Unread 04-14-2007, 01:08 PM   #3
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Question

Thank you Martin, I have come to look more and more in that direction. When I did the sjphone audio wizard test, in desparation, as my guru had installed the phone and said all was well. Well hard to question him??

Anyway when I carefully did the test I came to the conclusion that the Audio I was producing was terrible, seems I was lucky to do some improvement by switching off the AGC.

Further I have come to the conclusion that if I could 'force' GSM codec I had to be able to do ok as test we have carried out from a 'dial-up' connection , the audio was impressive. Tomorrow this will again be tested.

At the moment I have hit another problem with the sjphone, it will not respond to *600 echo check, well seems that WireShark sees lots of data but I don't hear anything, no ringing, no audio, just seems to run. My thinking is that if the echo test does not work then other problems will follow.

Any ideas?
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Unread 04-14-2007, 11:03 PM   #4
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Quote:
Originally Posted by handy9 View Post
Thank you Martin, I have come to look more and more in that direction. When I did the sjphone audio wizard test, in desparation, as my guru had installed the phone and said all was well. Well hard to question him??

Anyway when I carefully did the test I came to the conclusion that the Audio I was producing was terrible, seems I was lucky to do some improvement by switching off the AGC.

Further I have come to the conclusion that if I could 'force' GSM codec I had to be able to do ok as test we have carried out from a 'dial-up' connection , the audio was impressive. Tomorrow this will again be tested.

At the moment I have hit another problem with the sjphone, it will not respond to *600 echo check, well seems that WireShark sees lots of data but I don't hear anything, no ringing, no audio, just seems to run. My thinking is that if the echo test does not work then other problems will follow.

Any ideas?
What happens if you dial just 600?
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Unread 04-15-2007, 01:21 AM   #5
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My thinking too, dialed 600 and I get 1 ring, then silence.

First: I did find out how to enable GSM only so now only codec SJ Labs 6.10 codec running.

1.. sjlog indicates connection made, I can make an example copy available?

2.. WireShark [Ethereal] trace overview looks fine too, I get plenty of RTP packets but, now and then I get a packet that says

Destination unreachable - Protocol unreachable

I have a copy of the packet, but this is a bit too complex for me to decipher confidently. There is a mention of a protocol 17, have not found what that is yet.

I have saved a txt file with copy of WireShark dump first 15 packets and corresponding sjlog, un request I will forward them but too big for thread I believe.

As far as I can see connection is made, audio streaming according to WS but no audio on my system. The only Audio is the single call ring when connecting.

----

I have disconnected SJphone, not uninstalled and installed X-Lite3 which does the echo test perfectly. However here is another problem, will leave that til later.
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Unread 04-16-2007, 01:42 AM   #6
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Oooops , hmmmm it seems that sjphone 1.65 is not the program to use with windows 2000. My Guru friend started up sjphone on his xp machine and noticed that I should not be running version 1.65. I have uninstalled and reverted back to a copy of version 1.60.289a and I am having success with echo test and registration.

Initially I had a really terrible voice/audio performance but I had luck finding the solution, I had installed Advanced help, great little addition to the phone, and via that I found that my audio driver buffer size which defaults to 20 ms is too short. I have tested 40, 50, 60 ms and all work fine I have also increased 'Driver output queue length' to 5, which is +1 on default.

-------

New Problem:

When trying to call another voxalot registered member, we are both in Australia, and I know he is on line and not engaged, I am consistently diverted instantly to his voicemail. The call connection seems perfect but sjlog does show that the 'owner' suddenly is us.voxalot.com??

Looking with WireShark I can confirm that at some stage "202.60.75.46 voxalot.com [Australia]" is replaced with"64.34.163.35 voxalot.com [usa]" for the rest of the call. [copy of file available on request].

I assume that we are not showing up as members on the us.voxalot server, a bit of guesswork?

Any suggestions
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Unread 04-16-2007, 02:35 AM   #7
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We now know, if we leave a message then an email will eventually arrive to say there is a voice message.

Also if my friend phones me the same thing happens, told that I am unavailable, directs to voicemail.

I am using sjphone, free version, have no dial plan.

My friend is using Netcomm ATA and paid dial plan with Voxalot

I think timeout is required for me for a while.
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Unread 04-16-2007, 05:06 AM   #8
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A bit more information.

Have now tried xlite-3 softphone installation as well, calling my friend passes the call to the voicemail box.

I replaced domain 'names' with IP addresses and that makes it clear that my telephone number is registered with voxalot.com and not the proxy au.voxalot.com.

I have used a standalone NetComm v85 IP phone which registers if I use IP numbers for domains [not sure why, as it only seems to be voxalot registration this is required] when registered I can call my friend as intended.

When watching the WireShark packet analyser I now notice that both our numbers show up nicely in the format xxxxxx@64.35.163.35. When making a call they just won't 'meet' at the same time? Something to do with timezones
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Unread 04-17-2007, 03:01 AM   #9
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Ok, problem solved when using sjphone 1.60 for calling voxalot members.

I have found that the codec matching is extremely important as follows.

My SJphone is free version, which comes with audio codecs
GSM ( 8 kb/s plus overhead)
iLBC 30msec
iLBC 20 msec
G.711 A-law
G.711 U-Law

I am using 256 kb/s download and 64 kb/s upload
I had therefor concluded the best would be to use GSM and had disabled all the other codecs ( also with a view to check with a friend on Dialup at about 33 kb/s, hoping this would just sqeeze in understandable voise)

However the real testing was to another friend who runs a successful VoIP implementation using a Netcomm NB9W router and I think he now has a 1500/512 connection, ie he has no real audio issues using VoIP.

The modem default codec is G.729 and we believe it will respond to G.711's and I think G.723 but it does not have GSM capability and here is the problem.

My friend's voxalot number is recognised by the voxalot servers but also that the codecs can never match so I am passed to the voice service.

I think I proved this by enabling my sjphone with G.711 A and the phone now rings out first before I am then passed to voice service.

When using the sjphone log, I was beginning to be concerned about the statement:

Channel audio input: started
Capability: undefined

I think there is a clue here that the codecs do not match.

Conclusion is that if codecs don't match odd things can happen and it is not necessarily clear what the problem is.

well, in hind sight if I had purchased a registration including the g.729 I would never have spent many hours wading through different settings. On the positive side I know a bit more. Now to figure out why the xlite-3 has similar problems.
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