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Unread 02-25-2009, 03:42 PM   #1
casch
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Angry me too: sometimes only oneway Audio (VoipCheap-Voxalot-ATA)

Same problem in little different way.

My ATA is conected to eu.voxalot.com,
and Voxalot to VoipCheap.

(all times is GMT-1, Vienna-time, Austria)

Just (15:59:54) made a call (40xx36-> +431514xx66xx) and she couldn't hear me,

then I made check calling myself (16:00:22 +431xxxx362) ok, could hear and speak to my own answ. machine.

Then redialed her and this time it was ok: 16:01:19 +431514xx66xx we spoke 9 min.

Sometimes it is perfect, sometimes one does not hear me, sometimes I can't hear anything!!

Please - PLEASE - check and get rid of that which way works lottery, it is really annoying - to me it seems as if every other call is only one way!!

Hi, here is -- do you hear me? DO YOU HEAR ME? I CAN HEAR YOU PERFECTLY, DON'T SHOUT AT ME ..and so on.

Regards,
Carl
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Unread 02-25-2009, 04:23 PM   #2
kendid
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Default Same here

Yeah, I got the same problem... Sometimes one way audio, sometimes not. Did not seem to be a problem in the past.

I'm also connected to eu server

I've also tried this with echo test -- sometimes works, sometimes doesnt...
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Unread 02-25-2009, 08:38 PM   #3
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Default

casch and fuzzuy,

As everything is working reliably for most of us, the problem may be on your end. If you are using a Linksys phone/ATA, please answer the following questions.

Is your phone/ATA behind a NAT router?

What settings are you using for the following?

Handle VIA received:
Handle VIA rport:
Insert VIA received:
Insert VIA rport:
Substitute VIA Addr:
Send Resp To Src Port:
STUN Enable:
STUN Test Enable:
STUN Server:
NAT Keep Alive Intvl:

NAT Mapping Enable:
NAT Keep Alive Enable:
NAT Keep Alive Msg:
NAT Keep Alive Dest:

Proxy:
Outbound Proxy:
Use Outbound Proxy:
Use OB Proxy In Dialog:
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Unread 02-25-2009, 09:01 PM   #4
chatalot
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Default

I have also just started to notice this problem with voipcheap.com. Need to try a second time to make a call - first time the person I am calling can't hear me.

I am suspecting that after making no calls for a while that my first calls audio is failing. (perhaps the NAT setup is getting lost after a while due to my settings)

My setup in case anybody with same problems notices anything similar

Linksys WRT54GL router using NAT with ports 5060-5061 forwarded to Linksys PAP2 ATA with Static IP
Voipcheap.com via voxalot (not registered)

Symetric NAT = on - optimise audio = yes - just turned these off as recommended in the post above will let you know if it helps.

As requested above my setting are as follows but remember that my setup has been ok for over 12 months

Handle VIA received: NO
Handle VIA rport: NO
Insert VIA received: NO
Insert VIA rport: NO
Substitute VIA Addr: YES
Send Resp To Src Port: YES
STUN Enable: YES
STUN Test Enable: NO
STUN Server: stun.voipcheap.com:3478
NAT Keep Alive Intvl: 28

NAT Mapping Enable: YES
NAT Keep Alive Enable: YES
NAT Keep Alive Msg: $NOTIFY
NAT Keep Alive Dest: $PROXY

Proxy:
Outbound Proxy: us.voxalot.com
Use Outbound Proxy: YES
Use OB Proxy In Dialog: YES

Last edited by chatalot; 02-25-2009 at 09:13 PM.
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Unread 02-25-2009, 09:37 PM   #5
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Default

chatalot,

If your ATA is not used at any other location then you don't need to depend on a stun server. I would recommend the following settings.

In addition to the SIP ports, forward the entire RTP port range.

Handle VIA received: YES
Handle VIA rport: YES
Insert VIA received: YES
Insert VIA rport: YES
Substitute VIA Addr: YES
Send Resp To Src Port: YES
STUN Enable: NO
STUN Test Enable: NO

NAT Mapping Enable: YES
NAT Keep Alive Enable: NO

Proxy: us.voxalot.com:5060
Outbound Proxy: us.voxalot.com:5060
Use Outbound Proxy: YES
Use OB Proxy In Dialog: YES
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Unread 02-26-2009, 09:35 AM   #6
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Default

Hallo,
Which is RTP port range. I have seen one way audio with FritzBox ATA behinde router. When FritzBox is used als router then every thing is fine. I have used stun settings but even then there was one way audio- out going audio fine but no incomming on some calls.
I don't think that with STUN setting one need any sort of port forwarding. I don't have access to router where my FritzBox ATA is connected so can't do any port forwarding.

Regards.

Majo
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Unread 02-26-2009, 09:46 AM   #7
casch
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Default port forwarding, but ..?

I was thinking of port forwarding as well, but what if my pc - beside my ATA - wants to 'telefone', then all its answers where send to the wrong device - no?

Beside this, all these troubles came up only in the recent past, so why are the
setting now not working, but seemed to work 'once upoon a time'?

Greetings
carl
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Unread 02-26-2009, 09:52 AM   #8
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Quote:
Originally Posted by majo View Post
Which is RTP port range?
Typically this is a setting under user control in the ATA. I am not familiar with the Fritzbox so I can't give specifics about it.

Quote:
Originally Posted by majo View Post
I don't think that with STUN setting one need any sort of port forwarding.
That is true. If you use STUN and NAT-keep-alive packets you won't need to forward any ports. I recommend a 179 second interval between NAT-keep-alive packets. It depends on your router. Most routers are fine with 179 or 299 seconds. Rarely one might have to drop that down to 119 seconds.

Normally, registration to the proxy should be done every 3600 seconds. Some people use frequent registration (like every 3 minutes) as a substitute for NAT-keep-alive packets. That's probably a bad method because it creates unnecessary work for the proxy.

Last edited by boatman; 02-26-2009 at 10:18 AM.
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Unread 02-26-2009, 10:08 AM   #9
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Quote:
Originally Posted by casch View Post
I was thinking of port forwarding as well, but what if my pc - beside my ATA - wants to 'telefone', then all its answers where send to the wrong device - no?
The router keeps a NAT map and will sort out what goes to where. If you use port forwarding for both units then you should make sure each unit is using unique port numbers.

Quote:
Originally Posted by casch View Post
Beside this, all these troubles came up only in the recent past, so why are the setting now not working, but seemed to work 'once upon a time'?
Other than what's posted here in the forum I don't know of any changes on Voxalot's end.

Last edited by boatman; 02-26-2009 at 08:10 PM.
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Unread 02-26-2009, 10:39 AM   #10
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Default getting worse..

ahoy boatman,

things getting worse

I have set now in my ata:


Domainname/Realm: eu.voxalot.com:5060
Local Port: 5060
Proxy Server: eu.voxalot.com:5060
Register Server: eu.voxalot.com:5060
OutboundServer: eu.voxalot.com:5060
RTP Audio Port: 9000
RTP pkt Periode:20
pref. Codec: G.711A

RPORT: activated
STUN activated
(why this might harm?)

And (but?) under sip-status I see:
SIP NAT Typ:
Port Restricted NAT
(what does this mean?)


Ok, as I switched off the stun, I couldn't hear my answ.machine - so same thing the other way around?
Now the echo tests worked twice, but the call to my ans.machine failed twice, both times I couldnt hear what it was saying - I'm going to switch on the stun again.
No change with stun switched on, I could hear my ans.machine, but couldn't speak to it.

Latest test, I called: *0@proxy01.sipphone.com
and I heard: "You are behind a sip-compatible router, you are now ready to ...."

Now boatman please show me the direction.


Carl
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Last edited by casch; 02-26-2009 at 11:23 AM.
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