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Unread 11-12-2009, 12:25 PM   #1
Tranworld
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Default Strange one way audio

My voxalot is 968032 and I have the following providers for dialling out

VoIPCheap
Pfingo

my E52 can logon to Voxalot and make call out but call through Pfingo only have one way in that I can't hear the other side but they can hear me. But if I call the same number using VoIPCheap then the audio is fine.
Now if I logon to pfingo directly then the audio is also fine, therefore only via Voxalot that the audio in pfingo only is disrupted, the question is where should I look ?

By the way don't blame the E52 since my E51 also displayed the same problem

Remember VoIPcheap works fine and pfingo also works fine on the same phone if used directly.
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Unread 11-12-2009, 12:45 PM   #2
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Default

Quote:
Originally Posted by Tranworld View Post

By the way don't blame the E52 since my E51 also displayed the same problem
.
Actually that is not correct, the E51 works fine on all account, this seems to be a problem with the new E52. Does anyone has any experience running Voxalot via the E52? How to make it work?
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Unread 11-12-2009, 06:30 PM   #3
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Quote:
Originally Posted by Tranworld View Post
...the E51 works fine on all account, this seems to be a problem with the new E52. Does anyone has any experience running Voxalot via the E52? How to make it work?
Are the STUN settings identical in both phones? In your Voxalot account, do you have "Symmetric NAT Handling" set yes?
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Unread 11-12-2009, 11:10 PM   #4
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Default Stun

There is no way to set the STUN parameter in the Nokia, both E51 and E52, I'll be grateful if someone can show me where?

However the symmetric NAT handling did it, the E52 now also work, I will check whether the E51 continues to work without problem. Thank you so much
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Unread 11-13-2009, 06:32 PM   #5
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Apparently your E52 supports STUN, see this page. Maybe STUN is not yet configured in your phone. Without STUN the E52 uses it's private IP address/port inside SIP packets, including the RTP contact address, the address/port to which the other party attempts to send the voice packets. If the VSP does not support SIP re-routing of RTP stream you will still hear the remote party because RTP packets come to you through Voxalot. It is good to use Voxalot's "Symmetric NAT Handling" because that routes voice packets through Voxalot only when necessary to complete the call.
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