Voice quality within Voxalot - bad

[boatman]

I believe the problem will be solved if the two ATAs can be made to send RTP packets directly to each other. I don't know why the ATAs are not already doing that during a Voxalot to Voxalot call.

[Tranworld]

It may or may not but even then it is a partial solution because I am also using a Nokia E51 with its build in VoIP stack, not much can be configured there. Looks like I have to use the long route via provider, in itself not a problem for me and does not reduce the Voxalot real benefits, just perplexing why that is so. Thanks for helping anyway

[boatman]

I have been experimenting here. It seems that Voxalot will give the calling party the direct route only if the receiving party's NAT traversal settings are a set certain way. If Voxalot detects incorrect NAT traversal settings on the receiving end, then it forces the RTP packets through the Voxalot proxy server (this is what's happening in your case), presumably this is done in order to successfully complete the call, whereas some other VSPs might connect you but you would have one-way audio or no audio.

[boatman]

Originally Posted by Tranworld

Looks like I have to use the long route via provider, in itself not a problem for me and does not reduce the Voxalot real benefits, just perplexing why that is so...

Martin has confirmed what I suspected about the NAT traversal settings having something to do with getting a direct RTP connection. See his post here.

If you wish to test for mis-configured NAT traversal settings you can call this SIP number: *0@proxy01.sipphone.com . The easiest way to call that number may be to place it into one of the ATA's Speed Dial slots, "Speed Dial 9" for example, then press 9# to dial the number. If the test says "you may experience problems behind this router" then you know you have mis-configured NAT traversal settings.

[emoci]

First of all thanks boatman for taking the time to look into this thoroughly...

As a sidenote, instead of calling the SIP URI, you can also call *747*0 to reach the testing facility at SipPhone

[boatman]

Originally Posted by emoci

As a side note, instead of calling the SIP URI, you can also call *747*0 to reach the testing facility at SipPhone

I would suggest sticking with the fundamental test number, *0@proxy01.sipphone.com. The *747*0 method works only if NAT settings are correct, however if certain NAT settings are incorrect (such as 'NAT Mapping Enable' = no in a PAP2) then no message will be heard and the caller may wonder what happened to the expected voice announcement.

[boatman]

Originally Posted by Tranworld

How can we explain that? surely I would have expected that the Voxalot to Voxalot call, cutting out Pfingo, to produce the clearest sound, but the reverse is happening which I can't understand

Hi Tranworld,

After reading this post by emoci, I realized that your poor quality audio issue can probably be solved by following these instructions.