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Unread 09-30-2009, 10:34 AM   #1
casch
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Default the first call (even *600) after registration is not working?

Hi,

I have a stange problem - maybe s.o. can help?

The first call after registration of the phone at voxalot fails allways.
I hear everything but nothing I say is 'broadcasted'.
Even if I call *600, the echo test: I hear everything, but nothing I say is repeated. I hang up and redial *600 fine, no problem!

Same thing if I want to make an outbound call after registration - nobody hears me.
If I call the second time - now it works?

NAT is set to NO. voxalot's stun is entered.

My WLAN-phone behind a WLAN-Router (dd-wrt).
The phone's MAC is set to the max pass-through (QoS: Premium).
The phone registers first at the router, hereafter at eu.voxalot.com: No problem.

What can I do that even my first call after my phone's registration at eu.voxalot.com works correctly?

Thanks in advance
Carl
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Last edited by casch; 09-30-2009 at 10:37 AM.
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Unread 10-01-2009, 04:11 AM   #2
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See this post:

http://forum.voxalot.com/voxalot-sup...html#post25221
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Unread 10-01-2009, 07:51 AM   #3
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Angry where is Voxalot's Router-Setup Howto?

Thanks Ron,
at your link I can read:
Quote:
One-way audio problem are frequently (nearly always?) caused by routing/router/NAT issues. The first step is to try using a STUN server if you're not already. The next is to forward the RTP range of ports, and if still having problems, the SIP port(s).
I am using voxalot's stun and still have that problem!
Due to the fact that this appeares even with the echo-test *600, it is s.th. between voxalot and me - fine, at least we can address the problem.

So this raises some questions, that are essential but not answered anywhere @voxalot!

1) what are the rtp-ports (Voxalot uses)?
Here I read that rtp uses:
0 16383 unclassified lowest
16384 32767 audio highest
32768 49151 whiteboard medium
49152 65535 video low

Does that mean I should forward the range from 0 - 32767 to my wlan-phone?
Seems a bit stange to me! See 3).
2) The sip-port Voxalot uses 5060 (and others?)
Voxalot uses 5060 - can I use for the other 5061?
I read (in german) that SIPPS needs a second port two counts higher - is this required by voxalot?

Can I use a unique voxalot port not to interfear with others?
3) what if my wlan-phone is not the only voip-device?
What can I do if I like to do Voip (skype and other provider) via my pc (at least one second voip-device) as well, not only with my wlan-phone.

Wouldn't it be nice if voxalot usese a set of 'free' and 'unique' port numbers to make the router set up and the port forwarding much easier but under the condition that both (voxalot)-provider and the Voip-device are not the only one in the world!!
Why is there nothing @voxalot that describes a clean setup of a router?

I am sure that I am not the only person that experiences theses problems and Voxalot knows best what ports they are using. So where is the HowTo for the router/Firewall - all what I found by google ends with: call you administrator

Greetings
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Unread 10-01-2009, 09:00 PM   #4
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Your ATA/Sip Phone/Soft Phone determines what RTP ports will get used. You would forward the RTP ports that it will use to that particular device.

AFAIK, SIP only uses a single port. The default is 5060, but you can also use 5061, 5062, etc.

With multiple VoIP devices, you would have to set each device to a particular range of RTP ports and use multiple forwarding entries to route the appropriate ranges to the appropriate device. Skype doesn't use SIP/RTP and handles itself properly in most cases without any help.

The relationship between SIP and RTP is extremely unclean at the protocol design level when it comes to routing behind and through NAT routers. This has nothing to do with Voxalot. Audio problems with VoIP where NAT routers are involved are a significant problem with all providers and aggregators. I wish I could point you to a one-size fits all cookbook set of entries to make in your router, but it simply doesn't exist.
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Unread 10-06-2009, 07:56 AM   #5
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Hmm, Roon,

bad news.
But due to this unsatifactory situation I'd like to suggest a second fast echo test.

*600 explains everything needed to know for a first time use -ok!
But in case you want to use it anytime you register it is pretty ennoying to wait until *600 has finished it explanation to ckeck the other audio-way by saying s.th..

So I suggest to establish a fast echo-test *666 (easy and fast to enter!) that just says "Voxalot's echo test" or "Voxalot's echo test. Now speak and listen" or s.th. like this?

By the way - I once suggested this - it really could help to localize problems if voxalot's voice error messages would be like "Sorry Voxalot can't do this"!

See, here in Vienna is one provider (#1) offering a access to e164.org (#2) to forward a call to voxalot (#3). Beside that I have two providers (#4,#5) and none all these 5 tell in case of an error, where this error appears. Bad service by all!

At least Voxalot should start to be better by adding just Voxalot to all its error messages - no?

Thanks,
Carl
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Unread 10-07-2009, 04:05 PM   #6
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Don't know if this helps you,
but Brujula, a voip proider, has a echo test that starts, without intro, so maybe, by peering to their sipbroker code you can also do your tests ?
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