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Unread 03-26-2009, 08:08 AM   #1
docknden1
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Post ATA Frequent Registration Failure

Hi Everyone,

I am encountering a registration problem with voxalot for a while Now.

I have a Linksys PAP2 to which i have assigned a static ip from my router pool and which is registered to my voxalot account,I am using my ATA on non standard SIP ports as (Telephone / ISP ) company is blocking the standard 5060 port.I am also using the STUN parameter which are currently set as follows(OPTION 1 ON WIKI)

Handle Via Received :- NO Handle Via rport :- NO
Insert Via Received :- NO Insert Via rport :- NO
Substitue Via Addr :- YES Send Resp to src Port :- YES
Stun Enable :- YES Stun Test Enable :- NO
STUN Server :- stun.xten.com / stun.voxalot.com:3478

Now the problem is that inspite of me using the non standard sip ports the ATA adapter fails the registration atleast once daily and then i have to change the port to another non standard sip port and restart the ATA after which things are back to normal, This even happens on port 80 which atleast i think cannot be blocked by the ISP.

I would be glad if any one could shed light on the issue and what other setting do i need to do in order to overcome this problem.

Thanks in advance to everyone out there.

Last edited by docknden1; 03-26-2009 at 10:45 AM.
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Unread 03-27-2009, 11:06 AM   #2
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Hi Everyone,

Looking for any inputs on the issue please.
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Unread 03-27-2009, 11:48 AM   #3
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The source of your problem could emerge from variety of circumstances.

1. It could be bad internet connection. Your modem/router/ISP may simply be failing at one point causing connectivity problems for your ATA.

2. What's also important is the type of NAT your router implements for the devices connected behind it. If your NAT is symmetric, STUN will NOT help you. Only outbound proxy will.

3. If you are experiencing problems as it is, you might want to create port forwarding from your router for the designated ports of your ATA. One port for the registration/auth. to take place and the second port for RTP/voice packets. Implementing port forwarding eliminates symmetric NAT-ing and will likely turn in it to Full Cone Nat which will 'open the door' for your phone adapter to communicate with the SIP server much easier.
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Unread 03-28-2009, 06:41 AM   #4
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Quote:
Originally Posted by docknden1 View Post
Hi Everyone, Looking for any inputs on the issue please.
Apparently someone/ something is noticing your VoIP packets. Possibly they/ it only see the SIP packets. I suggest that you reduce the number of SIP packets to the absolute minimum. Maybe STUN packets are noticed too, so stop using STUN as it's not needed.

Do the following:
1. In your router, forward the SIP ports and the RTP port range to the ATA.
2. Set STUN Enable: no
3. Set NAT_Keep_Alive_Enable: no
4. In order for the ATA/phone to know it's public IP address, make sure the ATA/phone is registered with at least one SIP registrar, or enter your public IP address into the appropriate setting in the ATA (in Linksys ATAs this setting is EXT_IP).

Also set:
SIP T1: 1
Reg Min Expires: 3600
Reg Retry Intvl: 600
Register Expires: 3600

We don't know what kind of packet inspection is being used, maybe they have even learned to look for common SIP user agent names. Therefore, the following settings may help.

SIP User Agent Name: Mozilla/5.0 (compatible; MSIE 7.0b; Windows NT 6.0)
SIP Server Name: Mozilla/5.0 (compatible; MSIE 7.0b; Windows NT 6.0)

Put your ATA on an electronic timer to power off during periods when you can't use it, like when you are sleeping. Well, that's all I can think of just now. Let us know how it goes.

Last edited by boatman; 03-28-2009 at 06:08 PM.
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Unread 03-28-2009, 07:25 AM   #5
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Thanks Boatman I'll try to do that and get back with the results.
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Unread 03-28-2009, 07:40 AM   #6
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One more thing. If your router supports UPnP you can enable it as the PAP2 is an UPnP device and thus it will automatically forward the necessary ports for signaling and voice transmission.
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Unread 03-28-2009, 06:15 PM   #7
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If you are using port 80 on Voxalot's end as a browser would then you may want to use fairly high port numbers for the Line 1 and Line 2 SIP ports in the PAP2, as a browser would typically do. Example; SIP port 34496 for Line 1, and 34497 for Line 2.
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