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Posted by dkerr on 20.02.2006 at 17:23:

Sipura 2002

I'm having problems setting up my Sipura 2002 to work properly. I can register and I can send/receive calls. However there seems to be a problem with the REGISTER timeout. The Sipura thinks that the timout is immediate and it has to register again straight away. They results is an awful lot of REGISTER traffic, like it is in a loop.

I do not have this problem with my X-Lite softphone so it is possibly a configuration error in my Sipura setup?

Here is a trace from Sipura (read from bottom up). The Sipura receives an OK that clearly shows an expire of 600, but the Sipura says RegOK NextReg in 1 (1) and immediately tries to REGISTER, receives a TRYING then an OK which it determines is ok but needs to register again in 1 second.

02-20-2006 09:59:40 Local7.Debug 192.168.1.4 [0]RegOK. NextReg in 1 (1)
02-20-2006 09:59:40 Local7.Debug 192.168.1.4 [0]ExtIpChanged:0
02-20-2006 09:59:40 Local7.Debug 192.168.1.4 <010>
02-20-2006 09:59:40 Local7.Debug 192.168.1.4 <010>
02-20-2006 09:59:40 Local7.Debug 192.168.1.4 SIP/2.0 200 OK<013><010>Via: SIP/2.0/UDP 69.177.32.9;branch=z9hG4bK-fd4ce4b3;received=217.195.32.11<013><010>From: David Kerr <sip:dkerr-103@pbx.i-p-tel.com>;tag=89fa8f7997cec199o0<013><010>To: David Kerr <sip:dkerr-103@pbx.i-p-tel.com>;tag=as7bfab74a<013><010>Call-ID: 7380eef9-cf2fe819@192.168.1.4<013><010>CSeq: 723 REGISTER<013><010>User-Agent: Asterisk PBX<013><010>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<013><010>Max-Forwards: 70<013><010>Expires: 600<013><010>Contact: <sip:dkerr-103@217.195.32.11><013><010>Date: Mon, 20 Feb 2006 14:59:37 GMT<013><010>Content-Length: 0
02-20-2006 09:59:40 Local7.Debug 192.168.1.4 [0:5060]<<217.195.32.11:5060
02-20-2006 09:59:40 Local7.Debug 192.168.1.4 [0:5060]<<217.195.32.11:5060
02-20-2006 09:59:40 Local7.Debug 192.168.1.4 <010>
02-20-2006 09:59:40 Local7.Debug 192.168.1.4 <010>
02-20-2006 09:59:40 Local7.Debug 192.168.1.4 SIP/2.0 100 Trying<013><010>Via: SIP/2.0/UDP 69.177.32.9;branch=z9hG4bK-fd4ce4b3;received=217.195.32.11<013><010>From: David Kerr <sip:dkerr-103@pbx.i-p-tel.com>;tag=89fa8f7997cec199o0<013><010>To: David Kerr <sip:dkerr-103@pbx.i-p-tel.com><013><010>Call-ID: 7380eef9-cf2fe819@192.168.1.4<013><010>CSeq: 723 REGISTER<013><010>User-Agent: Asterisk PBX<013><010>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<013><010>Max-Forwards: 70<013><010>Contact: <sip:dkerr-103@217.195.32.11><013><010>Content-Length: 0
02-20-2006 09:59:40 Local7.Debug 192.168.1.4 [0:5060]<<217.195.32.11:5060
02-20-2006 09:59:40 Local7.Debug 192.168.1.4 [0:5060]<<217.195.32.11:5060
02-20-2006 09:59:39 Local7.Debug 192.168.1.4 <010>
02-20-2006 09:59:39 Local7.Debug 192.168.1.4 <010>
02-20-2006 09:59:39 Local7.Debug 192.168.1.4 REGISTER sip:pbx.i-p-tel.com SIP/2.0<013><010>Via: SIP/2.0/UDP 69.177.32.9:5060;branch=z9hG4bK-fd4ce4b3<013><010>From: David Kerr <sip:dkerr-103@pbx.i-p-tel.com>;tag=89fa8f7997cec199o0<013><010>To: David Kerr <sip:dkerr-103@pbx.i-p-tel.com><013><010>Call-ID: 7380eef9-cf2fe819@192.168.1.4<013><010>CSeq: 723 REGISTER<013><010>Max-Forwards: 70<013><010>Authorization: Digest username="dkerr-103",realm="asterisk",nonce="4608f19b7f74ba781fe0ef5927b764e075037955",uri="sip:pbx.i-p-tel.com",algorithm=MD5,response="b967234f1d52f1a50e21994fe9732a44"<013><010>Contact: David Kerr <sip:dkerr-103@69.177.32.9:5060>;expires=600<013><010>User-Agent: Sipura/SPA2002-3.1.5<013><010>Content-Length: 0<013><010>Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER<013><010>Supported: x-sipura
02-20-2006 09:59:39 Local7.Debug 192.168.1.4 [0:5060]->217.195.32.11:5060
02-20-2006 09:59:39 Local7.Debug 192.168.1.4 [0:5060]->217.195.32.11:5060
02-20-2006 09:59:39 Local7.Debug 192.168.1.4 RSE_DEBUG: reference domain:pbx.i-p-tel.com
02-20-2006 09:59:39 Local7.Debug 192.168.1.4 [0]REG: STUN c0a80104->45b12009, 5060->5060
02-20-2006 09:59:39 Local7.Debug 192.168.1.4 [5060]STUN trying 0
02-20-2006 09:59:38 Local7.Debug 192.168.1.4 RSE_DEBUG: unref domain, pbx.i-p-tel.com
02-20-2006 09:59:38 Local7.Debug 192.168.1.4 [0]RegOK. NextReg in 1 (1)
02-20-2006 09:59:38 Local7.Debug 192.168.1.4 [0]ExtIpChanged:0
02-20-2006 09:59:38 Local7.Debug 192.168.1.4 <010>
02-20-2006 09:59:38 Local7.Debug 192.168.1.4 <010>
02-20-2006 09:59:38 Local7.Debug 192.168.1.4 SIP/2.0 200 OK<013><010>Via: SIP/2.0/UDP 69.177.32.9;branch=z9hG4bK-22ec6339;received=217.195.32.11<013><010>From: David Kerr <sip:dkerr-103@pbx.i-p-tel.com>;tag=89fa8f7997cec199o0<013><010>To: David Kerr <sip:dkerr-103@pbx.i-p-tel.com>;tag=as7bfab74a<013><010>Call-ID: 7380eef9-cf2fe819@192.168.1.4<013><010>CSeq: 722 REGISTER<013><010>User-Agent: Asterisk PBX<013><010>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<013><010>Max-Forwards: 70<013><010>Expires: 600<013><010>Contact: <sip:dkerr-103@217.195.32.11><013><010>Date: Mon, 20 Feb 2006 14:59:36 GMT<013><010>Content-Length: 0
02-20-2006 09:59:38 Local7.Debug 192.168.1.4 [0:5060]<<217.195.32.11:5060
02-20-2006 09:59:38 Local7.Debug 192.168.1.4 [0:5060]<<217.195.32.11:5060



Thanks
DAK

Posted by dkerr on 21.02.2006 at 00:38:

Maybe STUN settings?

Further testing indicates that the problem I describe above may have something to do with my STUN settings. If I have STUN enabled then I get the problem described above. If I have STUN disabled then the register "expire" time works fine.

Why?

DAK

Posted by supernettel on 24.03.2006 at 17:56:

What firmware versionm is in the Sipura, have you updated it recently?

Posted by dkerr on 24.03.2006 at 20:17:

firmware version is 3.1.5. Further investigation (google search) turned up several other people reporting the same problem with Sipura and STUN settings. It appears that there is a bug which causes the Sipura to think that the IP address is changed and that it has to register again immediately.

In my case, I have discovered that I can disable all STUN and NAT support in the Sipura and I still work. It seams that my NAT/Firewall (Linksys WRT54G with firmware v4.20.7) does not require it... but I did setup port forwarding to send all 5060/5061 and 16384 to 16482 ports directly on to the Sipura. This is proving the most reliable way to get my VoIP working.

DAK

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