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hermes
Junior
Joined: October 27 2006
Posts: 64
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| Posted: December 17 2009 at 6:32am | IP Logged
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Hello,
we're testing a new SIP network configuration and we've detected one problem.
Our network diagram is the next one:
When Phone 1 calls to Phone 2, connection is established but Phone 2 doesn't hear nothing.
I've been checking sip log and I think I've found the problem, but I don't know how to solvent it.
Phone 1 - SIP Log
Code:
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#2, [12:33:45.406] 4987 Ms, To: 192.168.1.7:5061) >>>>
INVITE sip:SipPhone2@192.168.1.7:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.43:5063;rport
From: SipPhone1 <sip:SipPhone1@192.168.1.7:5061>;tag=2 a00b52c
To: <sip:SipPhone2@192.168.1.7:5061>
Contact: <sip:SipPhone1@192.168.1.43:5063>
Call-Id: 10c7fa3e-3e98-43da-b932-fa27596f6591-00000a10@192.1 68.1.43
CSeq: 67635 INVITE
Max-Forwards: 70
Organization: 94A77FEF-A87E-4AC2-9E66-F56D3B2A063A
Content-Length: 172
User-Agent: LanScape VOIP Media Engine/5.12.8.1 (www.LanScap eCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
v=0
o=SipPhone1 704682890 704682890 IN IP4 192.168.1.43
s=LanScape
c=IN IP4 192.168.1.43
t=0 0
m=audio 9002 RTP/AVP 0
a=rtpmap:0 PCMU/8000/1
a=sendrecv
a=ptime:20
[...]
[...]
[...]
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#4, [12:33:45.483] 57 Ms, From: 192.168.1.7:5061) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.43:5063;received=192.168.1.43;rpo rt=5063
Record-Route: <sip:10.100.11.42:5061>
From: SipPhone1 <sip:SipPhone1@192.168.1.7:5061>;tag=2 a00b52c
To: <sip:sipphone2@192.168.1.7:5061>;tag=523f4eb;x-ori ginal-dest=SipPhone2
Call-ID: 10c7fa3e-3e98-43da-b932-fa27596f6591-00000a10@192.1 68.1.43
CSeq: 67635 INVITE
Contact: <sip:SipPhone2@10.100.1.2:5063>
Server: LanScape Centrex Proxy/3.42.0.2 (www.LanScapeCorp.co m)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 173
v=0
o=LanScape 3470038426 3470038426 IN IP4 10.100.11.42
s=LanScape
c=IN IP4 10.100.11.42
t=0 0
m=audio 9129 RTP/AVP 0
a=rtpmap:0 PCMU/8000/1
a=sendrecv
a=ptime:20
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I've sniffered RTP traffic and all RTP packets from Phone 1 to Phone 2 have got as destination address 10.100.11.42 (LAN IP Proxy) but the proxy is behind Router B (with a NAT) so they would have to be sended to 192.168.1.7
Centrex Proxy configuration:
Code:
Local IP Address: 10.100.11.42
WAN IP Address: 192.168.1.7
Gateway IP Address: 192.168.1.7
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and Media Proxy:
Code:
Local IP Address: 10.100.11.42
WAN IP Address: 192.168.1.7
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I'd be thankful if you could help me.
Thank you very much.
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: December 18 2009 at 9:37am | IP Logged
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Hello hermes,
Good to hear from you - I hope you are well.
I will respond to your post in full this morning... Thanks for your patience!
Thank you,
Randal
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: December 18 2009 at 10:12am | IP Logged
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Hello hermes,
Your LS SIP proxy and media proxy configurations sound OK. The network deployment you have is supported. We will have to investigate further to see what is going on.
You are absolutely correct, the 200 OK response being sent back to phone1 should contain the WAN IP address of NAT router 192.168.1.7.
SIP Proxy Version:
From your SIP, I can see the version of the LS SIP proxy you are using. It is not the latest software. The current version of the LS SIP proxy is v3.42.2.12. We may want to update you.
Media Proxy Version:
What version of the media proxy do you have?
Diagnosing the problem further:
I would like to see the SIP traffic from the standpoint of the LS SIP proxy. Turn on SIP file logging to do this.
Here is what I would like you to do for me if you can
1)
Start the SIP and media proxies. SIP proxy should have SIP logging turned ON.
2)
Allow phone2 to register with LS SIP proxy.
3)
Allow phone1 to register with LS SIP proxy. On the SIP proxy GUI display, is phone 1 being detected as being NATed when the REGISER request is received by the SIP proxy? You can turn ON REGISTER loging in the SIP proxy if you have to. Go to Configure->Proxy Server Settings->SIP Logging and uncheck the REGISTER checkbox in the Filter selected SIP messages from logs group. Phone1 should not be detected as NATed.
4)
Using phone1, make a phone call to phone 2 and allow it to connect. Talk for a moment and then hang up phone1.
5)
Terminate phone1 and phone2 so they unregister with the LS SIP proxy.
6)
Terminate the media proxy and then terminate the LS SIP proxy.
7)
Please send to me the LS SIP proxy log and I will look deeper into the problem. Email the log to supportgroup at lanscapecorp dot com.
Media Proxy WAN IP setting:
You can disable the WAN IP/NAT detection in the media proxy. It is no longer used. You can do this by removing the check box setting Enable WAN IP Address in the media proxys WAN IP/NAT config dialog.
Thank you,
Randal
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: December 18 2009 at 10:14am | IP Logged
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...by the way, excellent network diagram! Thanks! :)
I loaded the image to the LS web server so you can remove the original image from your server if you want.
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hermes
Junior
Joined: October 27 2006
Posts: 64
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| Posted: December 21 2009 at 3:24am | IP Logged
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Hello Randal,
thanks for your help. This morning I'll try to test it and I'll send you all logs.
Thanks again
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LanScape Support Forum -> LanScape VOIP Media Engine - Technical Support
Topic: Network configuration problem
