Multiple IP address problem

[bsotet]

1. Brekeke Product Name and Version: 3.3.4.4/379

2. Java version: 1.7.0_51

3. OS type and the version: Ubuntu 12.04.4

4. UA (phone), gateway or other hardware/software involved: Grandstream GXP1405 1.0.5.32

5. Your problem:
There is an not relevant ip address in SIP message (marked with red color).

Ifconfig:

eth1.900 | 10.110.0.253
eth1.901 | 10.110.1.253
eth1.902 | 10.110.2.253
...etc

/etc/hosts file:
127.0.0.1 localhost **domain**
**PUBLIC_IP** **FQDN** **domain**
10.110.0.253 **FQDN** **domain**
10.110.1.253 **FQDN** **domain**
10.110.2.253 **FQDN** **domain**
...etc

Phone IP: 10.110.1.115

SIP message:

#
U 2014/03/17 13:45:14.241692 10.110.1.115:5069 -> 10.110.1.253:5060
INVITE sip:**CALLED_NUMBER**@10.110.1.253;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.110.1.115:5069;branch=z9hG4bK448043095;rport
Route: <sip:10.110.1.253:5060;lr>
From: <sip:**CALLING_NUMBER**@10.110.1.253;user=phone>;tag=1936434973
To: <sip:**CALLED_NUMBER**@10.110.1.253;user=phone>
Call-ID: 889522868-5069-6@BA.BBA.B.BBF
CSeq: 50 INVITE
Contact: <sip:**CALLING_NUMBER**@10.110.1.115:5069;user=phone>
Max-Forwards: 70
User-Agent: Grandstream GXP1405 1.0.5.32
Privacy: none
P-Preferred-Identity: <sip:**CALLING_NUMBER**@10.110.1.253;user=phone>
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 241

v=0
o=**CALLING_NUMBER** 8000 8000 IN IP4 10.110.1.115
s=SIP Call
c=IN IP4 10.110.1.115
t=0 0
m=audio 5004 RTP/AVP 0 8 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

#
U 2014/03/17 13:45:14.243086 10.110.1.253:5060 -> 10.110.1.115:5069
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.110.1.115:5069;branch=z9hG4bK448043095;rport=5069
From: <sip:**CALLING_NUMBER**@10.110.1.253;user=phone>;tag=1936434973
To: <sip:**CALLED_NUMBER**@10.110.1.253;user=phone>
Call-ID: 889522868-5069-6@BA.BBA.B.BBF
CSeq: 50 INVITE
Server: Brekeke SIP Server rev.379 Evaluation
Content-Length: 0


#
U 2014/03/17 13:45:14.264958 10.110.1.253:5060 -> 10.110.1.115:5069
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.110.1.115:5069;branch=z9hG4bK448043095;rport=5069
From: <sip:**CALLING_NUMBER**@10.110.1.253;user=phone>;tag=1936434973
To: <sip:**CALLED_NUMBER**@10.110.1.253;user=phone>;tag=b6d5f433fs
Call-ID: 889522868-5069-6@BA.BBA.B.BBF
CSeq: 50 INVITE
Server: Brekeke SIP Server rev.379 Evaluation
Proxy-Authenticate: Digest realm="**FQDN**",nonce="8c56a8e2a32a5c5f860d68",algorithm=MD5
Content-Length: 0


#
U 2014/03/17 13:45:14.313735 10.110.1.115:5069 -> 10.110.1.253:5060
ACK sip:**CALLED_NUMBER**@10.110.1.253;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.110.1.115:5069;branch=z9hG4bK448043095;rport
Route: <sip:10.110.1.253:5060;lr>
From: <sip:**CALLING_NUMBER**@10.110.1.253;user=phone>;tag=1936434973
To: <sip:**CALLED_NUMBER**@10.110.1.253;user=phone>;tag=b6d5f433fs
Call-ID: 889522868-5069-6@BA.BBA.B.BBF
CSeq: 50 ACK
Content-Length: 0


#
U 2014/03/17 13:45:14.341707 10.110.1.115:5069 -> 10.110.1.253:5060
INVITE sip:**CALLED_NUMBER**@10.110.1.253;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.110.1.115:5069;branch=z9hG4bK1020451164;rport
Route: <sip:10.110.1.253:5060;lr>
From: <sip:**CALLING_NUMBER**@10.110.1.253;user=phone>;tag=1936434973
To: <sip:**CALLED_NUMBER**@10.110.1.253;user=phone>
Call-ID: 889522868-5069-6@BA.BBA.B.BBF
CSeq: 51 INVITE
Contact: <sip:**CALLING_NUMBER**@10.110.1.115:5069;user=phone>
Proxy-Authorization: Digest username="**CALLING_NUMBER**", realm="**FQDN**", nonce="8c56a8e2a32a5c5f860d68", uri="sip:**CALLED_NUMBER**@10.110.1.253;user=phone", response="820e81a0d9640937a92", algorithm=MD5
Max-Forwards: 70
User-Agent: Grandstream GXP1405 1.0.5.32
Privacy: none
P-Preferred-Identity: <sip:**CALLING_NUMBER**@10.110.1.253;user=phone>
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 241

v=0
o=**CALLING_NUMBER** 8000 8000 IN IP4 10.110.1.115
s=SIP Call
c=IN IP4 10.110.1.115
t=0 0
m=audio 5004 RTP/AVP 0 8 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

#
U 2014/03/17 13:45:14.342521 10.110.1.253:5060 -> 10.110.1.115:5069
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.110.1.115:5069;branch=z9hG4bK1020451164;rport=5069
From: <sip:**CALLING_NUMBER**@10.110.1.253;user=phone>;tag=1936434973
To: <sip:**CALLED_NUMBER**@10.110.1.253;user=phone>
Call-ID: 889522868-5069-6@BA.BBA.B.BBF
CSeq: 51 INVITE
Server: Brekeke SIP Server rev.379 Evaluation
Content-Length: 0


#
U 2014/03/17 13:45:15.871932 10.110.1.253:5060 -> 10.110.1.115:5069
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.110.1.115:5069;branch=z9hG4bK1020451164;rport=5069
From: <sip:**CALLING_NUMBER**@10.110.1.253;user=phone>;tag=1936434973
To: <sip:**CALLED_NUMBER**@10.110.1.253;user=phone>;tag=00E0F5100658DADAFD201ABE2CCC
Call-ID: 889522868-5069-6@BA.BBA.B.BBF
CSeq: 51 INVITE
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY, SUBSCRIBE, UPDATE
Content-Type: application/sdp
Content-Length: 204

v=0
o=- 297504952 0 IN IP4 10.110.0.253
s=session
c=IN IP4 10.110.0.253
t=0 0
m=audio 10018 RTP/AVP 8 101
c=IN IP4 10.110.0.253
a=rtpmap:8 PCMA/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000

#
U 2014/03/17 13:45:17.942546 10.110.1.253:5060 -> 10.110.1.115:5069
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.110.1.115:5069;branch=z9hG4bK1020451164;rport=5069
From: <sip:**CALLING_NUMBER**@10.110.1.253;user=phone>;tag=1936434973
To: <sip:**CALLED_NUMBER**@10.110.1.253;user=phone>;tag=00E0F5100658DADAFD201ABE2CCC
Call-ID: 889522868-5069-6@BA.BBA.B.BBF
CSeq: 51 INVITE
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY, SUBSCRIBE, UPDATE
Content-Type: application/sdp
Content-Length: 204

v=0
o=- 297504952 1 IN IP4 10.110.0.253
s=session
c=IN IP4 10.110.0.253
t=0 0
m=audio 10018 RTP/AVP 8 101
c=IN IP4 10.110.0.253
a=rtpmap:8 PCMA/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000

#
U 2014/03/17 13:45:20.151894 10.110.1.253:5060 -> 10.110.1.115:5069
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 10.110.1.115:5069;branch=z9hG4bK1020451164;rport=5069
Record-Route: <sip:10.110.0.253:5060;lr>
From: <sip:**CALLING_NUMBER**@10.110.1.253;user=phone>;tag=1936434973
To: <sip:**CALLED_NUMBER**@10.110.1.253;user=phone>;tag=00E0F5100658DADAFD201ABE2CCC
Call-ID: 889522868-5069-6@BA.BBA.B.BBF
CSeq: 51 INVITE
Contact: <sip:10.110.0.253:5060>
Allow-Events: refer
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY, SUBSCRIBE, UPDATE
Supported: 100rel, timer, replaces
Content-Type: application/sdp
Content-Length: 204

v=0
o=- 297504952 2 IN IP4 10.110.0.253
s=session
c=IN IP4 10.110.0.253
t=0 0
m=audio 10018 RTP/AVP 8 101
c=IN IP4 10.110.0.253
a=rtpmap:8 PCMA/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000

#
U 2014/03/17 13:45:20.646678 10.110.1.253:5060 -> 10.110.1.115:5069
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 10.110.1.115:5069;branch=z9hG4bK1020451164;rport=5069
Record-Route: <sip:10.110.0.253:5060;lr>
From: <sip:**CALLING_NUMBER**@10.110.1.253;user=phone>;tag=1936434973
To: <sip:**CALLED_NUMBER**@10.110.1.253;user=phone>;tag=00E0F5100658DADAFD201ABE2CCC
Call-ID: 889522868-5069-6@BA.BBA.B.BBF
CSeq: 51 INVITE
Contact: <sip:10.110.0.253:5060>
Allow-Events: refer
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY, SUBSCRIBE, UPDATE
Supported: 100rel, timer, replaces
Content-Type: application/sdp
Content-Length: 204

v=0
o=- 297504952 2 IN IP4 10.110.0.253
s=session
c=IN IP4 10.110.0.253
t=0 0
m=audio 10018 RTP/AVP 8 101
c=IN IP4 10.110.0.253
a=rtpmap:8 PCMA/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000

#
U 2014/03/17 13:45:21.649313 10.110.1.253:5060 -> 10.110.1.115:5069
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 10.110.1.115:5069;branch=z9hG4bK1020451164;rport=5069
Record-Route: <sip:10.110.0.253:5060;lr>
From: <sip:**CALLING_NUMBER**@10.110.1.253;user=phone>;tag=1936434973
To: <sip:**CALLED_NUMBER**@10.110.1.253;user=phone>;tag=00E0F5100658DADAFD201ABE2CCC
Call-ID: 889522868-5069-6@BA.BBA.B.BBF
CSeq: 51 INVITE
Contact: <sip:10.110.0.253:5060>
Allow-Events: refer
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY, SUBSCRIBE, UPDATE
Supported: 100rel, timer, replaces
Content-Type: application/sdp
Content-Length: 204

v=0
o=- 297504952 2 IN IP4 10.110.0.253
s=session
c=IN IP4 10.110.0.253
t=0 0
m=audio 10018 RTP/AVP 8 101
c=IN IP4 10.110.0.253
a=rtpmap:8 PCMA/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000

[janP]

There are several ways to solve the issue.
Let you read the documents and wiki.

One solution is "bind".
http://wiki.brekeke.com/wiki/Bind-Brekeke-SIP-Server-to-one-IP-address
----------------------
net.bind.interface = 10.110.1.253
net.net1.interface = 10.110.1.253
net.net1.interface-restrict = 10.110.0.253
net.net2.interface-restrict = 10.110.2.253
----------------------


Another solution is "Remote Address Pattern".
----------------------
In the [Configuration]->[System] page.
Set the [Interface address 1] = 10.110.1.253
Set the [Remote Address Pattern 1] = ^10.110.1.
----------------------

Another solution is "$ifdst" and "$ifsrc".
Read the document.
http://www.brekeke.com/doc/sip/sip_admin_v3.pdf

[bsotet]

Dear JanP!

Thank you for your answer. We have lot of (70+) VLAN interfaces, therefore the first and the second solution is not possible to set in web GUI.
It is possible to set the routing with rules, but what do you think the overall server performance with 70+ rules?

[janP]

Hi bsotet,
70+ rules will not impact the performance much if your SIP Server is an Advanced Edition.