Dual IP Networking Issue

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rob_acs
Posts: 4
Joined: Mon Jul 02, 2012 9:56 am

Dual IP Networking Issue

Post by rob_acs »

1. Brekeke Product Name and version:
> Brekeke SIP Server 2.4.8.6/286.3 Advanced
2. Java version:
> 1.7.0
3. OS type and the version:
> Linux Centos 2.6.18-238.19.1.el5
4. UA (phone), gateway or other hardware/software involved:
> Voip Provider
5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html :
> 9
6. Your problem:
Server is configured with Dual IP addresses on the same NIC (ie: 208.xx.yy.141 & 208.xx.yy.142, both are public routable addresses on same subnet) we had to do this because our Voip termination provider required a different target ip address for t38 usage.

Server needs to listen and respond on BOTH addresses.

Both addresses are showing under the status screen and are configured in interface1 and interface2 in the sv config file

The server is listening on both interfaces but the problem we are running into is that it seems to respond at the network level to any packets coming in the 142 address using the 141 address

example:
- incoming invite packet from itsp_provider (src) > 208.xx.yy.142 (dst)
> to header is 208.xx.yy.142
> from header is itsp_provider
> contact is itsp_provider
- trying response packet from 208.xx.yy.141 (src) > itsp_provider (dst)
> to header is 208.xx.yy.142
> from header matches invite packet

we verified this with a packet dump on the interface

currently our itsp doesnt seem to have any issues with this but Im sure it will probably break something down the line.

This is also causing an issue with registrations on the second IP, when a phone registers to the 142 address it gets a response back from the 141 address, if the phone is behind nat the router drops it, if the phone is public the phone responds with an error because the source address isnt as expected

settings on config > system screen:
interfaceaddress1 > 208.xx.yy.141
interfaceaddress2 > 208.xx.yy.142
3-5 > blank
auto interface > off
external ip > blank

under config > domains
multiple domains > unchecked
table has both ip addresses listed and everything is on except invite

the sv.properties file doesnt have any restrict or bind options
ambrosio
Posts: 215
Joined: Thu Mar 27, 2008 12:20 pm

Post by ambrosio »

Have you looked at the following wiki topic?
http://wiki.brekeke.com/wiki/Bind-Breke ... IP-address
rob_acs
Posts: 4
Joined: Mon Jul 02, 2012 9:56 am

Post by rob_acs »

yes, the problem is that I dont want it to only use one address I need it to use BOTH addresses
ambrosio
Posts: 215
Joined: Thu Mar 27, 2008 12:20 pm

Post by ambrosio »

How about ... install two SIP servers ?
One is for 208.xx.yy.141 . Another is for 208.xx.yy.142.
voipwell.com
Posts: 528
Joined: Tue Sep 20, 2005 9:10 am
Location: Tannersville, Pennsylvania

Post by voipwell.com »

Hi,

I think $ifdest is what would direct outgoing traffic over a particular interface but don't know if the sip headers will provide the correct info in the contact etc. Record route options are explained also in the Brekeke Advanced SIP Server v3 advanced administrator guide.
rob_acs
Posts: 4
Joined: Mon Jul 02, 2012 9:56 am

Post by rob_acs »

I dont even see that option in the WIKI anywhere, is there documentation for it you know of??

The incoming invite shows phone_num@208.xx.yy.142 so I can pull that off of the invite / to part of the headers if needed

Is that a feature on V3 only??

We just started up and are trying to keep our costs down and cant afford to purchase a second license of the software at the moment, or the cost to upgrade from v2.x to v3.x

thanks
Laurie
Posts: 245
Joined: Mon Jan 07, 2008 12:25 pm

Post by Laurie »

because our Voip termination provider required a different target ip address for t38 usage.

is it for T38's RTP?
rob_acs
Posts: 4
Joined: Mon Jul 02, 2012 9:56 am

Post by rob_acs »

we have one provider for both our standard voip and t38 traffic but they setup the t38 on a seperate account, as a result the endpoint they route calls to needs to be different for both accounts, so we have them routing standard voip to 208.xx.yy.141 and traffic from the t38 account to 208.xx.yy.142

the problem we are seeing and they have already sent us a trouble ticket because its setting off alarms in their system, is that the brekeke sip server is receiving traffic on the 142 address from their servers, but when it responds to the traffic its sending it out using the 141 ip address, which shouldnt happen for normal network traffic at all.

the brekeke sip server is configured to do RTP relay if the client is behind nat

the testing we were doing is currently using public ip's on the UA side of the equation, the address issues im seeing are currently just with the SIP control messages (acks, trying, etc) and it is also affectnig UA's if we try to have them register to the 142 address (we wanted to try and set it up this way so if we do break the t38 accounts to a seperate server in the future we dont have to reprovision a bunch of devices) and they are donig the same, although with the UA's they are dropping the traffic because they arent expecting it to come back from 141

they dont care if the initial invite comes from the wrong ip address since all their incoming invites are routed to one big concentrator, but they need to have the response traffic from a connection from them to one of the two ip addresses to come back out the ip address they originally sent it to.
janP
Posts: 336
Joined: Sun Nov 25, 2007 2:55 pm

Post by janP »

I'm using two separated SIP servers for such a case.
voipwell.com
Posts: 528
Joined: Tue Sep 20, 2005 9:10 am
Location: Tannersville, Pennsylvania

Post by voipwell.com »

The Wiki is a great resource for information. I use it for further clarification of what i find in the Brekeke SIP Server Admin Guide in their documents section. What i referred to was on page 75.

If you haven't seen the document before, check it out. It's amazing how much time and effort puts into their manuals. Along with Brekeke's stability and power, their manuals are their best kept secrets.

You'll find your SIP dial plans will constantly evolve as you find out more new things. However, not having tested these functions i can't assure they are what you are looking for. i don't know what the headers would look like using these functions. My guess ifdest and ifsrc would be correct because it looks so much that the functions were designed for that but you should try it an let us all know. Both would be needed to accomplish what i think you are doing.
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