Re: SIP, NAT, and load balancing problems

Linux Advanced Routing and Traffic Control

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I unfortunately think that I can't use that solution (if I understood it well).

My box actually has two functions, it's an Asterisk box and a load balancing router.

For LAN clients, as this box represents their default gateway, there would be no problem in implementing a pure routing solution. I could create a new subnet on a dummy interface, and reconfigure all LAN SIP phones to point to that IP, the box itself would route packets to its dummy interface.

For WAN clients, what I need is to have a unique interface (a unique public IP) accepting SIP connections, and outgoing traffic always passing by this interface. My current issue is with outgoing SIP/RTP traffic that sometimes gets load-balanced and uses the other public IP, which i have to force to the other interface, with lots of NATing/Re-Routing problems, as a single call can have multiple UDP flows (SIP and RTPs).

My main problem with the DummyNet solution on the WAN side is that I cannot access to the internet routers behind this box, so I can't add routes to reach a new subnet. This means that I'm back with the same type of problem trying to NAT, but this time not only the box's outgoing traffic, but also the clients incoming traffic, for them to reach the dummy0 interface.

Tell me if i'm wrong, but that solutions appears to me as more complicated in my particular case. Aouch, that's much harder than I thought it would be. :-(

François.


Grant Taylor wrote:
François Delawarde wrote:
What i meant is that people (in #asterisk on freenode) told me that Asterisk could be bound to a unique IP, or to all IPs (binding it to 0.0.0.0). But if you know a way to bind it to only some IPs, then yeah! I need your help :-) I guess we need to put something in the bindaddr parameter of sip.conf. Right now I have:

[general]
bindaddr=0.0.0.0

I have 3 IPs in 3 interfaces:
eth0 (LAN): 192.168.10.1
eth1 (WAN): 192.168.1.2 (gw 192.168.1.1)
eth2 (WAN): 192.168.2.2 (gw 192.168.2.1)

How can I bind Asterisk SIP to 192.168.10.1 and 192.168.2.2 only, to work around my load balancing problem?

I'll email you off the mailing list as this does not pertain to LARTC.

If Asterisk is only listening to one IP and you are routing to get to your other network, you could end up with some really weird issues that will be very difficult to over come, probably MUCH harder than resolving the issue with Asterisk only binding to one interface.

I don't really understand what you mean, but that's right, i have really weird issues.

What I was saying is that if Asterisk is only bound to one IP address,
be it loopback, eth0, eth1, or even a dummy0 interface, you will have to
route traffic to that address.

If you can indeed only bind Asterisk to only one IP address or all IP
addresses on the system, I would recommend that you use DummyNet to bind
Asterisk to.  However this may be a problem down when NATing comes in to
play.  (More on this later.)

Supposing that you bind Asterisk to the dummy0 interface, either all
equipment will need to its self know how, or the default router for the
equipment will need to know how to reach the subnet on the dummy0
interface.  This usually means that you will have to have the default
gateway for all client systems / phones know how to reach the subnet on
the dummy0 interface.  I.e. the default gateway will have to have a
route to the subnet on the dummy0 interface via the interface on the
Asterisk box facing the router(s).

Consider:
                   +----------------------+
                   |     Asterisk Box     |
                   |         [A.B.C.D/NM]-|---(INet)
(192.168.0.0/24)---|-[192.168.0.254/24]   |
                   |   [192.2.0.254/24]   |
                   |          |           |
                   +----------------------+
                              |
                           [dummy0]

In this case, 192.168.0.254/24 is the LAN, the internet is it's own IP, and 192.2.0.254/24 is assigned to the dummy0 interface. If you bind Asterisk to the 192.2.0.254 IP on the dummy0 interface, you will have to route all traffic that is to or from Asterisk in to and out of the dummy0 network.

Now that you can easily see that you would have to route traffic in to and out of the dummy0 interface, I can probably better explain the weird routing issue that you have. You are binding Asterisk to an IP on your system. No matter what IP you bind Asterisk to, traffic from any other subnet will have to be routed to that subnet to reach Asterisk.

With this in mind, now consider if you bind Asterisk to one WAN interface, traffic to / from your LAN or the other WAN interface will have to be routed to be able to reach Asterisk. If you bind Asterisk to the LAN interface, traffic to / from either WAN will have to be routed to be able to reach Asterisk.

Usually routing traffic is not an issue. However, as you have pointed out, when you MASQUERADE traffic as it leaves either of your WAN interfaces, the port numbers are changed and thus breaking your SIP connection.

So, you need to be able to not alter the SIP packet stream. So, what you need to really do is only alter traffic that is not originating / terminating on your firewall. You could do this a few different ways. Probably the easiest way would be to not MASQUERADE any traffic, save for traffic that originates on your LAN, not the firewall / Asterisk box it's self.

You will probably also need to do something to make sure that your SIP traffic is not subject to load balancing. If you set up some sort of identifier for your SIP traffic, say locally originated / terminated, you could use a custom routing table to not load balance the traffic via multiple next hops.

One advantage of having Asterisk bind to a completely different IP, i.e. on the dummy0 interface is that you could set up a rule that looked for source or target IPs in the subnet on dummy0 as a VERY easy and clear identifier as the traffic would belong to Asterisk.

What happens in my case, where default subnet (0.0.0.0/0 subnet) has two IPs (2 WAN with load balancing)? And do you know at what moment this IP is chosen? Do you think I can trick the routing subsystem (or whoever decides the IP) to force the decision?

Sorry, I don't know what moment the decision is made. Nor do I think you could ""Trick the routing sub system once it has made a decision. Sure, you can do some things to over ride which interface is used to carry out the decision that was made. I think what would be better would be to influence / control the possibilities that the routing sub system has to choose from.

I'll try to check on that, if i can't resolve the issue with Asterisk bindings.

*nod*



Grant. . . .
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