[asterisk-users] SIP and RTP on different IP's

[Duncan Turnbull]

On 24/11/2012, at 2:19 AM, Tiago Geada <tiago.geada at gmail.com> wrote:

> Hello Folks, I am looking for a way that makes asterisk tell remote SIP party that the IP where they will send RTP is not the same as the one I am comunicating via SIP
> 
> Can this be done anyhow?
> 
> I can try and explain:
> 
> We have placed a asterisk box in our partners office.
> 
> It has eth0 with IP 172.16.1.10 and eth1: 10.34.18.250
> 
> linux has its routes set so it can comunicate with several networks in their offices.
> 
> now there is a cisco call manager that we need to communicate with. Normally via our IP 172.16.1.10, however seems that this cisco uses some sort of 'directmedya=yes' and sets both ends speaking RTP with themselves.
> 
> There are some extensions in cisco that have a network 10.134.0.0/16 that we can only comunicate via eth1
> 
> thus when calling cisco (always via eth0) sometimes we need to say that OUR IP to recieve RTP is not 172.16.1.10, but 10.34.18.250
> 

This is a routing issue, not asterisk I think. You are saying you route to cisco via eth0, it sets up connections to its end points and then drops out of the media flow, but the end points have no route to the eth0 address so they fail

Linux usually sorts this out and asterisk replies on the address of the interface it sends out with. So for the most part the response in my experience if its going out eth1 should use the eth1 ip address.

If you can get to it via eth0 and thats the preferred route then it will have the eth0 address. If so why can't you change your routing table to use eth1 when you need to go to the cisco then you will have the right address and the far extensions can respond to you correctly

Or change the cisco network endpoints so they can successfully access your address on eth0


> can this be done?
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