[Asterisk-Users] How to bind RTP when IP alias are configured

[SW]

Hi Folks,

I have a situation where my Colo insists on a particular IP setup for my *
server box. They allocate two blocks of IPs to my colo server. One set as my
own (ex 20.20.20.20.4/30 - 4 ips) and the other as a transit lan (es
10.10.10.0/29). These are all public IP addresses and there is no NAT
involved in.

So essentially I have to set-up IP aliases in my Linux box as follows;

Example:

TRANSIT LAN:		10.10.10.0/29
CUSTOMER LAN:		20.20.20.20.4/30

RedHat LINUX

FILE: /etc/sysconfig/network-scripts/ifcfg-eth0

DEVICE=eth0
IPADDR=20.20.20.20.4
NETMASK=255.255.255.255
ONBOOT=yes

FILE: /etc/sysconfig/network-scripts/ifcfg-eth0:99

####################################
## TRANSIT IP: DO NOT UNCONFIGURE ##
####################################
DEVICE=eth0:99
IPADDR=10.10.10.4
NETMASK=255.255.255.248
NETWORK=10.10.10.0
BROADCAST=10.10.10.3
GATEWAY=10.10.10.1
ONBOOT=0

----------------------------
First of all. I can ping to customer lan and telnet to it, therefore IP
routing (at least for unicast traffic) works fine.

Now question arises when asterisk start to work on this box. Since the IP
that I am supposed to use is 20.20.20.4, I set that as bindaddress in my
sip.conf file. As far as SIP messages are concern * users that IP address,
no problem. However for RTP stream * users 10.10.10.4 as it's source
address. Because of this obviously calls will not go through asterisk, as
the ip phone is expecting RTP packets from the SIP server which is bound to
IP 20.20.20.4.

Is there a way to tell * to use the same bind address in SIP.conf
(h323.conf, iax.conf) for RTP ?????

I read rtp.conf file but that does not show any bind address.

It seems like LINUX always select it's src address as the interface (alias)
which has the gateway tied to it unless otherwise an application
specifically asks Linux to use a particular ip address.

Cheers

SW