[Asterisk-Users] FW: 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
>
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