Load balance SIP/RTP
cervis last edited by
we are using pfsense 1.2.3-RELEASE with two WAN and two LAN interfaces.
WAN1 is a less quality cable connection
WAN2 is a high quality SDSL connection
LAN1 is for all network protocols other than SIP/RTP
LAN2 is especially reserved for SIP/RTP protocol
Unfortunately, the high quality SDSL line has a limited capacity of 6 VoIP connections.
Now, I am looking for a solution to use the less quality cable line as a fallback for the rare situation of more than 6 parallel VoIP connections.
But, I have a problem, I have just one asterisk v1.4 server and one SIP provider. So, on both ends of the VoIP connections there are only single IP addresses. The asterisk server can be configured to use another line, if the first line reaches a maximum of 6 VoIP connections, but the second line has to use the same destination IP address. So I can not differentiate the two lines within the pfsense gateway.
Is it possible to rewrite the destination IP address of outgoing packets on the WAN1 interface? Because, I could use a fake IP address for the second line in the asterisk configuration.
Does anyone have another idea?