Nat VOIP

[hakim12help]

Hello, I have a Cisco 1900 series which I use as footbridge SDSL for the voice over IP. I have dish ready(in position) a static nat towards my waiter(server) of voice(vote), but the problem it is on some calls(appeals) I do not receive the flow RTP coming from my operator. This below the config of my router

interface GigabitEthernet0/1
ip address privat ip submask
ip nat inside
no ip virtual-reassembly in
duplex full
speed 100
!
interface Ethernet0/1/0
ip address public ip mask
ip nat outside
ip virtual-reassembly in
!
ip forward-protocol nd
ip nat inside source static private ip public ip
ip route 0.0.0.0 0.0.0.0 passerelle opérateur
thanks,

[Philip D'Ath]

Your config looks good to me. If this is a SIP service then try adding this:

no ip nat service sip udp port 5060

[Meylaers Jan]

Hi,

you should check your pbx or voipgateway's configuration. Perhaps you are sending an unknown connection information address into the first invite towards your provider. Your provider receives a sip invite with a rtp pointer address that is equal to a private address and it does not know route back toward a private address.

2 options there;

1- ask your provider to configure nat on your sip trunk. In this case the provider will discard the value of the sip body into the sip invite packet and always sent back to the source address from where the packet is coming from. 

2-make sure that the sdp connection information contains the public address of your router @all time so that the rtp can be routed your way.

3- as Philip said. disable sip nat on your router.

no ip nat service sip udp port 5060

no ip nat service sip tcp port 5060

Jan Meylaers