Nating for VOIP on Cisco IOS

Hello Michael,

In my opinion it is not a good idea to NAT all of the VOIP server traffic like that. It is a major security concern. Usually NAT is used to allow access to the VOIP server for remote management. What are you trying to accomplish with this setup? If you want to allow remote (internet) users to call extensions behind the server, than you should setup a VPN.

To your question - I think there is no audio because all SIP traffic goes directly through the server, unlike audio which uses RTP and is established between the endpoints only - bypassing the VoIP server completely. For example - when you are dialing,  the source device sends a SIP message to the server, the VOIP server relays the SIP signals to the destination device, which makes the phone to ring. Now, once the phone is answered the audio is not relayed through the server, instead it is established between the caller and the called endpoint devices.

SIP Path: VoIP Phone <-> VOIP Server <-> VoIP Phone
RTP Path: VoIP Phone <-> VoIP Phone

Also, I found the following topology for a better visual representation - http://i.technet.microsoft.com/Ee690384.468a93eb-2b07-49be-96a2-ab158830c0e5%28en-us,TechNet.10%29.jpg

I hope that helps. Please let me know if you have any questions

Hi, i'm aware of the security concerns however at this point am not concerned. I'm just trying to understand why this is not working. If i'm natting all ports form y external IP to my voip server there should be nothing stopping this from working yet it doesn't. Im using a SIP trunk from my asterisk box so i don't believe my client should be establishing direct RTP connections to my trunk provider. Additionally I have a pstn card installed with a line attached. Again from outside i cannot hear audio however inside can.

For the asterisk gurus, I have the SIP iP settings configured correctly in terms of defining my External IP and internal IP addresses to ensure correct behavior across NAT.