If your question is specific to Cisco environment, then Gordon and Carlo has been specified the same.
Otherwise in standard VoIP and specifically standard SIP environment when used over WAN, you will frequently see NAT and PAT being used when SIP messages are transported over WAN. Two most frequently deployments you will ever see;
1. Terminals (deskphones) are behind the NAT
Considering branch office having only one public IP address deployed with IP Phones, all SIP messages sent to IP PBX (or Call Manager) will go through NAT and PAT being applied. This is same as for any data packet. VoIP packets are not different here.
For example, IP address assigned to Phone is 192.168.1.1 (and SIP Port 5060) and enterprise public IP is 1.1.1.1. When SIP REGISTER message passes through NAT router and reaches IP PBX, IP PBX will see the source IP as 1.1.1.1 and source port as <any port PATted by enterprise router>. While inside the SIP REGISTER message, you will still see IP and port as 192.168.1.1 and 5060.
Standard SIP based IP PBX's are smart enough to detect that UA is behind the NAT and ignore the IP headers used inside the SIP message and capture real network IP for further communication. In result, IP PBX will store UA address as 1.1.1.1 (and patted port) instead of 192.168.1.1:5060 in location service. Now when IP PBX needs to place call to that UA, it will send the call to 1.1.1.1 (and patted port). Since router will already be having NAT/PAT binding, it will route the packet (INVITE) back to the phone.
Same is applicable for SDP as well. IP PBX ignores the SDP IP address and use the real (natted) IP address for RTP setup. But in that case, direct RTP between the two phones won't be possible (unless other mechanism like TURN, ICE etc are not used) and IP PBX will have to relay the RTP with itself.
Since SIP by default works on UDP and UDP NAT bindings are likely to expire in 600 sec (or less) whereas default SIP registration timer is 3600 sec, IP PBX intelligently returns shorter duration in 200 OK (mostly 120 to 300 sec) for registration so that phone will keep sending REGISTER message which results NAT binding being refreshed.
2. SIP Trunking
Most of the SIP trunking provider takes care of those clients which are located behind the NAT. When you take services from provider like Vonage etc, they provide you small piece of adapter (ATA) which is installed behind the NAT in most of the cases. Here, service provider network takes care of SIP messages in the same way what IP PBX was doing in the previous example.
- Vivek
