VoIP-NAT Problem

Most residential and SOHO users have NAT based routers/firewalls that make the rollout of an IP based network of IP phones or residential phone adapters a challenge. The fact that VoIP protocols encapsulate original device’s IP address in the session layer packet and that firewalls won’t allow any session originated from the public Internet to traverse to internal-protected network bring all sort of problems like no audio or one way audio to VoIP implementations. Some attempts to resolve this issue such as STUN or reconfiguring end-users routers or firewalls not always work and become a support nightmare when facing different types of NAT.

ITSPs or VoIP service providers need to add a VoIP Session Border Controller and RTP bridge server to their networks in order to support IP devices behind NAT but most of the cases it relays media stream adding unnecessary latency and delays affecting end-users audio quality perception. Having to relay media stream for hundreds or thousands of connections certainly compromises the ITSPs own bandwidth and requires much more processing power from the servers.