Solving the firewall and NAT traversal issues for SIP-based VoIP

Voice over the Internet Protocol (VoIP) - telephony over the Internet - is gaining popularity. VoIP is an evolving technology that still has several problems. Some of them are caused by network topology, such as the deployment of firewalls and network address translation (NAT). This is referred to as the ldquoNAT and firewall problemrdquo. This paper discusses the problems of SIP-based VoIP. SIP is the most relevant signaling protocol for VoIP today. Unfortunately, a VoIP call cannot be established if one of the SIP softphones is situated behind a NAT gateway or behind a restrictive firewall. A VoIP call involves three random UDP ports for replies that must be allowed. This can be problematic due to security reasons. In this paper, we present a software suite that solves the aforementioned problems. This ldquoSIP-RTP-Proxyrdquo (SRP) works as a ldquosession border controllerrdquo (SBC) by accepting and relaying SIP signaling data and RTP / RTCP media data. A special feature of this solution is the possibility to place the SRP inside a private network, while former solutions such as SBCs have to be installed directly on the NAT gateway or in the public Internet. One goal of this project was to develop a universal solution that solves the NAT problem for VoIP in small business and home environments. It can be used in combination with various operating systems and does not depend on modified clients or a modified ldquoInternet Protocol Private Branch Exchangeldquo (IP PBX).