Security Concerns with VOIP

IP telephony was first introduced in the mid 1990's and has improved steadily in both reliability and sound quality. These improvements have gone hand in hand with increased network bandwidths and improvements in compression technology, allowing IP Telephony to become a viable technology. Current voice traffic on circuit switched networks has a very high level of quality because each connection is guaranteed a certain bandwidth (64kbps) for the life of the call. When voice traffic is transmitted over an IP based network the data is compressed down to 7.9 kbps or 6.3 kbps. This saving in bandwidth comes at a price in the quality level of the call. On an IP based network, packets can travel over any number of different routes so the quality of the transmission is tied to the quality of the network. Lost packets in a VOIP network degrade the quality of the system by appearing as gaps of silence in the conversation. As with any developing technology there are various standards being proposed as the best way to achieve industry acceptance. This paper will briefly discuss the following three standards dealing with IP Telephony implementations: H.323 from the ITU, Session Initiation Protocol (SIP) from the IETF, and MGCP by the Media Control Working Group.