Session Initiation Protocol (SIP) signaling is a key protocol for connecting IP networks together. It is also used to connect enterprise locations with data networks for voice calling. Signaling systems used in office phone systems need to be compatible with signals used in a carrier’s network if voice calls are to be sent over data links. This is so that the equipment can interpret addressing signals, ringing signals, and busy signals sent by office IP PBXs, and vice versa. Without this compatibility, organizations with IP telephone systems need to support two different sets of trunks, one set for voice and another set for data. Or they can use a gateway to translate their IP non-SIP calls to SIP. ( Trunks are used to carry voice and data on public networks.) Without SIP there are extra costs and long-distance fees for gateways, and voice quality is impaired. The signal’s quality is impaired when converting VoIP to formats compatible with SIP data networks when they are sent, and converting them back to VoIP when they are received. Impairment results because compression does not re-create voice in exactly the same format when it’s decompressed, and vice versa. Thus, its quality is degraded every time voice data is converted. Because not all IP telephone systems are SIP compatible, they are not all able to share data links for voice traffic without using gateways. In addition, the telephone company that carries the traffic must use SIP-compatible signaling, as well. Most providers now offer SIP-compatible trunks.