|Installation Planning / Hardware Infrastructure|
Vocera SIP Telephony Gateway is based on Internet Engineering Task Force (IETF) standards for Session Initiation Protocol (SIP) 2.0 and Real Time Transport Protocol (RTP).
Vocera SIP Telephony Gateway communicates via a SIP trunk with a SIP-enabled PBX or a SIP Gateway and provides basic SIP telephony functionality, including placing and receiving calls, OPTIONS keep-alive messages, and obtaining ANI and DNIS information. The Vocera SIP Telephony Gateway is interoperable with SIP-enabled PBXs and SIP Gateways as long as they follow SIP 2.0 and RTP standards.
For audio transport, Vocera SIP Telephony Gateway uses Real-time Transport Protocol (RTP), an Internet protocol standard for delivering multimedia data over unicast or multicast network services. For more information refer to RFC 3550 at http://tools.ietf.org/html/rfc3550 and RFC 35515 at http://tools.ietf.org/html/rfc3551.
Vocera SIP Telephony Gateway uses Vocera proprietary signaling and transport protocols for all communication between the server and Vocera badges. Consequently, Vocera SIP Telephony Gateway converts from SIP and RTP protocols to Vocera protocols, and vice versa, to enable communication between the Vocera SIP Telephony Gateway and the IP PBX.