Unify OpenStage 15 SIP Black IP phone

OpenStage 15 SIP, Fast Ethernet, PoE, lava The phone OpenStage 15 is part of the OpenStage phone family. It can be used with SIP or HFA software.

Supported Codecs:
- G.711
- G.729AB
- G.722 Wideband (7kHz)
Standard based SIP support according to RFC 3261 (with SIP software)
Full-duplex hands-free talking with high quality housing microphone and loudspeaker
Computer Telephony Integration (CTI for HFA), Third party call control (for SIP)

Connections
- 2 ports 10/100Base-T built-in Ethernet switch
- Power over Ethernet (PoE) according to IEEE 802.3af or external power supply (EU, US and UK power adapters)
- Sidecar module support for OpenStage Key Module 15

- 3 navigation keys
- 8 free programmable keys (with paper labels)
- 3 fixed function keys for speaker, messages and menu with red LEDs
- Volume keys (Loudspeaker/+/-)
- Keypad with 12 keys
- not tiltable LCD 2 line display
- Various housing colors (ice blue, lava)
- Wall mountable

SIP = Session Initiation Protocol
In telecommunication, "telephony" encompasses the general use of equipment to provide voice communication over distances. Traditional telephone systems were originally based on the analogue system of physical connection via fixed cables and transmission via electrical pulses. Both digital systems and new connectivity media (fibre, satellite) extended the scope and reliability of telephony systems, such that very few analogue systems are still in use.

Internet Protocol (IP) telephony differs from digital telephony in that it transmits digitised voice data via networks originally built for data transmission, not for voice transmission. At its basic level, IP is a set of rules that allows computer networks to communicate with other, with IP telephony building on these rules. IP is based on internet protocols, not telephony protocols. This means that IP networks use the same rules that govern the World Wide Web, not those that govern standard circuit-switched PABX telephony.

SIP is an extension of the IP protocol. Essentially, SIP is the future standard for communications of all kinds via a data network and its appeal lies largely in its simplicity: SIP sets up, handles and ends "sessions" over IP networks. A session could be a simple two-way telephone call or it could be a collaborative multi-media conference session. The ability to establish these sessions means that a host of innovative services become possible, such as voice-enriched e-commerce, web page click-to-dial and Instant Messaging.

Over the last couple of years, the Voice over IP community has adopted SIP as its protocol of choice for signalling and SIP is now a specification of the International Engineering Task Force (IETF). Until the IETF's SIP standard was embraced, VoIP phone systems were in danger of proprietary vendor lock-in. Gradually, SIP is evolving from the prestigious protocols it resembles -- the Web's Hyper Text Transfer Protocol (HTTP) formatting protocol and the Simple Mail Transfer Protocol (SMTP) email protocol -- into a powerful emerging standard.

Like the Internet, SIP is easy to understand, extend and implement. In telephony this means that all communication is managed by the network, from a simple phone call through to voice-enriched eCommerce, web page click-to-dial and instant messaging. SIP users may locate and contact one another-regardless of media content or number of participants. SIP extends the open-standards spirit of the Internet to messaging, enabling disparate computers, phones, televisions and software to communicate with each other.