Voicefinder AP602 SIP paging terminal provides high-performance SIP signaling based VoIP paging service for bank, securities, military, large enterprise, and government. AP602 SIP paging terminal should be used together with SIP paging server for dedicated IP paging service, or can be connected to legacy SIP call manager like as IP phones for SIP paging service. Generally, SIP call manger supports the SIP paging service for simple paging service via IP phones. Using SIP VoIP signaling and internal 40watt digital amplifier, AP602 performs SIP paging service for overall room paging announcement. At front side, this device provides UP, DOWN volume button for easy speaker volume control. Also, AP602 SIP paging terminal supports internal 40watt digital AMP. to connect external speaker for various VoIP based paging service environment. AP602 SIP paging terminal provides excellent performance and satisfaction for integrated security management environment like as intrusion detection application (government oil station, military, national power plant, etc). Installation and management is very simple. Beginners can handle the device easily.

DSP Hardware Type High Performance SIP Paging Terminal

AddPac SIP paging service solution consists of SIP paging server, SIP paging management system and AP602, AP1605 SIP paging terminals. Designed on the foundation of high performance embedded RISC CPU + Voice DSP, AP602 SIP paging terminal supports two(2) 10/100Mbps fast ethernet interface, one(1) port RS232C console port. Providing built-in 40watt internal digital amplifier, AP602 SIP paging terminal is designed to support VoIP (Voice over IP) paging service without external amplifier via directly connected to SPEKER. And, this paging terminal supports high quality 16KHz G722 voice codec beside traditional 8KHz G.711, G.726, G729A, G.7231.1,etc.

Real Time Live VoIP Paging Service

AddPac SIP based paging solution opens a whole new world of high quality real time audio/voice broadcasting service based on the high performance and stability RISC + DSP embedded hardware. DSP based voice compression technology and real-time transmission network technology like as RTP, RTSP protocol supports stable VoIP paging service under general data combination network through QoS algorithm. Especially, a user may setup an appropriate bit rate in accordance with network status so that it would accept from low bandwidth to high bandwidth.

High Quality SIP Paging and Supplementary Service

A various IP based VoIP paging solution is widely recognized throughout the world in terms of its stability and performance. AP602 SIP paging terminal will satisfy the various needs of customer demand for IP based audio/voice paging service. It supports stable voice paging service in internet network with AddPac QoS (Quality of Service) algorithm

Network Diagram

High Performance RISC + DSP Processor Architectures

Support various VoIP codec: G.722(16KHz), G.711, G.729A, G723.1, etc

Internal built-in digital AMP. (Option)

Multiple point-to-point IP voice broadcasting (unicasting scheme)

IGMP based IP multicasting routing

2-port 10/100Mbps fast ethernet interface (RJ45)

Up, Down Volume Control Button

On-AIR LAMP LED Support

3.5mm SPK, MIC Interface Support

I/O Interface Support : Alarm 1-Port, Relay 1-Port, I2C Interface

1-port RS232C Interface(RJ45) for Command Line Interface

High-performance IP-Routing Capability with Reliability

Static and IEEE 802.1Q VLAN Routing Protocols

Traffic Queuing, and SNMP MIB v2 for Network Management Features

Standard & Extended Access List for Security Functions

Remote Software Upgrade using FTP & TFTP

SIP protocol support compliant with IETF RFC3261 (or RFC2543)

External Power Supply

AddPac Technology