AddPac Technology’s SIP voice paging solution consists of SIP paging server, SIP paging terminal, SIP phone, Speed-Dial extension pack, SPMS(SIP Paging Management Software) SIP paging management software, and SPCS(SIP client paging software) SIP paging client software.
SIP paging solution is an internet access ready IP broadcasting system to support SIP based VoIP broadcasting transmission by using Standard VoIP protocol in finance, enterprise, and public office.
SIP paging solution supports SIP (Session Initiation Protocol) protocol and RTP (Real-time Transmission Protocol) VoIP standard protocols for IP based voice transmission service. Also, this system supports standalone mode and general IP-PBX clone mode for SIP paging service. Also, SIP paging solution supports excellent voice and audio broadcasting through internet in enterprise environment to satisfy the needs of customer demand.
AddPac compact size SIP paging terminal solution consists of AP601, AP602 terminals. These paging terminals provide high performance and stability based on embedded system architecture for IP public announcement application.
SIP Paging Solution Table
AP601 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, AP601 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, AP601 SIP paging terminal supports internal 40watt digital AMP. to connect external speaker for various VoIP based paging service environment. Designed on the foundation of high performance embedded RISC CPU + Voice DSP, AP601 SIP paging terminal supports one(1) 10/100Mbps fast ethernet interface, one(1) port RS232C console port. Providing built-in 40watt internal digital amplifier, AP601 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, G729ab, G.7231.1,etc.
AP601 SIP based paging terminal 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.
Designed on the foundation of high performance embedded RISC CPU + Voice dedicate 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 G.722 voice codec beside traditional 8KHz G.711, G.726, G729a, G.723.1, etc. At rear side, this device supports I2C interface and alarm Input 1-Port, relay output 1-port for various application services.