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.
SIP Paging
Solution Table
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.
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.
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.
Network Diagram
