HTTPS enhanced
security protocol function for Enterprise VoIP Gateway Solution like as AP2620,
AP2120N, AP2640 and AP2650.
AP2650 is a High Performance VoIP
gateway supports maximum 32ports analog voice interface and 2ports digital
E1/T1 voice modules. AP2650 supports dual power supply as redundancy. Analog
and digital interface of AP2650 provide an optimized call scenario when it
interoperates with analog PBX.
AP2650 VoIP gateway is modular type
device and supports four(4) VoIP card slots. FXS, FXO, E&M, E1 and T1 interface
module can be used in AP2650.
AP2640 is a High Performance VoIP
gateway supports maximum 32ports analog voice interface and 2ports digital
E1/T1 voice modules. AP2640 VoIP gateway is modular type device and supports
four(4) VoIP card slots. FXS, FXO, E&M, E1 and T1 interface module can be
used in AP2640.
AP2120N is an enterprise VoIP gateway supports maximum 16ports analog voice
interface voice modules. AP2102N VoIP gateway is modular type device and supports
two(2) VoIP card slots. FXS, FXO, E&M interface module can be used in
AP2120N.
AP2620 is a High Performance VoIP
gateway supports maximum 8ports analog voice interface and 2ports digital E1/T1
voice modules. AP2620 VoIP gateway is modular type device and supports two(2)
VoIP card slots. 4 port FXS, 4 port FXO, 4port E&M, E1 and T1 interface
module can be used in AP2620.
AddPac enterprise VoIP gateways
support various enhanced security protocols like as HTTPS beside legacy
protocol like as HTTP, Telnet, FTP. It
supports HTTPS security protocol, secure shell like SSH, SFTP protocol, and TCP
MD5 and password security.
User
can use new enhanced security protocols like as HTTPS through a firmware
upgrade.
The new and versatile AddPac IP
phone bring the integrated solution for the IP based voice communication and
the broadcasting feature to maximize business potentials. It provides feature
keys, customizable hot-keys, two(2) ethernet ports, QoS function, public IP
sharing. It supports not only the major VoIP signaling protocols such as SIP,
H.323 but also G.722, G.711, G.726 voice codec, stereo audio in/out interfaces
for external Headset MIC, etc. AP-IP120 series provides 12 speed-dial key with
user presence LED and 4 line text graphic LCD.
AP-IP120 Demo. Youtube :
Demonstration Scenario
1. Show the direct LAN connection between the AP-IP120 LAN1 port and a PC.
2. Access the default IP address
192.168.10.1 via Telnet to configure the LAN0 IP address.
3. Log in to Smart Web Manager to
set up 12 speed dial key.
4. Make a call using the configured
speed dial number (the recipient's phone is AP-IP90 IP Phone).
5. Log in to Window based Smart
Messenger Software and make a call using Smart Messenger.
6. Receive a call using Smart
Messenger.
AP-IP120 Smart Web Manager : 12 speed-dial button configuration
Smart Messenger Software
As Smart Messenger interoperates with AddPac’s Customer Premise Equipment (CPE) products, which are based on its Next Generation Multi-Media Telephony Solution, it can provide all different kinds of services such as Messenger Service , Telephone Directory, User’s presence, Unified Massage (Voice Mail, Short Message), Call Control and Forward Setup functions.
Smart Messenger can be operated by AddPac Technology’s proprietary protocol called Smart Service Control Protocol (SSCP), which is managed between AddPac’s IP PBX and IP CPE’s in an environment of Microsoft Window based PC Platform. In order to provide User Presence features on real-time basis, such as user busy, on-line, user away, etc, the Presence server can be mounted and operated on IP-PBX. The Presence Server can be operated on the same hardware platform and can be operated independently in the platforms separated to each other.
Smart Messenger Protocol Structure
AddPac Technology’s Smart Messenger executes Messenger function via IPC (inter-process communication) or network communication between Presence Server and Call Manager. Through SSCP session, Smart Messenger is connected to Call manager and Presence Server and consists of Messenger Window and Call Control Window in a broad sense.
Smart Messenger Diagram
Smart Messenger System Diagram
Smart Messenger Login
AddPac Smart Messenger’s Log-in Display can be categorized in 3 parts largely. Set-up part enables IP address and port number of Presence Server to fixed and it also provides Automatic Log-in and Automatic Input features.
Smart Messenger Login
Smart Messenger Structure Main Display
Main Display of Smart Messenger consists of 4 major parts. As you can see from the following diagram, Part A provides information on a configuration of internal and external telephone directory of an organization. External telephone directory includes a list of customer directory. Internal and external telephone directory also provides information on public and private telephone directory. Configuration of private telephone means a telephone directory which can be edited by user. The telephone icon at the end of the left in the middle provide Call History for outgoing and incoming telephone numbers.
Smart Messenger Main Display
Part B of Main Display provides indication of a set up status and shows control set ups, Message Box On/Off, User Search function and button all together Part C provides list display function for Inbox List Display. Voice Mail List and Short Message List can also be exhibited. Part D provides the supplementary functions of IP Phone and Video Phone. It also provides a dial pad and soft key features.
AddPac
Gigabit Ethernet IP Phone Solution is designed to provide optimal IP telephony
services to meet the needs of various enterprise customers. In AddPac Gigabit Ethernet
IP phone solution, there are AP-IP90G, AP-IP120G, AP-IP160G, AP-IP230G and
AP-IP300G. This AddPac Gigabit Ethernet IP Phone solution supports high quality
voice in general internet environment as well as in local LAN environment
because it supports various VoIP voice codec like G.722, G.711, G.726, G.729ab and
QoS function according to internet bandwidth environment.
AddPac
Gigabit Ethernet IP phone supports not only VoIP call function but also IP voice
broadcasting function. These gigabit Ethernet IP phones provide two Gigabit
Ethernet ports, audio port for external headset, various function keys, and
Power over Ethernet function.
In
addition, AddPac Gigabit Ethernet IP phone solution supports various call
service features like music on hold, coloring service, and call transfer
feature by interworking with various IP-PBX, soft switch and call manager as
well as AddPac IP-PBX.
Demonstration
AP-IP90G IP Phone Bridge
Mode LAN-LAN Full Throughput Demo. Youtube :
Network Diagram
Demo. Scenario:
1.Connecting AP-IP90G to
Laptop:
Show
the connection of LAN1 on the AP-IP90G to the Laptop's Ethernet port.
2. Displaying Gigabit
Connection:
Demonstrate
on the laptop that the connection is recognized as Gigabit.
3.Showing Internet
Connectivity:
Display
the laptop accessing the internet to confirm connectivity.
4. FTP Server File
Transfer:
Use
the AP-PNC2000 Server as the FTP Server.
Upload
and download files while showcasing the transfer speeds.
AddPac
Technology's VoIP packet voice recording solution is designed to record voice
over AddPac or Third-party IP telephony solution networks. It analyzes VoIP packets from
third-party IP switches (using port mirroring) or from LAN switches connected
to IP switches, providing high-performance network-based digital voice
recording and processing functions.
For example, the AP-SoftNR, an IP-based network voice recording
server, captures VoIP voice packets and analyzes them, delivering RTP voice
packets and signaling information to the application software. Instead of the
AP-SoftNR voice recording server capturing, analyzing, and storing VoIP packets
directly, an additional capture device in the form of a voice packet probe can
be used, allowing remote operation of the IP voice recording server. Because
this configuration is IP-based, it supports system scalability and redundancy
smoothly.
The AddPac VoIP packet recording server operates on the
AP-Server3 and AP-Server2 Linux hardware servers. It supports the following
software features:
lUser registration and access control management (User
Management)
lVoice recording server status monitoring (Recording Server
Status Monitoring)
lLive call recording list and monitoring (Live Call Recording
List and Monitoring)
lEvent management (Event Management), including event monitoring,
event configuration, and system monitoring
STT (Speech to Text) functionality (optional)
AP-Server3
Key Features
Network Interface: 4 x 10/100/1000 Mbps Gigabit Ethernet ports
Modular redundant power supply
Nvidia H/W Engine
AP-Server2
Key Features
Network Interface: 4 x 10/100/1000 Mbps Gigabit Ethernet ports
Modular redundant power supply
Nvidia H/W Engine
Demonstration
VoIP Packet based IP Voice Recording Demo Youtube :
Demo 1 Scenario :
The
IP exchange used is the IPNext180 IP-PBX, and the AP-SoftNR hardware operates
on the AP-Server3. The IP phones installed are the AP-IP300 and AP-IP120,
demonstrating the call recording functionality. The recorded files are played
back using the Smart DvoiceR Manager, confirming that the recordings are
functioning correctly.
Demo1 Network Diagram
(example, Direct Connect)
2. Demo2 Scenario :
The
IP exchange used is the IPNext180 IP-PBX, and the AP-SoftNR hardware operates
on the AP-Server3. Under demo2 scenario, instead
of the AP-SoftNR voice recording server capturing, analyzing, and storing VoIP
packets directly, using additional VoIP capture devices like as AP-PNC2000L, we
demonstrate a configuration that transmits captured VoIP packets to the remote
AP-SoftNR voice recording system. The IP phones installed are the
AP-IP300 and AP-IP120, demonstrating the call recording functionality. The
recorded files are played back using the Smart DvoiceR Manager, confirming that
the recordings are functioning correctly.
Demo2. Network
Diagram (example, VoIP Packet Capture Probe, VoIP Recording Server at Remote
Site)
Live call recording list and monitoring (Live Call Recording List and Monitoring)
Event management (Event Management), event monitoring, event configuration, system monitoring
STT (Speech to Text) functionality (optional)
Real-time STT
Background STT
Keyword search and automatic alarm/notification functions
2.Next-Generation VoIP Voice Recording Server Linux Server Solution
Introduction to IP voice recording server on YouTube:
AP-ITMS3000 High Performance Linux Server
Key Features
Network Interface: 4 x 10/100/1000 Mbps Gigabit Ethernet ports
Modular redundant power supply
Nvidia H/W Engine
AP-ITMS2000 High Performance Linux Server
Key Features
Network Interface: 4 x 10/100/1000 Mbps Gigabit Ethernet ports
Modular redundant power supply
Nvidia H/W Engine
3. Windows Application Voice Recording SoftwareProvides software based on MS Windows.
AddPac IP voice call recording storage server delivers a high-quality real-time voice call storage solution with excellent performance and stability, which was difficult to achieve with previous solutions. Voice data captured from End-Points (IP video phones, etc.) is transmitted over the IP network to the call recording storage server. The stored voice recording data can be played back through speakers or headsets using AddPac's dedicated MS Windows-based voice recording application.AP-SVRM IP Voice Recording Server Management Software
AP-SVRM (Smart Voice Recording Management Software) is the management software for voice recording server devices in AddPac Technology’s next-generation IP-based voice recording solution.
AP-SVRM is implemented as a server/client model running on an MS Windows-based PC platform. Voice recording storage devices like AP-SoftNR-Voice serve as the server equipment for AP-SVRM client software.
The software supports user registration and access control management, voice recording server status management, recording file management, recording file waveform analysis, live call list and monitoring, event management, and recording board management.User Registration and Access Control Management (User Management)The smart voice recording management software is designed so that only registered administrators can log in. User registration information includes username, user ID, and user password. Through the settings menu on the login screen, the IP address and port number of the voice recording server can be configured, and an automatic login feature is provided via password saving.
Voice Recording Server Status Monitoring
The voice recording server status management function provides settings and displays a list of connected clients. Settings include configuring the maximum number of client sessions and the Keep Alive time. The client session view shows the list of clients currently connected to the voice recording storage server.
Recording File Management
This feature provides management of recorded voice files. It offers search filters and engines to easily display a list of recorded files for the desired time period. It supports playback, deletion, and Excel report functions. Clicking on a searched file provides voice recording information, call records, destination number, and caller number details. The media player supports Play, Seek, Pause, Resume, and Stop functions.
Recording File List Saving
The voice recording file management function displays a list of searched recorded voice files and supports Excel export. It provides a report feature that allows viewing the searched recorded voice file list as an MS Excel file.
Recording File Waveform AnalyzerThe voice recording waveform analyzer displays graphical voice waveform for incoming and outgoing signals. It is used when detailed waveform analysis of a specific call is needed. It offers various functions such as segment repeat playback, time domain zoom in/out, amplitude domain zoom in/out, and bookmark features.
Live Call Recording List and Monitoring
This function shows a list of currently recording calls. Clicking on the information provides detailed data about the incoming and outgoing signals. This is a software function block.
Event Management
Supports event history management for the voice recording server. The event setting function allows configuration of the event source server address and port number where events occur. Users can set the event level and event logging level. Event categories are divided into recording, play, and system. It also provides settings for emergency alarm sound activation.
Event Configuration
Event monitoring
System Monitoring
4. Radio Communication Audio Output IP Recording Gateway Solution
4.1 AP-SVG3000 8-Port High-Quality G.722 STT (Speech to Text) Voice Recognition VoIP Gateway
AP-SVG3000 Introduction Youtube :
AP-SVG3000 Demo. Youtube : (16bit Linear PCM STT VoIP Gateway for AI Voice Recognition)
The VoiceFinder AP-SVG3000 is a next-generation voice recognition high-quality 16KHz G.722 VoIP gateway aimed at NGN (Next Generation Network), supporting up to 8-port STT (Speech to Text) VoIP interface voice calls. It features independent two analog VoIP modules that can be installed in four VoIP slots, offering excellent specifications in terms of scalability and environmental adaptability. The analog voice recognition VoIP module supports a 1-port audio input interface, radio interface, 10/100Mbps high-speed Ethernet, and RS232C console interface.
The rapid expansion of broadband wireless services, including 5G wireless networks, is driving increasing demand in the voice recognition field, which is a core technology alongside cloud computing, robotics, AI, and big data technologies. The high-quality remote voice recognition VoIP gateway is the first step in integrating into an IP network, so careful consideration is necessary before implementation. If you are looking for a product with high scalability and proven functionality at an affordable price, pay attention to the AP-SVG3000. The AP-SVG3000 offers a solid opportunity to start small while thinking big as a high-quality voice recognition VoIP gateway.
Network Diagram
4.2 AP-SVG2000 4-Port High-Quality G.722 STT (Speech to Text) Voice Recognition VoIP Gateway
The VoiceFinder AP-SVG2000 is a next-generation voice recognition high-quality 16KHz G.722 VoIP gateway aimed at NGN (Next Generation Network), supporting up to 4-port STT (Speech to Text) VoIP interface voice calls. It features independent two analog VoIP modules that can be installed in two VoIP slots, offering excellent specifications in terms of scalability and environmental adaptability. The analog voice recognition VoIP module supports a 1-port audio input interface, radio interface, 10/100Mbps high-speed Ethernet, and RS232C console interface.
The rapid expansion of broadband wireless services, including 5G wireless networks, is driving increasing demand in the voice recognition field, which is a core technology alongside cloud computing, robotics, AI, and big data technologies. The high-quality remote voice recognition VoIP gateway is the first step in integrating into an IP network, so careful consideration is necessary before implementation. If you are looking for a product with high scalability and proven functionality at an affordable price, pay attention to the AP-SVG2000. The AP-SVG2000 offers a solid opportunity to start small while thinking big as a high-quality voice recognition VoIP gateway.
Network Diagram
5. Analog PSTN FXO 16KHz VoIP Codec STT Gateway Solution
The VoiceFinder AP-STTFXO2 VoIP module is a next-generation analog FXO 16KHz G.722 VoIP gateway module aimed at NGN (Next Generation Network), supporting 2-port STT (Speech to Text) VoIP interface voice calls. The AP-STT FXO2 VoIP module supports two independent analog VoIP interfaces, with each analog voice recognition VoIP module offering a 1-port audio input interface, a 1-port FXO interface (RJ11), 10/100Mbps high-speed Ethernet (RJ45), and an RS232C console interface (RJ45).
The AP-STTFXO2 2-port G.722 voice recognition FXO VoIP module can be installed in the AP-SVG3000 (which supports 4 module slots) and the AP-SVG2000 (which supports 2 module slots), providing excellent specifications in terms of scalability and environmental adaptability.
Satisfaction of Both Economy and Scalability
The rapid expansion of broadband wireless services, including 5G wireless networks, is driving increasing demand in the voice recognition field, which is a core technology alongside cloud computing, chatbots, robotics, AI, and big data technologies. The high-quality remote voice recognition VoIP gateway is the first step in integrating into an IP network, so careful consideration is necessary before implementation. If you are looking for a product with high scalability and proven functionality at an affordable price, pay attention to the AP-STTFXO2. The AP-STTFXO2 offers a solid opportunity to start small while thinking big as a high-quality voice recognition VoIP gateway.
Configurable for Up to 8-Port STT VoIP Channels
The AP-SVG3000 is an STT (Speech to Text) voice recognition VoIP gateway that can be configured for up to 8 ports of analog VoIP voice channels. Initially, you can use the AP-STTFXO2 2-channel STT VoIP module, and when the need arises to increase the number of channels, it can flexibly accommodate up to 8 channels. The AP-SVG3000 will be a great choice for customers seeking high-performance, high-quality voice recognition VoIP gateways. The STT VoIP modules used in the AP-SVG3000 operate independently per port and are designed to support various VoIP audio codecs, including 16KHz G.722 wideband voice codec, as well as 8KHz G.711, G726, G.729A, and G.723.1. The voice interface supports an analog trunk FXO interface (RJ45) in addition to audio input/output interfaces. It also supports an RS232C (RJ45) interface for CLI (Command Line Interface).
High Satisfaction in Call Quality and Value-Added Services
Adpak's various VoIP gateway series are recognized for their performance and reliability in both the Korean and global markets. The STT voice recognition module, which incorporates Adpak’s experience and know-how gained in the enterprise and telecommunications markets, will provide high satisfaction in various service aspects to meet the demands of customers seeking high-quality VoIP services. Firstly, it concurrently supports VoIP control protocols such as H323 and SIP VoIP signaling. Additionally, to ensure excellent call quality in any network environment, Adpak has enhanced the standard QoS (Quality of Service) algorithms with its proprietary QoS algorithms.
Network Diagram
6. 3.5mm audio Input IP voice recording solution
The VoiceFinder AP601 3.5mm audio input terminal is designed to work with AddPac's IP-based voice recording server, providing voice recording service functionality over the internet using VoIP SIP signaling protocols. It is installed in financial institutions, corporations, government offices, military facilities, and educational networks to serve remote branches, locations, and sites.
The front panel features volume up and down switches, allowing users to adjust the desired volume according to their preferences. An audio input interface (RCA) for legacy voice recording sources is supported, and if needed, an RCA to 3.5mm converter can be used.
The AP601 audio input gateway terminal provides an excellent solution for supporting AddPac's IP voice recording solution over local area networks, designed to ensure maximum satisfaction for users with its outstanding performance. Its installation and operational maintenance are very simple, making it easy for beginners to operate.
High-Performance SIP Broadcasting Terminal
The AP601 audio input gateway terminal is designed to accommodate PSTN voice codecs such as G.722 (16KHz), G.711, and G.726. It combines high-performance DSP hardware developed directly by AddPac with a high-performance processor to encode analog voice signals into VoIP voice signals in real time and transmit them to the IP voice recording server.
Network Diagram
7. VoIP Packet (Sniffing Method) Recording Solution (Recording for Third-Party IP Switches, IP Phones, and VoIP Gateways)
AddPac Technology's VoIP packet voice recording solution is designed to record voice over third-party IP telephony solution networks. It analyzes VoIP packets from third-party IP switches (using port mirroring) or from LAN switches connected to IP switches, providing high-performance network-based digital voice recording and processing functions.
For example, the AP-SoftNR, an IP-based network voice recording server, captures VoIP voice packets and analyzes them, delivering RTP voice packets and signaling information to the application software. Instead of the AP-SoftNR voice recording server capturing, analyzing, and storing VoIP packets directly, an additional capture device in the form of a probe can be used, allowing remote operation of the IP voice recording server. Because this configuration is IP-based, it supports system scalability and redundancy smoothly.
IP Port Mirroring (Example)
AddPac IP-PBX (Figure A in the above diagram) supports ERSPAN port mirroring functionality for all traffic (including SIP signaling packets, RTP packets, etc.) entering the switch through the Primary LAN port, via the second LAN port, for application servers such as VoIP recording servers.
Three Methods of Port Mirroring:
1.SPAN: sniffer is at the same switch
2.RSPAN: sniffer is at different switch
3.ERSPAN: sniffer is across IP network
ERSPAN Packet Format
Field
Description
C
Checksum bit. Set to 1 if a checksum is present.
K
Key bit. Set to 1 if a key is present.
S
Sequence number bit. Set to 1 if a sequence number is present.
Reserved 0
Reserved bits; set to 0.
Version
GRE Version number; set to 0.
Protocol Type
Indicates the ether protocol type of the encapsulated payload.
Checksum
Present if the C bit is set; contains the checksum for the GRE header and payload.
Reserved 1
Present if the C bit is set; is set to 0.
Key
Present if the K bit is set; contains an application-specific key value.
Sequence Number
Present if the S bit is set; contains a sequence number for the GRE packet.
AddPac IP-PBX ERSPAN Port Mirroring CLI (Command Line Interface)
VoIP Packet based Voice Recording Demo Youtube :
Network Diagram (example, Direct Connect)
Network Diagram (example, VoIP Packet Capture Probe, VoIP Recording Server at Remote Site)
Network Diagram (example, VoIP Packet Capture Probe, Dual Redundant VoIP Recording Server at Remote Site)
8. AddPac IP-PBX, IP Phone and VoIP Gateway Voice Recording Solution
AddPac Technology's VoIP product line voice recording solution is configured with AddPac's IP-based voice recording equipment, along with AddPac SIP Call Manager (IP PBX), VoIP gateways, and IP phones, providing next-generation high-performance network-based digital voice recording and processing capabilities. For example, the Adpak IP PBX (SIP Call Manager) and IP phones send the captured IP voice packets to an IP-based network voice recording server such as AP-SoftNR. Since it is IP-based, the system also supports scalability and redundancy smoothly.
IP Phone Solution with Voice Recording Service Features
VoIP Gateway Solution with Voice Recording Service Features
AddPac's network-based voice recording solution is not just a simple voice recording solution at a single access point; it is a next-generation voice recording solution that supports a total voice recording service architecture on a broadband IP integrated network, working in conjunction with various IP-based VoIP devices such as IP phones, IP PBXs, and VoIP gateways.
AddPac's IP PBXs, VoIP gateways, and IP phones are advanced devices that combine high-performance CPU modules and hardware to provide a network-based voice recording server solution. The IP-based voice recording server equipment maximizes performance and efficiency using high-performance processors that support gigabit Ethernet and a stable Linux OS. The IP voice recording server stores and manages compressed voice signals transmitted from the IP PBX and performs the function of sending them to a media player for playback.
The IP Centrax voice recording solution uses high-performance processors to provide a real-time high-quality voice recording solution that was difficult to achieve with previous solutions, based on excellent performance and stability. The stored voice recording data can be listened to through speakers or headsets using Adpak's Windows-based dedicated voice recording program.
Network Diagram
Network Diagram (Satellite)
9. PTSN Analog/Digital Voice Recording Solution
VoIP Gateway Solution with Voice Recording Service Features
