AddPac Technology's Official Blog.: VoIP Total Recording Solution (Radio System, Analog FXO PSTN, 3.5mm Audio Output Record, VoIP Packet Recording, AddPac IP Telephony Solution Recording,etc))

1.AP-SoftNR-Voice Software (IP Voice Recording Solution, Linux Server, Windows Application) Features
User registration and access control management (User Management)
Voice recording server status monitoring (Recording Server Status Monitoring)
Recording file management (Recording File Management)
Recording file list storage
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 Software Provides 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 Analyzer The 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