AP-EIP70 SIP비상통화장치 , AP-IP300 SIP 관제 전화기 사이에 PBX 없는 구성으로 데모 유투브 : 데모 내용 : AP-IP300 SIP 관제전화기에서 AP-EIP70 비상통화장치로부터의 VoIP 콜 착신시에 미리 설정된 해당 단축버튼에 램프가 깜박거리고, LCD 에는 해당 사이트 이름 표시
1. AP-EIP70 SIP 비상통화장치 (엘리베이터, 장애인화장실 외)
AP-EIP70 데모 유투브 :
AP-EIP70 IP 비상 호출 전화기는 인터넷을 이용해 고음질의 음성통신이 가능한 SIP VoIP 시그널링 기반의 비상 호출 전화기입니다. 비상호출 기능을 필요로 하는 여러 장소에서 사용될 수 있으며, 다양한 음성코덱 과 마이크, 스피커 인터페이스, PoE 기능을 통해 ADSL 같은 일반적인 인터넷 환경 및 애드팍 IP-PBX 솔루션 환경에서 IP 텔레포니 서비스를 최적으로 지원합니다.
AP-EIP70 IP 비상 호출 전화기는 애드팍이 그동안 VoIP 분야에서 축적해온 음성처리기술, 신호처리기술, 네트워크 기술이 탑재된 IP 비상 호출 전화기입니다. 여러 기능에서 AP-EIP70은 확실하게 다릅니다. 고품질 음성 DSP 기반으로 설계되어 Acoustic Echo Canceller 기능을 기본 탑재하여 내장 마이크, 내장스피커를 사용한 Full Duplex 방식의 양방향 통신 기능을 지원합니다. 또한 AP-EIP70은 SIP VoIP 시그널링 프토콜을 비롯해 대역폭에 따라 다양한 음성 코덱 G711, G726, G729, G723.1등을 지원하기 때문에 인터넷으로 연결된 곳이라면 어디서나 사용할 수 있습니다. 특히 제한된 대역폭 상황에서 인터넷 상의 다양한 지연 및 패킷 오류에 대응하기 위해 QoS기술을 적용해 최적의 음질을 보장합니다. 애드팍 IP-PBX 솔루션과 연동하여 IP 비상호출전화 기능을 수행할 수 있습니다.
AP-EIP70 IP 비상 호출 전화기는 PoE 기능이 탑재된 한 개의 10/100M 고속이더넷 포트를 제공합니다. PoE 기능은 별도의 전원선이 필요없이 LAN 선을 통해 장비가 필요로 하는 전원을 급전하는 방식입니다. PoE 기능이 탑재된 L2 스위치나 PoE Converter에 연결되어 사용될 수 있습니다.
AP-EIP70 주요특징
1. 통신방식 : IP 기반
2. 프로토콜 : SIP 지원
3. 음성 : G.722(16KHz), G.711, G.729a, G.726 등
4. 네트워크 : 10/100Mbps 고속 이더넷 (1포트)
5. 마이크, 스피커 내장
6. PoE(Power over Ethernet) 전원공급기능 탑재 (옵션)
7. I/O 인터페이스 지원 : 릴레이아웃 (2포트), 알람입력 (2포트)
8. RS232/RS485 인터페이스 지원
9. 외부 오디오 출력 포트 지원 (앰프스피커 연결용, 소음이 심한곳에 사용목적)
10. 외부 전원 공급 어뎁터 지원 : 12V, 1A
11. 웹기반 관리 소프트웨어 탑재 (Smart Web Manager)
12. 전문가급 설정지원을 위한 CLI (Command Line Interface) 지원
13. 네트워크 프로토콜 Enable/Disable
14. 외부 접근제어 IP 액세스 필터링 기능
15. CLI (command Line Interface) 설정기능 (via Telnet)
SIP 비상통화장치 수신용 AP-IP300 SIP 관제 전화기는 25개의 LED 프리젠스 기능이 탑재된 단축다이얼 기능이 기본적으로 탑재되었습니다. . AP-IP300 애드팍 SIP 전화기는 인터넷을 이용해 고음질의 음성통신이 가능한 IP 전화기입니다. 다양한 음성 코덱과 25개의 단축다이얼 버튼, 다양한 I/O 인터페이스를 통해 ADSL 같은 일반적인 인터넷 환경 및 애드팍 IP-PBX 솔루션 환경에서 음성 텔레포니 서비스 등을 최적으로 지원합니다. 애드팍 AP-IP300 SIP 전화기는 인터넷 음성통신과 방송을 결합한 새로운 IP 전화기입니다. SIP, H.323 멀티플 VoIP 시그널링 프토콜을 비롯해 G.722(16KHz), G.711, G.726, G729a, G723.1 음성 코덱등, 스테레오 오디오 인/아웃 입출력 인터페이스, Large Size 그래픽 액정 디스플레이, 다양한 기능키 지원, 두개의 기가비트 이더넷 포트, QoS 기능, 공인 IP 공유기능 등 첨단 기능과 서비스를 제공합니다.
5. Audio: G.722 (16KHz), G.711, G.726, G.729A, G.723.1, etc.
6. Camera: Built-in CMOS camera (Supports camera privacy cover)
7. Interface: 10/100/1000Mbps Gigabit Ethernet (2 ports), USB interface port, HDMI output port
8. PTT (Push to Talk) handheld microphone and built-in amplifier support (optional)
9. RoIP Service Support (LMR Gateway, PTT Server Integration Support)
10. Provision of 1:1 video call and multi-party call services (MCU connection)
11. Support for various screen layouts
12. Support for Secure SRTP Protocol (Cryptographic Algorithms: AES, etc.)
13. Real-time transmission bandwidth and packet loss display function
14. UI support for video conferencing
AP-VP230GHPTT Application Service Functions
1. Video Conferencing Application Service
2. Walkie-talkie Interconnected PTT Function Service
3. Video doorphone integration service function (e.g., AP-TAS280FR face recognition video doorphone)
4. SIP broadcasting service function
1. Video Conferencing Application Service
AP-VP230G One-on-One Call and Multi-Party Video Conferencing Feature Demo :
Introduction to Multi-Party Video Conferencing Features
The AP-VP230GHPTT IP video phone provides powerful multi-party video conferencing capabilities by linking with AddPac's multi-party video conferencing video MCU equipment and IP exchanges. In particular, the AP-VP230GHPTT supports an HDMI output port, enabling large-screen video conferencing services via external large Full HD monitors or TVs.
Key Video Conferencing Features
- Conference Room Settings
- Video conference participant invitation feature
- Video conference screen layout settings
- Participant name display feature
- Display Participant Information
- Display participant details
- Floor assignment function
- Floor Full Screen Feature
- Screen position setting function
- Information display functions (Chair/Participant, MIC Mute, Audio Mute, Video Mute (Privacy), Display Mute)
Introduction to Video Conferencing GUI Features (Example)
2. Walkie-talkie Radio Interconnected PTT Function Service
Gigabit AP-VP230GHPTT Radio Interconnection PTT Function Demo: (Motorola Radio via LMR G/W)
Network diagram
Marine RoIP Applications
3. Video doorphone integration service function (e.g., AP-TAS280FR face recognition video doorphone)
Network diagram
5-inch LCD Gigabit AP-VP230GHPTT Video Doorphone Control Terminal Function Demo :
AP-TAS280FR HD SIP Video Doorphone with Face Recognition Function
The AP-TAS280FR facial recognition IP video doorphone is installed at the front door or entrance to perform access control functions while viewing video feeds via the Internet. Additionally, the AP-TAS280FR features an optional time and attendance management function for internal users. Utilizing SIP standard VoIP signal processing capabilities, it offers excellent scalability by integrating with AddPac IP-PBXs, video phones, and softphones. Furthermore, it is equipped with a high-resolution Full HD camera sensor, designed to be applicable to a wide range of applications. The selection criteria for IP video doorphones today are compatibility with standard VoIP signaling systems and convenience. It provides an enterprise IP video access control solution in conjunction with the AP-ACS1500 door access control system. To support diverse applications, the AP-TAS280FR is equipped with a 3.5-inch TFT-LCD.
The New Selection Standard for IP Video Doorphones: “Video Calling Function”
The AP-TAS280FR Face Recognition HD IP Video Doorphone is an IP video doorphone that combines the signaling and voice processing technologies AddPac has accumulated in the VoIP field with advanced video processing technology. Since the AP-TAS280FR supports networks ranging from 256Kbps to 4Mbps, it can be used anywhere with an internet connection. In particular, it guarantees optimal image quality by applying a 'rate control' function to ensure optimal picture quality and frame rates in limited bandwidth situations, as well as high-end error resilience technology to cope with various packet errors on the internet. It is designed to offer optional support for two-way video calling, allowing users to view the screen of the internal caller on the doorphone. It supports the latest H.264 video codec.
SIP-based standard VoIP signaling protocol
The AP-TAS280FR supports the SIP standard protocol in terms of call control protocols. Therefore, it offers excellent interoperability with various VoIP terminals, such as video phones, that use different call control protocols. Furthermore, to ensure compatibility with IP-PBXs and SIP proxy servers, which are becoming increasingly popular, as well as with existing analog PBXs, it is provided alongside AddPac door control systems, enabling a service that allows access control while viewing video even within an internal network rather than an IP environment.
Low-light night call mode feature
The AP-TAS280FR face recognition HD IP video doorphone is equipped with a Sony CMOS Full HD camera sensor that offers excellent low-light performance. It is designed to allow video calls not only in night mode but also in dark places or basements without light.
Built-in face recognition and RF card functions
The AP-TAS280FR facial recognition IP video doorphone is installed at a company's entrance or lobby to primarily support video-based access control for external visitors from within the company. For convenient access control for internal employees, it provides access functions via a high-performance facial recognition module and RF cards.
Firmware upgradeable structure
Since both the high-performance RISC CPU and DSP of the AP-TAS280FR feature a programmable architecture, continuous functional improvements, changes, and additions are possible. Whenever new features are added or changed, you can download them directly from the website or set up automatic upgrades to use the latest features without any extra effort .
Smart Web Manager features
The AP-TAS280FR face recognition IP video doorphone is equipped with a Smart Web Manager function for management capabilities. Users can easily configure and manage the IP video doorphone via the web from PCs, smartphones, tablets, and other devices.
AP-TAS280FR Key Features
1. Communication method: IP-based
2. Protocol: SIP Support
3. Video: H.264 codec support, HD (1280x720), D1, CIF
4. Voice: G.722 (16KHz), G.711, G.726, etc.
5. Camera: Equipped with high-definition Full HD camera (with low-light function)
11. Night Performance: High-performance Full HD low-light Sony camera sensor
12. Touchscreen 3x4 Keypad
13. 3.5-inch TFT Color LCD: 320 x 240 video resolution
14. Access Control and Management: Facial recognition, RFID card interface, password support
15. Supports full-duplex voice communication (equipped with acoustic echo canceller)
16. Support for two-way video communication (optional); by default, video is transmitted from the doorphone to the control room.
17. RTP/RTSP protocol support, RTP video calling and door screen CCTV RTSP support
4. SIP broadcasting service function
Network diagram
Server-based SIP broadcasting server (e.g.)
The AP-PNC2000 SIP paging broadcast server is a SIP-based broadcasting equipment for Ethernet and Internet access that is installed in financial institutions, corporations, government offices, the military, and school networks based on a Linux-based hardware platform, and provides voice broadcasting transmission functions over the Internet network using the SIP protocol to remote branches, offices, and locations.
The AP-PNC2000 SIP Paging Broadcast Server utilizes SIP-based standard VoIP signaling protocols and the RTP Real-Time Transport Protocol, and can be used in standalone mode or as a legacy SIP IP-PBX clone. It provides an excellent solution for supporting voice and audio broadcasting operations over the Internet in enterprise environments and is designed to ensure maximum user satisfaction through its outstanding performance. The AP-PNC2000 SIP Paging Broadcast Server supports stable services through a Linux server-based hardware platform. Installation and maintenance are very simple, allowing even beginners to operate it easily.
The AddPac SIP broadcasting solution consists of SIP broadcasting servers such as the AP-PNC2000 and SIP broadcasting terminals including the AP601, AP602, and AP650. The AP-PNC2000 SIP paging broadcasting server provides SIP paging server functions using the Linux OS server hardware platform. It is designed to support stable service even in the event of a failure by providing system redundancy capabilities.
AddPac’s diverse series of IP broadcasting equipment, along with its VoIP product line, is recognized for its performance and stability not only in Korea but also in the global market. The AP-PNC2000 SIP paging broadcast server, infused with the experience and know-how AddPac has accumulated in the enterprise and telecommunications carrier markets, will provide high satisfaction in various service aspects to meet the needs of customers requiring audio broadcasting services based on the SIP standard protocol. By utilizing AddPac’s proprietary QoS algorithms in addition to standard QoS, it can provide stable service even on the Internet network, where jitter and packet loss fluctuations are inevitable.
SIP broadcasting software
AP-SPMS is a purely web-based paging broadcast management software. This software operates in various web browser environments, such as Internet Explorer, Chrome, smartphones, tablets, and tablet web browsers, and supports user registration and access restriction management for SIP paging broadcast servers, management of AddPac SIP paging broadcast equipment, management of broadcast sources, groups, and favorites, scheduled broadcasts and time signal broadcasts, remote broadcast management, broadcast equipment inspection, and event history inquiry functions.
AP-SPCS (SIP Paging Client Software) Smart Communicator is MS-Windows-based SIP broadcasting client software that enables high-quality voice broadcasting over the Internet. It optimally supports Internet-based SIP voice broadcasting services through the latest voice codecs and a convenient Graphic User Interface (GUI). The selection criteria for SIP broadcasting client software are now support for private IP environments, TTS (Text to Speech) functionality, and convenience.
Web-based next-generation smart SIP broadcasting control and software
Web browser-based
- Supports Internet Explorer and Chrome on PC (or Notebook)
IP-based remote broadcasting management software (PC or laptop-based)
Supports intuitive dashboard-style UI
Supports custom screen layout based on the user
GIS MAP or E-MAP-based broadcast terminal status display function
Broadcast server status monitoring
Introduction to AP602 SIP Broadcast Terminal
The VoiceFinder AP602 SIP VoIP Broadcast Terminal is a SIP VoIP broadcast terminal used in conjunction with AddPac's SIP-based VoIP broadcast server. It is installed in financial institutions, corporations, government offices, the military, and campus networks to provide voice broadcast services over the Internet network using the VoIP SIP signaling protocol to remote branches, offices, and locations. It is used in conjunction with the AddPac SIP VoIP broadcast server and can be used independently as a dedicated broadcast network as a SIP VoIP broadcast solution, or it can be connected to an existing legacy IP-PBX network like an IP phone. In this case, the AddPac SIP broadcast server is connected to the IP-PBX (SIP Call Manager) via a SIP trunk to perform broadcast services to standard SIP IP phones and the AP602 SIP broadcast terminal.
It is designed with volume up and down switches on the front panel, allowing users to adjust the desired broadcast volume according to their preference. Additionally, it supports a broadcast module with a built-in 40-watt (8-ohm speaker) digital amplifier, enabling direct connection of external speakers, making it a suitable solution for various IP broadcasting environments.
The AP602 SIP voice broadcast terminal provides an excellent solution for supporting SIP voice broadcasting over a local LAN, particularly in integrated security environments such as intrusion detection alerts. It is designed to ensure maximum user satisfaction through its outstanding performance. Installation and maintenance are very simple, allowing even beginners to operate it easily.
The AddPac SIP broadcasting solution consists of SIP VoIP broadcasting servers (or Legacy IP-PBXs) such as the AP-SPS180 and AP602 SIP VoIP broadcasting terminals. Depending on the options selected, the AP602 SIP VoIP broadcasting terminal is designed to accommodate a digital amplifier, allowing for voice broadcasting by directly connecting speakers without the need for an external amplifier. The AP602 SIP VoIP broadcasting terminal is designed to support VoIP voice codecs such as G.722 (16KHz), G.711, G.72, and G.729A. By combining high-performance DSP hardware and a high-performance processor designed and developed in-house by AddPac, the AP602 IP voice broadcasting terminal can decode VoIP voice signals in real time for broadcast playback.
Introduction to AP650 120-Watt Network PA Amplifier SIP Broadcast Terminal
The AP650 SIP broadcast terminal is a SIP broadcast terminal designed to be used in conjunction with the AddPac SIP broadcast server. It is installed in locations where wired networks are difficult to support and provides voice broadcasting services to remote locations using the IP protocol via a 4G LTE wireless network. It is designed to support voice codecs such as G.722 (16KHz) and G.711 by adopting a RISC+Voice DSP signal processing processor.
The AP-PA6000 SIP network broadcasting card, located on the front panel, is designed using a RISC+DSP chipset and features a rotary volume control switch, allowing users to adjust the desired broadcast volume according to their preference. In addition to the LAN port, it supports external device control ports. It supports an alarm input port (1 port) and relay output ports (2 ports), and beyond basic broadcasting functions, it can be used for various purposes by connecting a warning light (siren) or an emergency call button. It also supports an ON-AIR blue lamp to display the real-time broadcast status. The 3.5 MIC input interface on the AP-PA6000 card can be used to diagnose or monitor the broadcast output status at the site from a remote location. By connecting an external microphone, real-time broadcast audio can be listened to in the center control room, making it a crucial function for identifying broadcast failures.
The AP650 120-watt SIP broadcast terminal with built-in PA amplifier provides an excellent solution for supporting IP broadcasting operations via Local LAN, particularly in integrated security environments such as intrusion detection alert broadcasts. It is designed to ensure maximum user satisfaction through its outstanding performance. Installation and maintenance are very simple, allowing even beginners to operate it easily.
The AddPac SIP broadcasting solution consists of the AddPac SIP broadcasting server (AP-SPS180), AP-SPMS SIP broadcasting management software, SIP broadcasting telephones, and AP650 SIP broadcasting terminals. The AP650 SIP broadcasting terminal is designed to provide broadcasting by directly connecting speakers without an external amplifier. Supporting a 120-watt PA amplifier (100V, 70V, 4 ohms), it allows for the direct connection of various types of external speakers, such as high-impedance speakers, making it a suitable solution for diverse IP broadcasting environments.
