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VoIP (viết tắt của Voice over Internet Protocol, nghĩa là Truyền giọng nói trên
giao thức IP) là công nghệ truyền tiếng nói của con người (thoại) qua mạng
máy tính sử dụng bộ giao thức TCP/IP. Nó sử dụng các gói dữ liệu IP (trên
mạng LAN, WAN, Internet) với thông tin được truyền tải là mã hoá của âm
thanh. Công nghệ này bản chất là dựa trên chuyển mạch gói, nhằm thay thế
công nghệ truyền thoại cũ dùng chuyển mạch kênh. Nó nén (ghép) nhiều
kênh thoại trên một đường truyền tín hiệu, và những tín hiệu này được truyền
qua mạng Internet, vì thế có thể giảm giá thành.
Để thực hiện việc này, điện thoại IP, thường được tích hợp sẵn các nghi thức
báo hiệu chuẩn như SIP hay H.323, kết nối tới một tổng đài IP (IP PBX) của
doanh nghiệp hay của nhà cung cấp dịch vụ.
The Session Initiation Protocol (SIP) is a communications protocol for signaling,
for the purpose of controlling multimedia communication sessions. The most
common applications of SIP are in Internet telephony for voice and video calls,
private IP telephone systems, as well as instant messaging over Internet
Protocol (IP) networks.
Voice over Internet Protocol (VoIP)
Session Initiation Protocol (SIP)
VoIP Platform
WebRTC
Use Cases
Content
WebRTC Gateway & Client
VoIP Platform1
Sipwise VoIP Platform
Sipwise RTC:engine
Aarenet VoIP System Aarenet VoIP SystemIP NetworkEnd-User PSTN Inter-
connectionPSTN
Call Balancer
Load Balancer
Call AgentCall Agent
Service CenterService Center
Config Center
Load Balancer
Config Center
Call Balancer
Service Center
Call Agent
IP-PBX2
PBX (Private Branch Exchange) is a system that connects telephone extensions
to the PSTN (Public Switched Telephone Network) and provides internal
communication for a business. An IP-PBX is a PBX with Internet Protocol
connectivity and may provide additional audio, video, or instant messaging
communication utilizing the TCP/IP protocol stack.
Asterisk is an open source framework for building
communications applications. Asterisk turns an
ordinary computer into a communications server.
Asterisk powers IP PBX systems, VoIP gateways,
conference servers and other custom solutions. It is
used by small businesses, large businesses, call
centers, carriers and government agencies,
worldwide. Asterisk is free and open source. Asterisk
is sponsored by Digium
FreeSWITCH is a scalable open source cross-platform telephony
platform designed to route and interconnect popular communication
protocols using audio, video, text or any other form of media. It was
created in 2006 to fill the void left by proprietary commercial solutions.
FreeSWITCH also provides a stable telephony platform on which many
applications can be developed using a wide range of free tools.
FreeSWITCH was originally designed and implemented by Anthony
Minessale II with the help of Brian West and Michael Jerris. All 3 are
former developers of the popular Asterisk open source PBX.
WebRTC Gateway & Client3
WebRTC is a free, open project that provides browsers and
mobile applications with Real-Time Communications (RTC)
capabilities via simple APIs. The WebRTC components have been
optimized to best serve this purpose.
WebRTC enable rich, high quality, RTC applications to be
developed for the browser, mobile platforms, and IoT devices,
and allow them all to communicate via a common set of
protocols.
The WebRTC initiative is a project supported by Google, Mozilla
and Opera, amongst others.
WebRTC Architecture
WebRTC Gateway – Frafos
Protocol Reference Diagram
WebRTC GatewayWebRTC Browers SIP Equipment
WebRTC Client – JsSIP
JsSIP is a simple to use JavaScript library which
leverages latest developments in SIP and WebRTC to
provide a fully featured SIP endpoint in any website.
With JsSIP any website can get Real Time
Communications features using audio, video and
more with just a few lines of code.
SIP over WebSocket transport.
Audio/video calls, instant messaging and presence.
Lightweight!.
100% pure JavaScript built from the ground up.
Easy to use and powerful user API
JsSIP
JsSIP
JsSIP
Use Cases4
E1 Gateway
SIP
Customer have PBX
VHTVoIP Platform
FXS Gateway
SIPFXO
Analogue PBX
E1
E1 PBX
E-SBC
SIP
IP PBXIP Phone
SS7 or SIP Trunking
IP NetworkSIP
VHT
SIP Trunking: CMC, FPT,
VNPT MetroNET, Viettel E1
SS7: Mobifone, Vinaphone,
Viettel, Vietnamobile, Gtel
VHT – VoIP Platform
IP NetworkSIP
SIP
IP Phones
Router / PoE Switch
SIP
IP Phones
Router / PoE Switch
VHTVoIP Platform
VHT
SS7 or SIP Trunking
SIP Trunking: CMC, FPT,
VNPT MetroNET, Viettel E1
SS7: Mobifone, Vinaphone,
Viettel, Vietnamobile, Gtel
Customer do not have PBX
Vcare - VStack
VHT – VoIP Platform
VCare is a solution for receiving and taking care of Omni channel customers,
including: Facebook, Email, SMS, Web chat, Web form and Call…
VHT – VCare
VHT – VCare
Rich Chat
Voice Call
Video Call
Call-out to number
Broadcast
Conference
Live Streaming
VStack is an integrated SDK Mobile App
support Full Communication Stack for
Startup / Mobile Developer integrates Chat
/ Call / Video Call ... and Call-out to
number (SIP Trunking) into Android, iOS
and Web applications.
VHT – VStack
Voice Call
Video Call
Call Out
VHT – VStack
VHT - VStack Architecture
Customer Relationship Software - GetFly
Point Of Sales - iPOS
WebRTC Gateway
Call Me
3. SIP call
5. Exchange Media
Online Customer Interaction
1. Users download the application as a Javascript
2. Caller establishes a call with the WebRTC gateway
3. WebRTC gateway established a VoIP call to Call center
4. Caller and Call center exchange data
Contact Center Software
Team Communicator
Network Operation Center
Human Resource Management
Dating Site
Online Medical Consultation
