Grokking TechTalk #18B: VoIP Architecture For Telecommunications

VoIP Architecture for Telecommunications

Au Duong Dat, VHT Chairmandat@vht.com.vn - 090 8495911

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