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TELECOMMUNICATION LABORATORY

TL8303 VoIP TELEPHONY TRAINER

MP-SCIENTIFIC

The VoIP Telephony Trainer is designed for student to

understand the principle and theory of VoIP. The topicsand experiments based on VoIPs hardware and software

cover from beginner level to advance level.

3. Voice compression and IP streaming techniques

To study and elaborate the voice compression

coding techniques and outcomes over differencebandwidth network environments and voice qual-

ity demands.

To study and elaborate the methods of voice data

streaming over internet and the communication

protocol for internet phone RTP/RTCP

4. SIP communication protocol and its functions

To study the basic principle of SIP internet

phone communication protocol and its utilization

on various internet telephony call flow.

To setup the SIP internet phone communication

environment and verify its working principle and

various call flow.

5. Introduction to IP-PBX

To introduce the functions and operations of IP-

PBX based on CC100.

6. PC internet phone Softphone

To carry out softphone on PC by introducing

softphones software structure, operations and

working principle.

To explain the softphones open source codesusing example.

To compare the characteristics of various typesof softphone.

To implement softphone using CC100 IP-PBX

and ET747 internet phone to perform phone call.7. Webpage internet phone Web Call

To introduce and implement web call using

CC100 as server.

8. VoIP Gateway

To introduce the software structure and working

principle of VoIP gateway.

To setup, configure and implement EG202 VoIP

gateway via CC100 IP-PBX to make call to

internet phone or PSTN landline.

9. PCs IP-PBX Softswitch

To learn the installation of Asterisk.

To implement IP-PBX functions using

Softswitch.

SPECIFICATIONS

1. CC100 Internet Phone IP-PBX

Built-in SIP Proxy server

IAX2

Automated Attendant

Interactive Voice Response

Voicemail

Call Detail Records

Support G.723.1 (6.3K / 5.3K), G.729 A/B,G.711 (A-law / -law), G.726 voice codec

Backup system configuration through Web and

USB flash disk

Built-in NAT and Firewall functions

One Touch Dialing with ET747K

Build-in 3 FXO/CO-lines, 1 FXS for life line or

FAX

2. ET747 Internet Phone

Supports SIP (RFC3261)

Supports SDP (RFC2327)

Supports RTP (RFC1889)

Supports RTCP (RFC1890)

Support G.723.1 (6.3K / 5.3K), G.729 A/B,

G.711(A-law / -law)

Adjustable Audio Frame Per Packet Adaptive Jitter Buffer Control In-band DTMF, Out-of-Band DTMF Relay

(RFC2833, SIP INFO)3. Ethernet HUB 10 Mbps

4. EG202 VoIP Gateway

Following RFC-3261 SIP standard

Dynamic IP support (DHCP and PPPoE)

Support G.723.1, G.729A/B, G.711(A-law / -

law) voice codecs

5. Analog Telephone6. CC100 SDK tools and example codes

Topics Covered

1. Introduction to basic principle of internet telephony

2. Introduction to internet telephony system

To introduce the basic configuration of ET747

internet phone and CC100 internet phone PBX.

To setup the internet telephony system and tounderstand internet telephony protocol and the

usage of network protocol analysis software

Wireshark.

7/31/2019 Voip Telephony Trainer

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TELECOMMUNICATION LABORATORY

TL8303 VoIP TELEPHONY TRAINER

MP-SCIENTIFIC

Subjecttochangewithoutnotice

ACCESSORIES

AC Power Cord

Experiments Manual

10. Asterisk Dial Plan setup

To introduce the setup of Asterisk Dial Plan andexplain the setup by using examples.

To setup the Asterisk Dial Plan under certain

environment with combination of CC100 IP-

PBX, EG202 VoIP gateway and ET747 internet

phone to verify and test the Dial Plan.11. IP-PBX setup under various environments and its

applications

To simulate the interconnection environment

between IP-PBX, PSTN and internet telephonyservice provider by using CC100 IP-PBXs and

ET747s.

To understand the operation and interconnectionof the actual internet telephony system by config-

uring the setting of the IP-PBX.

12. Internet value added voice service, Asterisk Gateway

Interface (AGI)

Other then powerful dial plan mechanism, Asterisk IP

-PBX provides extension mechanism function, Aster-

isk Gateway Interface (AGI), which enable PBXmanager to self-develop the AGI program and inte-

grate into the dial plan to allow Asterisk PBXs core

for AGI program execution. Asterisk provides vari-

ous AGI instructions which gives AGI program to

control the call flow.

To introduce the basic principle of AGIs mecha-nism and using open source code AGI library -

CAGI as example to describe AGI programming

methods.

To describe the functions of the AGIs program

examples.

13. Firewall and NATThe functions of Firewall and NAT may block inter-

net phone call process.

To describe the problems of internet telephony

due to Firewall and NAT and its solutions.

To introduce various solutions to solve the prob-

lems between internet telephony and Firewall /

NAT. To elaborate how the popular STUN protocol is

used to break through Firewall / NAT and verify

the function of STUN by experiment.

14. PJSIP library internet telephony Implementation

To introduce PJSIP library open source code as

example to write SIP softphone program, setting

up a simple SIP internet telephony program to

register with IP-PBX and to setup call for verify-

ing its functions during experiment.

15. IP-PBX management system

To introduce IP-PBX management system and itsvarious functions.