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TELECOMMUNICATION LABORATORY

TL8303 VoIP TELEPHONY TRAINER


The VoIP Telephony Trainer is designed for student to understand the principle and theory of VoIP. The topics and experiments based on VoIPs hardware and software cover from beginner level to advance level. 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 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

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(RFC2833, SIP INFO)


3. 4. Ethernet HUB 10 Mbps 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 Analog Telephone CC100 SDK tools and example codes

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Topics Covered 1. 2.

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Introduction to basic principle of internet telephony 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 to understand internet telephony protocol and the 9. usage of network protocol analysis software Wireshark.

Voice compression and IP streaming techniques To study and elaborate the voice compression coding techniques and outcomes over difference bandwidth network environments and voice quality demands. To study and elaborate the methods of voice data streaming over internet and the communication protocol for internet phone RTP/RTCP 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. Introduction to IP-PBX To introduce the functions and operations of IPPBX based on CC100. 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 codes using example. To compare the characteristics of various types of softphone. To implement softphone using CC100 IP-PBX and ET747 internet phone to perform phone call. Webpage internet phone Web Call To introduce and implement web call using CC100 as server. 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. PCs IP-PBX Softswitch To learn the installation of Asterisk. To implement IP-PBX functions using Softswitch.

MP-SCIENTIFIC

TELECOMMUNICATION LABORATORY
TL8303 VoIP TELEPHONY TRAINER
10. Asterisk Dial Plan setup To introduce the setup of Asterisk Dial Plan and explain the setup by using examples. To setup the Asterisk Dial Plan under certain environment with combination of CC100 IPPBX, 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 telephony service provider by using CC100 IP-PBXs and ET747s. To understand the operation and interconnection of the actual internet telephony system by configuring 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, Asterisk Gateway Interface (AGI), which enable PBX manager to self-develop the AGI program and integrate into the dial plan to allow Asterisk PBXs core for AGI program execution. Asterisk provides various AGI instructions which gives AGI program to control the call flow. To introduce the basic principle of AGIs mechanism 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 NAT The functions of Firewall and NAT may block internet 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 problems 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 verifying its functions during experiment. 15. IP-PBX management system To introduce IP-PBX management system and its various functions.

Subject to change without notice

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