VOIP TELEPHONY TRAINER

							            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 VoIP’s 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                                         3.   Voice compression and IP streaming techniques
      Call Detail Records                                     To study and elaborate the voice compression
      Support G.723.1 (6.3K / 5.3K), G.729 A/B,                   coding techniques and outcomes over difference
        G.711 (A-law / µ-law), G.726 voice codec                    bandwidth network environments and voice qual-
      Backup system configuration through Web and                 ity demands.
        USB flash disk                                          To study and elaborate the methods of voice data
      Built-in NAT and Firewall functions                         streaming over internet and the communication
      One Touch Dialing with ET747K                               protocol for internet phone – RTP/RTCP
      Build-in 3 FXO/CO-lines, 1 FXS for life line or   4.   SIP communication protocol and its functions
        FAX                                                     To study the basic principle of SIP internet
2.   ET747 Internet Phone                                           phone communication protocol and its utilization
      Supports SIP (RFC3261)                                      on various internet telephony call flow.
      Supports SDP (RFC2327)                                  To setup the SIP internet phone communication
      Supports RTP (RFC1889)                                      environment and verify its working principle and
      Supports RTCP (RFC1890)                                     various call flow.
      Support G.723.1 (6.3K / 5.3K), G.729 A/B,         5.   Introduction to IP-PBX
        G.711(A-law / µ-law)                                    To introduce the functions and operations of IP-
      Adjustable Audio Frame Per Packet                           PBX based on CC100.
      Adaptive Jitter Buffer Control                    6.   PC internet phone – Softphone
      In-band DTMF, Out-of-Band DTMF Relay                    To carry out softphone on PC by introducing
         (RFC2833, SIP INFO)                                        softphone’s software structure, operations and
3.   Ethernet HUB 10 Mbps                                           working principle.
4.   EG202 VoIP Gateway                                         To explain the softphone’s open source codes
      Following RFC-3261 SIP standard                             using example.
      Dynamic IP support (DHCP and PPPoE)                     To compare the characteristics of various types
                                                                    of softphone.
      Support G.723.1, G.729A/B, G.711(A-law / µ-
         law) voice codecs                                      To implement softphone using CC100 IP-PBX
5.   Analog Telephone                                               and ET747 internet phone to perform phone call.
6.   CC100 SDK tools and example codes                    7.   Webpage internet phone – Web Call
                                                                To introduce and implement web call using
Topics Covered                                                      CC100 as server.
                                                          8.   VoIP Gateway
1.   Introduction to basic principle of internet telephony      To introduce the software structure and working
2.   Introduction to internet telephony system                      principle of VoIP gateway.
      To introduce the basic configuration of ET747           To setup, configure and implement EG202 VoIP
          internet phone and CC100 internet phone PBX.              gateway via CC100 IP-PBX to make call to
          To setup the internet telephony system and to             internet phone or PSTN landline.
          understand internet telephony protocol and the 9.    PC’s IP-PBX Softswitch
          usage of network protocol analysis software –         To learn the installation of Asterisk.
          Wireshark.                                            To implement IP-PBX functions using
                                                                    Softswitch.

                                                                                  MP-SCIENTIFIC
            TELECOMMUNICATION LABORATORY
TL8303 VoIP TELEPHONY TRAINER

10. Asterisk Dial Plan setup                                  15. IP-PBX management system
     To introduce the setup of Asterisk Dial Plan and            To introduce IP-PBX management system and its
         explain the setup by using examples.                         various functions.
     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 telephony
         service provider by using CC100 IP-PBXs and
         ET747s.
     To understand the operation and interconnection
         of 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 PBX
    manager to self-develop the AGI program and inte-
    grate into the dial plan to allow Asterisk PBX’s 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 AGI’s mecha-
         nism and using open source code AGI library -
         CAGI as example to describe AGI programming
         methods.
     To describe the functions of the AGI’s program
         examples.
13. Firewall and NAT
    The 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
                                                                                                                     Subject to change without notice




         used to break through Firewall / NAT and verify
         the function of STUN by experiment.                  ACCESSORIES
14. PJSIP library internet telephony Implementation
                                                                 AC Power Cord
     To introduce PJSIP library open source code as
                                                                 Experiments Manual
         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.




MP-SCIENTIFIC