VoIP Test Base Unit: Framework And Protocol

    Figure 1 is the architecture of VoIP Test Base Unit. Each component of VoIP system interacts to others by SIP protocol.

    The following is the description about these components.

Figure 1: VoIP Test Base Unit


    SIP Proxy Server (Figure 1(a)) provides the basic communication connection control in VoIP Test Base Unit, and processes the request and response of SIP as SIP Proxy Server.

    Furthermore, it was used as Registrar Server for saving user contact information. Presently, We have a SIP Proxy Server executing on Windows 2000 industry computer, and another SIP Express Router on FreeBSD and Linux.

    Figure 2 is OAM (operation, administration & maintenance) system of SIP Proxy Server. The figure shows the SIP administrative page of SIP Proxy Server.



Figure 2: OAM system of SIP Proxy Server.


    PSTN (Public Switch Telephone Network) Gateway (Figure 1(b)) supports the operation between VoIP Test Base Unit and PSTN, and it allows users of IP phone contacts to other users of PSTN directly or indirectly by PBX (Figure 1(c)).

    PSTN Gateway deployed on VoIP Test Base Unit now is Cisco 2621xm.



Figure 3: Cisco 2621XM PSTN (Public Switch Telephone Network) Gateway


    SIP VoIP is a SIP phone client (Figure 1(d)) executed by hardware or software, and provides the basic function of phone such as dial, answer, hang up, hold on/do not hold on and transfer.

    We have installed SIP VoIP in the terminate such as PC, notebook (accessed with or without wireless LAN) and PDA (accessed with wireless LAN only). Figure 4 is the graphic user interface of softphone.


a) Windows Messenger 4.7(including a SIP UA) on Windows XP


(b) X-Lite


Figure 4: the graphic user interface of softphone.


    In addition, we also have hardware phones produced by Cisco, Leadtek, Pingtel and Snom (Figure 5).



(a) Cisco Hardware phone


(b)InnoMedia  video phone


(c) Pingtel Hardware phone


(d)Snom Hardware phone

Figure 5: user terminate.


  The same VoIP Test Base Unit also can be deployed at other sites. There are several SIP hardware and software phones at each site which is deployed with SIP Proxy Server and PSTN Gateway. Figure 1 is illustrated by SIP Test Base Unit of NCTU and NCNU. In the base unit of network, according to the position of sender and receiver, there are the following four communication building ways regardless of SIP phone or PBX/PSTN.

        Way 1: If the SIP phone UA1 in NCTU tries to dial to another SIP phone UA2 in the same campus, then the control signal for building communication would transport between UA1 and UA2 indirectly by the SIP Proxy Server of NCTU. Afterwards, UA1 and UA2 can build voice connection directly, instead of through SIP Proxy Server of NCTU.

        Way2: The SIP phone UA1 in NCTU tries to dial to the SIP phone in NCNU. At first, UA1 would transport control signal to SIP Proxy Server of NCTU. After SIP Proxy Server of NCTU confirms by the destination address that UA2 is in NCNU, it request SIP Proxy Server of NCNU to build connection. When SIP Proxy Server of NCNU finds out the register information of receiver, it builds the connection to UA2. Afterwards, UA1 and UA2 build a voice connection directly instead of through SIP Proxy Server of NCTU.

        Way 3: If the SIP phone in NCTU dials to the extension P1 or traditional PSTN phone in the same campus, the procedure of building communication will be similar to Way 1 except the following step: At first, SIP Proxy Server of NCTU confirms that the receiver is not a SIP phone and transfers it to PSTN Gateway of NCTU, then PSTN Gateway builds the communication to P1 directly (if P1 is a extension in the campus) or sends it to the local telephone service machine room (of telecommunication company) for further process.

  Way 4: If the SIP phone UA1 in NCTU tries to dial to the extension P1 in NCNU, SIP Proxy Server of NCTU would send the request to SIP Proxy Server of NCNU. The later procedure is similar to Way 3. SIP Proxy Server of NCNU would send it to PSTN Gateway for further process. To dial to SIP phone from PSTN, SIP phone needs to be assigned a phone number. According to the phone number assigned method (based onE.164 by suggestion) in Taiwan, the phone number format of SIP phone under VoIP Test Base Unit is 0944-nnn-xxx.(Electronic Number)
The front four numbers 0944 among it is the service code examined by Taiwan DGT for experiment purpose and under VoIP Test Base Unit. (Japan's ENUM (Electronic Number) experiment will begin from 050 as VoIP.) The following three numbers nnn is about the position. The deployment now is 001 for NCSU, 002 for NTU and 003 for NCTU. As for the rest numbers xxx stands for the automated produced client phone number by VoIP register system. Some numbers among them are reserved for emergency or special services.

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