Design and Develop a Telephony System for Establishing Calls between Two PBX Servers Using IAX2 Protocol

Telephony system is the most important part of communication. In Modern communication circuit-switched network is common way for person to person communication which is referred to as Public Switched Telephone Networks (PSTN). But communication cost is high, unreliable, maintenance is complex, call routing is complex in circuit-switched network. To reduce these problems, a new trend that is beginning to emerge in recent years is to provide telephony service over IP networks, known as IP telephony. Advantage of this network is cost savings, especially for corporations with large data networks. This system is easy to configure, easy to implement within campus, in a business office and other sectors with no cost. The main advantages of IP telephony is easily integrate with existing system like PSTN, GSM network. These systems are not only transfer calls but also send email, chat with others, establish video calls etc. In telephony system server to server communication or server to user communication there are different types trunking protocol such SIP, IAX2, RDP, H.323 etc. In this project IAX2 protocol is used.

The objective of this project is to design and develop a telephony system within a local LAN or Wireless network. This system maintains communication within an organization through telephony system which reduces communication cost dramatically. It provides communication within a local network.
• Develop a telephony system within a LAN or wireless network
• Configuration of two PBX server
• Connecting two PBX using IAX2 trunking protocol
• Maintaining communication of an organization within LAN or wireless network
• Reduce communication cost within an organization
• Users can use both soft phone and telephone

There are different trunking protocols such as H.323, SIP, and SDP etc. Comparing SIP with protocols we can say that SIP protocols best of them, but it has some limitation. Few years ago, IAX protocol is invented and then it is one of the best over SIP. Different types of Telecommunication Company such as Sussex [1], NetTelco [2] use IAX2 protocol for their communication.

Bandwidth consumption comparison of SIP and IAX [3,4] (Table 1)

Table 1:BW of some protocols with codec.

From Table 1 it is seen that
a. SIP & G.711 codec produce very good quality of voice, but it consumes highest bandwidth.
b. IAX & GSM codec consume lowest bandwidth, but it produced high traffic.
c. IAX & G.711 codec requires high bandwidth and ideal for power traffic level is relatively high.
d. SIP & GSM codec is ideal for plans that do not support IAX.

Analyzing the table data, we can say that IAX protocol is good call transferring.

So IAX2 is used and develop this system with the following stepsi.
Environment Setup
ii. Connecting Asterisk PBX [5] server with IAX2 trunking protocol and establish call session in real-time network.
iii. Check the system and protocol with others for development.

First of all, we need to setup open source Linux Operating then install Asterisk PBX, Free PBX web GUI on it. Calculate IP for PBX and terminal devices.
• Configuring IAX2 Trunks
• Configuring the Outbound Routes
• Create channels for end users and bind them with PBX
• Test Call generate and service testing

In this system Asterisk PBX, Different types of terminal devices, IAX2 tunneling protocol are used (Figure 1).

Asterisk architecture

Asterisk consists of five base components [6]:
• Dynamic Module Loader - When Asterisk was first started, the Dynamic Module Loader loads and initializes each of the drivers which provide channel drivers, file formats, call detail record backends, codecs, applications and more, linking them with the appropriate internal APIs.
• PBX Switching - The essence of Asterisk is a Private Branch Exchange Switching system, connecting calls together between various users and automated tasks. The Switching Core transparently connects callers arriving on various hardware and software interfaces.
• Application Launcher - launches applications which perform services for users, such as voicemail, file playback, and directory listing (Figure 2).

• Codec Translator - uses codec modules for the encoding and decoding of various audio compression formats used in the telephony industry. A number of codecs are available to suit diverse needs and arrive at the best balance between audio quality and bandwidth usage.
• Scheduler and I/O Manager - handles low-level task scheduling and system management for optimal performance under all load conditions.

There are many types of trunking protocols. Among them we differentiate SIP and IAX2 [7] (Table 2)

Table 2:Difference between IAX2 and SIP.

Implementation Step 1:

For setup Asterisk we need to run Linux Server and Centos is such kind of server. So, first setup centos 6.4 server from DVD or ISO (Figure 3).

Step 2:

After completing the installation, the login screen appears that means CentOS setup is finished. Now it is ready to provide root and centos as login username and password. Then it appears terminal and writes the command for installing Asterisk and FreePBX GUI [7-9]. After install Asterisk and Free PBX GUI, it is ready to use (Figure 4).

Step 3:

Normally server has default configuration with DHCP, and dynamic addresses are change frequently. So, it needs to change with static IP. For implementation IP with network 192.168.43.0 and subnet mask 255.255.255.0 is used.
IP addresses of each device (Table 3)

Table 3:Assigning IP address.

Step 4:

For configuring PBX first, it is needed to access the server. This can be described in two ways. One is by SSL by terminal of another PC and another is by accessing the GUI which is previously installed into it. This GUI runs on web server reside on it. For graphically access first find the IP address and put it on the remote browser in the same network. Then we access the Graphical login page (Figure 5).

Step 5:

Accessing the GUI from browser it asks username and password. After inserting the username and password “FreePBX System Status” page will appear. We start our main configuration from here (Figure 6) [10].

Step 6:

For integrate terminal device need to channel which transfer calls between end device and PBX. So, configure the extension file with user number and password including channel protocol (Figure 7).

Step 7:

For transfer call from one server to another server it is necessary to configure trunk with appropriate transfer rule. If the dialed number matches with the dial manipulation rule the trunk transfer the call, otherwise not (Figure 8).

Step 8:

Need to configure handset for connecting to PBX 1. For this it is necessary to provide account name, host, username, password etc. After providing the correct information click save button and it automatically connects the PBX server (Figure 9).

Implementing the protocols, the project found that IAX2 is better than SIP for trunking between two PBX servers. Comparisons are shown in Table 3 between IAX2 and SIP [11]. This system serves better than the system which is configured by SIP protocol used in various existing system. If the server resides different network, it fails to communicate [12].

• We design and develop a new trunking protocol which reduces more bandwidth.
• Develop a pre-configured bootable OS which automatic connect to the radius server using MAC address verification.
• Implementation of MAC addresses authentication and verification.

None.

No conflict of interest..