Software Engineering 4C03 VoIP: The Next Telecommunication Frontier

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1 Software Engineering 4C03 VoIP: The Next Telecommunication Frontier Rudy Muslim McMaster University Computing and Software Department Hamilton, Ontario Canada

2 Introduction Voice over Internet Protocol (VoIP) is one of the latest technologies that has become a mass-market technology. Liberalization of the telecommunication industry in many countries has helped fuel the demand for cheap and accessible telephone service. In addition the availability of high-speed (broadband) internet such as cable modem and Digital Subscriber Line (DSL) has also helped in popularizing the VoIP technology. VoIP research started in 1995 when a couple of hobbyists started exploring the idea of communicating through their computers. Currently in the United States, there are numerous companies competing for a share in this ever-growing market. The concept behind VoIP is that voice is digitized and then the digitized signal is compressed before being sent over the internet and when it reaches the intended party, the digitized voice is then converted back to voice which is heard at the end of the line. This means that as long as the sending and receiving party both have access to a computer and the internet, they are able to communicate through a computer and in the process save the cost of communication through a regular telephone. Most people are using VoIP without knowing that they are using this marvelous technology. When a person is using a calling card to make a long distance call, they might be using VoIP to communicate. VoIP helps in cutting the cost of communication for businesses. Consider the case where an executive has gone abroad. To call back to his office, he has to pick up the phone and dial the number. Through VoIP he will reduce the cost of communication since usually VoIP provider offers a monthly flat rate. The advances in VoIP have made this technology more accessible even to the ordinary people. There are products where you do not even need a computer to be able to use VoIP. It is predicted that the VoIP traffic will reach 25%-40% of the total world voice traffic by end of the year There are two main protocols that have been used in the VoIP industry - H.323 protocol and Session Initiation Protocol (SIP). The former was introduced by International Telecommunication Union (ITU) while the latter by Internet Engineering Task Force (IETF). Both protocols are widely used in the VoIP industry but H.323 is the protocol of choice. To better understand how VoIP works, one must look into the details of H.323 and SIP. H.323 H.323 is made up of several other protocols: H.255.0, H.245, Real-time Transport Protocol (RTP) and T.120. H is used for signaling to initiate the communication, H.245 is used to control the call and provide feedback during the call while RTP is used for the transfer of real-time data needed to maintain the call. Lastly, T.120 is used to exchange data during the call. These four which are encapsulated in H.323 are the basic protocols used in establishing IP telephony. This is then passed through the User Datagram Protocol (UDP) and Transmission Control Protocol (TCP) and then through the Internet Protocol (IP) to be transmitted over a network. Since H.323 uses UDP and TCP to send and

3 receive data, the reliability of H.323 depends on the reliability that the UDP and TCP connection provides. The diagram below illustrates how each protocol is used in dealing with IP telephony. Audio Video Application Video Audio codec codec RTP RTCP Signaling and Controlling H.255 Registering H.255 Signaling H.245 Control Data Application T.120 Data UDP TCP IP Fig1: H.323 Protocol (adapted from Internetworking with TCP/IP by Douglas E.Comer) Besides H.323 protocol, the protocol also includes several others system that make up the H.323 architecture. These include 1. Gateway functions to link a H.323 terminal with another H.323 terminal or with a Public Switch Telephone Network (PSTN). 2. Gatekeeper is similar to a switching board in a PSTN; also provides control and management functions such as bandwidth allocation, and translation of an IP address. 3. Multi Control Units (MCUs) allow more than two H.323 terminals to participate in a conference. Usually in addition to the H.323 basic protocol, other protocols such as H.450 suite are used to provide similar functionalities that a regular PSTN provides such as call waiting, caller ID and call forwarding. To establish a session from terminal A to terminal B there are several steps involved. First, terminal A must connect to the gatekeeper. Then the gatekeeper will setup the session between terminal A and B after which the initial exchange of capabilities of the type of communication is done. This step is then followed by the exchange of multimedia capabilities. The communication is then established. To terminate the session, the H.245 protocol is initiated to end the session. As seen here, one advantage that H.323 has over SIP is that it provides a complete suite of applications to complete a session of Voice over Internet Protocol communication. Session Initiation Protocol (SIP) As mentioned before, SIP is a VoIP protocol that is being pushed by IETF. SIP is mainly described in RFC SIP is being pushed as the next evolution of the VoIP technology as SIP is much simpler than H.323. In contrast to H.323 which provides all the necessary protocol to make a VoIP connection, SIP only provides the basic essential in signaling a session to start and end a session. As a result, there are many applications that can be built around SIP. Consequently, SIP is considered to be more versatile than H.323. The interaction that occurs in SIP is very similar to an HTTP, which is based on a client-server interaction

4 Since SIP is based on the client-server type interaction, there are two main components that make up a SIP network: the user agent server and the network server. The user agent server is the client-end part which provides the communication with the network server. The network server is used to initiate the session from terminal A to terminal B. A user of SIP protocol is assigned an -like identifier Like H.323 protocol, there are several steps involved in connecting one user to another. The first step is to send the request to connect in the form of the identifier above to the server. The second step involves a local SIP server resolving the destination and making the connection if the user is locally connected; in the case the server fails to resolve the destination, the server will make a request to resolve the destination similar with an http connection. If the destination is found, the server will forward the request to start a session. The destination party would have several options; it can accept or reject the request. If the request is rejected, it can be forwarded to other destination such as a voice mail. An advantage that SIP has over H.323 is that in a single connection, different type of media can be transmitted. Moreover, SIP has the capabilities to ring several extensions at the same time. Both H.323 and SIP have their advantages and disadvantages. The following section will discuss this issue. Comparison H.323 and SIP There are many advantages and disadvantages that these two protocols have. This following section will give a brief explanation in the differences of the two protocols. SIP is based on the understanding that VoIP connection is made through Wide Area Network while H.323 is based on the more restricted Local Area Network. SIP communication is done on text-based messages while H.323 communication is done in binary format. This might pose difficulties for certain implementations since most of the communication over the internet is actually text-based. There is error checking on H.323 which will cause more delay in the actually sending of the signals while on SIP the reliability is assumed since the error checking is done on the network level. SIP handles only the signaling part of the communication and therefore it allows for more freedom in the implementation while H.323 handles everything from the data coding to the signaling. As a result, H.323 protocol does not provide much flexibility for VoIPbased products. Conclusion The availability of bandwidth for the average user will propel VoIP to the next level. VoIP provides the reliability that one can expect from the regular PSTN. This will make VoIP the next logical step in the expansion of telecommunication. As more and more countries embrace VoIP, the cost to communicate through VoIP will decrease. One should try VoIP as it will reduce the cost of communication for the individual as well for businesses.

5 References E. INTERNETWORKING with TCP/IP: Principles, Protocols, Comer, Douglas Architectures, 4th Edition. New Jersey: Prentice Hall, History of VoIP. Last accessed: 24 March 2004 Lucy P. Roberts. History of the telephone. Last accessed: 24 March Last accessed: 25 March 2005 Voice over IP reference page. Last accessed: 25 March 2004

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