A Survey of Voice Over Internet Protocol (VOIP) Technology


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A Survey of Voice Over Internet Protocol (VOIP) Technology


1Department of Computer Science, Ramsar Branch, Payam Noor University (PNU), Ramsar, Iran,

E-mail: Jpourmail@gmail.com

2Department of Applied Mathematics, Lahijan Branch, Islamic Azad University, Lahijan, Iran,

E-mail: skarimimail@gmail.com

3Young Researchers Club, Lahijan Branch, Islamic Azad University, Lahijan, Iran,

E-mail: saedalatpanah@gmail.com

Abstract: The Voice Over Internet Protocol (VOIP) is as a combination of IP networks, voice applications and voice calls which being replaced by the old service conversation and created the revolution at the technical and conceptual framework of phone. This technology is an innovative form of phone that can dramatically increase performance and capacities of telephone service for business and individuals around the world. In this paper we give a survey of this new technology and present how this technology can be applied for the integration of voice and data networks. Keyword: VOIP, Communication methods, Quality of Service (QoS).


The Voice Over Internet Protocol (VOIP) is a technology that used to transmit voice conversations over a data network using Internet protocol [1-5]. VOIP is defined as “the ability to make telephone calls and send facsimiles over IP-based data networks with a suitable quality of service (QoS) and a superior cost/benefit” which can be used to transmit Voice over the Internet, provide a low cost communication medium. We know that, networks can be comprised of internet, intranet and network managed by the local ISP. The technology is opposed the traditional communication systems or public switched telephone network (PSTN).This technology is potentially much less expensive, but apart from this option, has created the dramatic changes in communication technology. As traditional telephone infrastructure in the last hundred years have made Meanwhile ,the new technology (VOIP) is related to modern Architecture which has implemented and developed On data networks, which potentially and inherently are Unreliable. VOIP significantly developed in recent years and is gaining the popularity and adoption of customer-friendly solutions such as SKYPE and BT strategies that are moving towards IP-based networks. In this paper we give a survey of this new technology and present some related definitions and challenges. Furthermore, we discuss the Implementation of VOIP, having an overview of all the protocols required to use the concept of Internet Telephony. We then discuss all the components required to set up a connection and describe the function of each.


VOIP is a set of technologies that can make voice calls over the Internet or other networks which indeed is replacement of the traditional PSTN systems. VOIP was invented By the VOIP Association


in the month may of 1996 as related groups to promote and develop higher quality products and services for launch the Internet telephone product. Voice over internet typically is associated with the communication, technological protocols, Methodology, methods of communication, voice communication and multimedia session of IP network.

The first objective of this invention was to reduce the cost of calls. While in traditional telephone networks a circuit must be implemented before any conversation is occurred between two contacts. Support for VOIP, has become especially attractive given the low-cost, flat-rate pricing of the public Internet. In fact, toll quality telephony over IP has now become one of the key steps leading to the convergence of the voice, video, and data communications industries. The feasibility of carrying voice and signaling message over the internet has already been demonstrated but delivering high-quality commercial products, establishing public services, and convincing users to buy into the vision are just beginning. VOIP have development the telecommunications, as use the IP protocol that had been designed as the Internet protocol to convert voice calls into digital packets and also transmit voice calls which are very sensitive to network delays and problems Similar to data transfer [6, 7]. In the PSTN mode each call dedicate certain portion of bandwidth will be available over the telephone network. Increasing number of call reduced the bandwidth Also the caller pays costs for the call time However in the VOIP method users pays a monthly charge to Internet service and VOIP calls and benefit free calls. Furthermore, the user shall pay a fee for special services. Packet switching is an efficient method was applied to enable multiple calls which are converted into IP packets for transmission over the multiplexed and shared network [8]. The advantage of this method is that the packets are directed to different routes and cannot be problems that are relevant to destruction of routers and affected lines. This bandwidth would not be particular to unit conversation and IP Packet will be moved with higher performance on shared networks. In addition this technology is able to manage many callers are in a moment. But the calls are divided into multiple packets that will face problems such as delays in receiving and be lost in crossing the channel.

Next, we describe some definitions about VOIP; 2.1 Internet Protocol Telephony-IP Telephony (IPT)

IPT refers to science or technology of phone service or computer network to convert analog audio to digital voices and then transferee them.Indeed it is the communication services which transfers data such as sms-fax-voice-video or massaging applications through the Internet using IP protocol without need to traditional telephone networks (PSTN) devices; see ([9]).

2.2 Internet Telephony Service Provider (ITSP)

A digital telecommunications service based VOIP which is provided via Internet and offers services directly to end users or other ITSP providers. This service uses protocols H.323, MEGACO, MGCP and SIP [10]. However, the VOIP is a subset of IPT, and is the technologies that use IPT to transmit voice calls. First VOIP providers presented business models and technological solutions that reflected the telephone network and second created limited network to specific users such as Skype.

It was very comfortable and free but does not possible to contact with the outside network. The third generations of providers such as Google talk accept federated VOIP content that was complete leap from traditional phone networks and enables users to communicate regardless of where the call came.



Most people know the VOIP through consumer of SKYPE that have found public recognition in recent years. However, SKYPE is just one example of implement of VOIP which have important technological history and close relationship whit telecommunications industry .The ideas of Voice over IP was first discussed in 1970 and Was introduced in 1995 by the VOLCALTEC Israeli company. These basic systems were for connecting computers and they had to contain Sound Card – Speaker – Microphone – Modem – VOIP software. The software codifies and compresses audio signals and converts them into packets to be transmitted over the network. From 1970, telecom companies have begun to offer Actuator IP software for their telephony equipment. The human voice is an analog wave signal and historically calls had been created on the network of analog circuits which provided End to end link for every call, and had been known as switching circuits. Most of companies that provided telephone service were the public agencies that are typically part of the postal office service’s country and these networks were identified as Post Office Telephone System (POTS). Public Switched Telephone Network (PSTN) is the name generally given to the networks that were created by the phone companies. Between 1950 and 1990, analog systems were replaced to digital networks and telephone exchanges were done by high-speed leased lines. This exchange were used from digital technology computers and digital signal protocols such as ISDN but communication still established via circuit switching, and copper wires. Since 1990 the companies that have manufactured phone equipment and communications, have started to increasing use of digital data transmission ideas between exchanges through packets related to IP. From mid-1990 manufacturers of telephony equipment added IP capabilities to existing telephony switch (PBX) and recently have been developed computer software which enable consumers to install VOIP adapter on the your regular telephones (in order to calls can be established on the PSTN network and through routing by GATEWAY on the internet network) [14]. Advances in VOIP technology were lead to the availability to telephone software of computer by many software providers. Gateway servers and voice processing card are as the interface between PSTN network and Internet that enable users to make calls through pc and the IP phone.

Furthermore, calls can be established through handsets that are similar to phone and are connected to the IP-based network and have more features than traditional phone.


The main processes of VOIP calls are include ([11-13]);

A. Convert the analog audio signals into digital format (ADC).

B. Compression and translates digital signals into internet protocol packets. C. Packet transmission over the Internet or a network based on IP.

D. Reverse translates the packets into analog signals for receiver.

The networks that are carried converted analog data to digital on them are organization’s intranet or a network can be rented. We need special software and broadband to make VOIP calls connections. VOIP software manage call routing to ensure that the recipient will receive contact. These types of software can be installed on the phones and computers and PDA. Usage of this software on the end user devices is one of the advantages of VOIP. In order to use VOIP calls, we will need to a VOIP server (ITSP). There are many types of servers such as traditional telephone carriers companies such as BT and special servers VONAGE and SKYPE. Some VOIP providers support calls only


from computer to computer meanwhile, other providers provide send and receive calls from devices which have enabled IP address for consumer of traditional phone network and the mobile network.


There are several ways to implement this service that are as follows ([15-17]). 5.1 PC to PC

In this way, both the caller must be the computer or device able to execute VOIP applications commands such as PDA .And both sides have to be connected to internet at the moment of contact. In this way This IP address must be identical for both. In PC to PC way both directly communicate with other through a computer (Heads phone) which currently use Internet-based voice applications; see Fig. 1.

Figure 1: VOIP communications (PC to PC) 5.2 Phone to Phone (over IP Call)

In this way both are subscriber of PSTN services and use traditional phones for communication. There are two approaches to this type of communication;

(i) This method use from gateway. In this case, the caller uses the phone, but the call deliver through the gateway to management IP network and at side the recipient again convert to first state by a gateway( IP network can be fixed or wireless ).

(ii) In this case callers can use adapter that will have function similar to modem. In this situation the person connects by your phone that is connected to an adapter, the adapter sends the call to the PSTN but here calls are driven via the ISP and Internet lines. At side of recipient the calls receive via the Internet and replay to adapter which adapter sends it to the phone that is connected to adapter. In this way, both sides of the conversation must be an ISP subscriber and access software must be installed on the adapter (both sides should use the same adapter; (see Fig. 2).


5.5 IP Phone to Phone

In this way, a caller use the IP phone and calls would be transferred by gateway from the IP network to a PSTN telephone network, which it is in fact the other side of conversation and use the regular phone.


As previously mentioned PSTN phone infrastructure was built about a hundred years ago and then developed to a reliable voice communication system. The VOIP technology is relatively young and its architecture is new that have implemented on a network that are inherently unreliable. The purpose of this part is check of the underlying protocol and technologies that are used in VOIP. It also discusses the potential and inherent challenges that exist in its deployment.

Like other communication system VOIP is structured by two factors; (i) Bearer (the actual sound that is sent over the network). 5.3 IP Phone to IP Phone

In this way callers use from IP Phone (VOIP Phone) in order to the call will be transferred via IP networks and will not need to install the software or use the adapter or Media gateway. The Fig. 3 show that how to make calls using IP Phone and Media gateway.

Figure 3: VOIP Communications (IP Phone to IP Phone)

Figure 4: VOIP Communications (PC to Phone) 5.4 PC to Phone

This method is actually a combination of the two previous methods. So when someone uses the computer (headphone) to call ,calls are transmitted via the Internet .Thus an (ITSP) receives call and using the gateway deliver call to the nearest point on the PSTN network and to phone devices (Fig. 4).


(ii) Signaling (necessary massage to control and management other components of call such as the digits dialed to destination). Now, we consider the protocol used in VOIP; 6.1 Components of VOIP

The Public Switched Telephone Network (PSTN) is the collection of all the switching and networking equipment that belongs to the carriers that are involved in providing telephone service. In this context, the PSTN is primarily the wired telephone network and its access points to wireless networks, such as cellular. The overall technology requirements of an Internet Protocol (IP) telephony solution can be split into four categories: signaling, encoding, transport and gateway control.

The purpose of the signaling protocol is to create and manage connections between endpoints, as well as the calls themselves. Next, when the conversation commences, the analog signal produced by the human voice needs to be encoded in a digital format suitable for transmission across an IP network. The IP network itself must then ensure that the real-time conversation is transported across the available media in a manner that produces acceptable voice quality. Finally, it may be necessary for the IP telephony system to be converted by a gateway to another format-either for interoperation with a different IP-based multimedia scheme or because the call is being placed onto the PSTN. 6.1.1 H.323

This protocol was generated in 1996 by the International Telecommunications Union Telecommunication (ITU-T) which provides protocols l for establishment Voice-video communications over a packet network. The protocol follow most of sequence messages deals that is adopted by the PSTN and has great compatibility with traditional phone systems. H.323 protocol, controls and addresses signals of call and provides the field of multimedia transmission and controls it .Furthermore, this protocol , controls bandwidth when making a conference one to one and one to few. The protocol is used by many real-time applications of the Internet such as NET MEETING and GUNGK which are use For voice and video services over IP networks; see ([18, 19]). 6.1.2 Session Initiation Protocol (SIP)

This protocol has been approved by IEIF for phone calls via IP network that applies set of operations similar to H.323 but is specifically designed for the Internet. SIP protocol is widely used for controlling communication sessions such as voice and video calls over the Internet. SIP like to HTTP service is used for create and modify and end communication sessions on the internet. Session content can be simple up to complex phone calls – a multi-sectoral and multi-media. The protocol provide services that is the combination of phone elements and internet-based applications such as e-mail, video conference, multimedia streaming ,instant messaging-file transfer and computer games; see ([20]). 6.1.3 Real-Time Transfer Protocol (RTP)

This protocol defines a standard format for distributing audio and video packets on IP network and is much used at the communication systems such as the telephone, television services, video conferencing programs and web-based talk .RTP with its control protocol (RTCP) is used which it overseen on the transfer statistic and quality of service and synchronize multiple stream. The RTP on the couple port numbers are make and receive while its control protocol uses odd ports number, it is worth noting that this protocol is base of VOIP and linked to signaling protocols that would help to set up a communication network; see ([21]).


6.1.4 Media Gateway Control Protocol (MGCP)

This is the protocol to control media gateway that role on the IP and PSTN networks and the protocol for call control, signaling of VOIP systems and PSTN, so that implement the model PSTN – Over – IP; see ([22]).

6.1.5 Session Description Protocol (SDP)

This is a standard format for define Initial parameters of data streaming media and descriptions for multimedia communication sessions; in order to achieve to objectives of meeting declaration, invitation to session and negotiate parameters. Furthermore, to negotiate on media type and format and other features related to the last point that called the session profile. SDP primarily have designed to support the development of new types of media and formats; see ([23]).

6.1.6 Codec

Before the voice transmission over the IP network, People’s voice should be coded and convert into digital form. Also the data compression can be used to saving bandwidth. The recipient’s side these operations should be done to reverse. Some encryption algorithms that have been standardized by the ITU are known as the G Series, which includes G. 711 which is widely used in the telecommunications industry in the traditional telephone networks and G.729.

These algorithms are different in the sampling and compression, resulting in has considered different bandwidth for each of the coding. For example, G.711 will be required the bandwidth equal 64Kbps and 8kbps for G729. It should be noted that the coding algorithms evaluation criteria are the required bandwidth and quality of the received Sound; see ([24]).

6.2 Some Factors that Should be Considered in Implementing VOIP

H.323, the global standard for packet-based multimedia communication like VoIP, provides telephone functionality that is comparable to the public switched telephone network. But the other key requirement for successful VOIP communications is quality of service (QoS). Voice communications requires networks with very low latency, low jitter, and minimal packet loss. Two factors drive these QoS requirements:

• Very high user expectations

• The technical requirements of real-time voice communications

Telephone users have very high expectations because they are accustomed to the QoS provided by the PSTN and private PBX based networks. These connection-oriented, circuit-switched networks provide each user with dedicated bandwidth for the duration of each call. The result is extremely low latency and jitter, and minimal disruption due to “noise” on the connections. Low latency allows users to carry on natural conversations. Users differ in their delay tolerance, but a good rule of thumb is to limit one-way delay to about 150 milliseconds (ms). This delay budget includes the processing delays introduced by the end systems plus the latency of the network.

Furthermore, a key consideration with VOIP is the real-time nature of voice. When voice is transferred as data, problems on the network will immediately affect the quality and reliability of the call and these can lead to callers missing part of the conversation, having echoes on the line, or getting poor sound quality – the so-called ‘Dalek effect’.


According to [25-26] the main technical issues for voice services over an IP network are: (i) Latency (delays in packet delivery).

(ii) Jitter (caused by variations in the delay of packet delivery; i.e. variations in the latency). (iii) Packet loss (packets are lost during transmission or simply arrive too late to be used. Alternatively, the network actually ‘drops’ packets during periods of network congestion.). 6.2.1 Latency

Latency is the most important factor and define as “the delays that occurs, during the process of voice packets transition over the network” . Latency contains the two types of delay; Propagation delay: the required time which packets to pass through copper wires or fiber optic.Handling delay: Include the process of digitization, locating the data in packets and passing through the hardware such as router. Delay times at the network are affected by variables among network traffic, size of packet and number of routers and gateways that packets must pass through them.When the delay exceeds from 150 milliseconds the normal flow of conversation will be interrupted.

6.2. 2 Jitter

Latency is more important than this subject. However, when consecutive packets do not receive in the same interval, distortion or deformation Problems will be created that can be appeared the form of Non-clear sound; see [25].

6.2.3 Packet Loss

Packet loss occurs when packets are lost during transmission or simply arrive too late to be used. Transmission of data (such as a webpage) makes use of the TCP/IP protocol suite which allows for retransmission of missing packets, but VoIP, which uses UDP, does not allow retransmission and the missing packets are simply left out of the call. Such loss causes voice clipping and SKYPS. This is less of a problem than latency or jitter, since the coders/decoders (codecs) used in voice processing can cope with a certain amount: up to 1% is usually undetectable, more than 3% is the maximum permitted within industry standards whilst 10% or more will not be tolerated by the listener ([26]).


Many institutions are already using of VOIP within their overall telecoms and data networking infrastructures and policies. Although many of these institutions are developing their voice services in an independent manner. In this paper we studied the basic VOIP features, including the Implementation Issues, Implementation and the various protocols required in the implementation of VoIP. We then discuss the Some factors that should be considered in implementing VOIP. The future work could be a detailed study on the Protocol Architecture of VOIP.


[1] Ascarto R., VOIP Vendors Sketch Industry Future, Computer Business Review (Online), (2005), Available Online at: http://www.cbronline.com/article_news.asp?guid =C0 8AC06B-4 296-4DA1-9AA3-8265063339AB [Last Accessed 14/09/06].

[2] Baset S. A , and Schulzrinne H., An Analysis of the Skype Peer-to-Peer Internet Telephony Protocol, Columbia University, New York, (15 September 2004), Available at: http://www1.cs.columbia.edu/ ~library/TR-repository/reports/reports-2004/cucs-039-04.pdf [Last Accessed 14/09/06].


[3] BECTA, What is Voice over IP? Technical Paper, British Educational Communications and Technology Agency, Coventry, (20 04), Available at: http://www.becta.org.uk/subsections/foi/d ocuments/ technology_and_education_research/voice_over_i p.doc [Last Accessed 14/09/06].

[4] Blackwell G., Skype: Big Bad Wolf? Part 1, VOIP Planet, (2005) Online at: http://www.voipplanet.com/ trends/article.php/3567391 [Last Accessed 14/09/06].

[5] CISCO, Brunel Wins Top Marks with Cisco IPT Solution, Case Study, Cisco, (2005), Available at: http://www.computerweeklyms.com/research/CiscoIPC/Brunel%20university.pd f [Last Accessed 14/09/06].

[6] Card D., Polin L., Parra J., Rhoads J. B., and Sartori T., Can You Hear me Now: The Return of Voice to Distance Learning, From Proceedings, Web-Based Education ,WBE 2006. ACTA Press, Calgary, Canada. January 2006, Available for Purchase at: http://www.actapress.com/PaperInfo.aspx?PaperID=22558 [Last Accessed 14/09/06].

[7] Chong H., and Matthews H., Comparative Analysis of Traditional Telephone and Voice-Over-Internet Protocol (VOIP) Systems, IEEE International Symposium on Electronics and the Environment, (May 2004).

[8] Danielsen P., The Promise of a Voice-Enabled Web, Computer, Vol. 33, Issue 8, IEEE, (August 2000). [9] Forman J., Distance Learning and Synchronous Interaction, The Technology Source, University of North Carolina, (July/August 2003), Available at: http://technologysource.org/article/distance_learning_ and_synchronous_interaction/ [Last Accessed 15/09/06].

[10] Goode B., Voice Over Internet Protocol, Proceedings of the IEEE, Vol. 90, No. 9, September 2002. [11] Gradwell P., VOIP Discussion at the BCS Specialist Internet Group, British Computer Society: London,

(February 2006), Available at: http://www.bcs.org/server.php?show=ConWebDoc.3556 [Last Accessed 13/09/06].

[12] INTEL, Quality Campus VOIP: An Intel® Case Study, Intel Technology Journal, Vol. 10, Issue 1, (15th Feb. 2006), Available Online at: http://www.intel.com/technology/itj/2006/volume10issue01/

art04_quality_campus_voip/p02_intro.htm [Last Accessed 15/09/06].

[13] Kapoor T., SIP: A Key to Convergence in the Enterprise, SIP Magazine (Online), (March 2006), Available at: http://www.tmcnet.com/scripts/printpage.aspx?PagePrint=http%3A%2F%2Fwww.tmcnet.com% 2Fsip%2F0306%2Fsip-feature-articleenterprise-0306.htm [Last Accessed 15/09/06].

[14] Jones G., Two VOIP Services Challenge Skype, Internet Week, CMP: San Francisco, (2005), Available at: http://internetweek.cmp.com/handson/174904417 [Last Accessed 15/09/06].

[15] Mehta P., and Udani S., Voice over IP: Sounding Good on the Internet, IEEE Potentials, (October/ November 2001).

[16] Mohamed A., Converged Networks: The VOIP Revolution, Computer Weekly, (September 2006). [17] Nooning T., VOIP Best Practices, ZDNet Australia, (November 2005) Available online at: http://

www.zdnet.com.au/insight/communications/soa/voip_best_practices/0,139023754,139220994,00.htm [Last Accessed 15/09/06].

[18] OFCOM, Regulation of VOIP Services, (February 2006), Available at: http://www.ofcom.org.uk/consult/ condocs/voip regulation/ [Last Accessed 15/09/06].

[19] PACKTIZER, H.323 versus SIP: A Comparison. Packetizer Website, (2006), Available at: http:// www.packetizer.com/voip/h323_vs_sip/ [Last Accessed 15/09/06].

[20] Robert C., Voice over IP, Centre for Critical Infrastructure Protection: New Zealand, (2005), Available online at: http://www.ccip.govt.nz/ccip-publications/ccipreports/voice_over_ip.pdf#search=%22voice %20over%20ip%20site%3Accip.govt.nz%22 [Last Accessed 15/09/06].

[21] Schulzrinne H., (Ed.), RPIDS – Rich Presence Information Data Format for Presence Based on the Session Initiation Protocol, Draft IETF Working Group Memo, Columbia University, New York.


(21 February 2003). Available at: http://www.cs.columbia.edu/sip/drafts/simple/draftschulzrinne-simple-rpids-02.pdf. [Last Accessed 15/09/06].

[22] Sherburne P., and Fitzgerald C., You Don’t Know Jack About VOIP, QUEUE, (September 2004). [23] Street S., IP Telephony: The End of the World as We Know It? Cause/Effect, Vol. 22, No. 2. EduCause,

(1999), Available at: http://www.educause.edu/ir/library/html/cem/cem99/cem992.html [Last Accessed 15/09/06].

[24] TELENDUS, New IP Network Enhances Learning Experience at New College Durham, Telindus Public Sector Case Study, (2005) Available at: http://www.telindus.co.uk/resources/telindus_newcollegedurham.pdf [Last Accessed 15/09/06].

[25] Varshney U., Snow A., McGivern M., and Howard C., Voice over IP, Communications of the ACM, ACM Press, (January 2002), Vol. 45, No. 1, http://www.jisc.ac.uk/index.cfm?name=techwatch_ report_0207 [last Accessed 15/09/06].

[26] TELENDUS. New IP Network Enhances Learning Experience at New College Durham, Telindus Public Sector Case Study, (20 05 ), Available at: h ttp://www.telin d u s.co .uk/resou rces/telind us_ newcollegedurham.pdf [Last Accessed 15/09/06].





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