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A survey of voice over internet protocol (VOIP) technology

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IJCMSA: Vol. 6, No. 3-4, July-December 2012, pp. 53– 62 © Serials Publications
ISSN: 0973-6786

A Survey of Voice Over Internet Protocol (VOIP) Technology

J. POURGHASEM1, S. KARIMI1 AND S. A. EDALATPANAH2, 3
1
Department of Computer Science, Ramsar Branch, Payam Noor University (PNU), Ramsar, Iran,
E-mail: Jpourmail@gmail.com
2
Department of Applied Mathematics, Lahijan Branch, Islamic Azad University, Lahijan, Iran,
E-mail: skarimimail@gmail.com
3
Young 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).

1. INTRODUCTION
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.

2. WHAT IS THE VOIP?

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
54 J. Pourghasem, S. Karimi and S. A. Edalatpanah

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.
A Survey of Voice Over Internet Protocol (VOIP) Technology 55

3. BACKGROUND AND HISTORY OF VOIP

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.

4. HOW DOES VOIP WORKS?

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
56 J. Pourghasem, S. Karimi and S. A. Edalatpanah

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.

5. DIFFERENT TYPES OF VOIP COMMUNICATIONS

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).

Figure 2: VOIP Communications (Phone to Phone)

A Survey of Voice Over Internet Protocol (VOIP) Technology 57

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)

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).

Figure 4: VOIP Communications (PC to Phone)

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.

6. STANDARDS – CHALLENGES – PROTOCOLS

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).
58 J. Pourghasem, S. Karimi and S. A. Edalatpanah

(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]).
A Survey of Voice Over Internet Protocol (VOIP) Technology 59

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’.
60 J. Pourghasem, S. Karimi and S. A. Edalatpanah

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]).

7. CONCLUSION
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.

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