Journal of Information Engineering and Applications www.iiste.org

ISSN 2224-5782 (print) ISSN 2225-0506 (online)

Vol.3, No.4, 2013

18

Voice over Internet Protocol (VOIP): Overview,

Direction And Challenges

1

U. R. ALO and

2

NWEKE HENRY FIRDAY

ABSTRACT

Voice will remain a fundamental communication media that cuts across people of all walks of life. It is therefore

important to make it cheap and affordable. To be reliable and affordable over the common Public Switched

Telephone Network, change is therefore inevitable to keep abreast with the global technological change. It is on this

basis that this paper tends to critically review this new technology VoIP, x-raying the different types. It further more

discusses in detail the VoIP system, VoIP protocols, and a comparison of different VoIP protocols. The compression

algorithm used to save network bandwidth in VoIP, advantages of VoIP and problems associated with VoIP

implementation were also critically examined. It equally discussed the trend in VoIP security and Quality of Service

challenges. It concludes by reiterating the need for a cheap, reliable and affordable means of communication that

would not only maximize cost but keep abreast with the global technological change.

internet or intranet networks. The technology does not use the traditional Public Switched Telephone Network

(PSTN); instead calls are made over an internet protocol data network. VoIP has great benefits of increased saving,

high quality voice and video streaming and several other added value services. Examples of VoIP software are:

Skype, Google talks and windows live messenger (Di Wu, 2002).

2. Overview of VoIP

VoIP stand for Voice over Internet Protocol. VoIP enables us to compress and convert voice signal to digital signal

and transmit it through Internet Protocol (IP)-enabled network like Internet, Ethernet and Wireless LAN. VoIP uses

Internet protocol to manage voice packet over Internet protocol (IP) network.

Types of VoIP Service

service: softphone software, a sound card and good Internet service. With computer to computer VoIP service, the

user may not be able to call either landline or mobile phones, and also the recipient must be online in order to call

him/her.

Computer to phone and vice versa

This is a software-based and hardware-based service. Softphone software is used to route the call to an Internet

protocol and hands off to a conventional telephone network . To use the service, one needs to subscribe and be

charged at a low rate. Examples include Skype, MSN and Google Talk that provides the service to enable their

customers to call landline from their computer. Computer to phone requirements are Internet-enabled phone and

computer, VoIP service subscription, modem and Analog Terminal Adapter to convert the call signal to digital signal

and also back to analog signal. Computer to phone does not allow emergency call users and needs to have a

Journal of Information Engineering and Applications www.iiste.org

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Vol.3, No.4, 2013

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telephone companies use this to handle long distance calls. VoIP convert the audio sound into data packets and

transfer these packets over the Internet. It allows emergence calls and does not need PSTN for initiation and

termination of calls.

2.1 VoIP system

Voice over Internet Protocol (VoIP) system is made up of End User Equipments, Network component, call

processor, Gateway and Server. The End User Equipment is the part mainly used to access the VoIP system and

communicates with other End point (Greg Tucker , 2004). End User equipments initiate and maintain the signals

needed to establish calls over the IP network. It also helps to convert voice to data packet, voice and video

communication, monitoring and network surveillance capability. They are connected to the network with cable and

responsible for providing address translation, bandwidth control, and access control and helps to improve the quality

of service and security of VoIP network. Gateway serves as an interface between public switched telephone networks

(PSTN) and VoIP network. Gateway is made up of Signalling Gateway (GS), Media Gateway (MG), Media

Controller and multipoint control unit (MCU). Signalling Gateway helps to manage signal traffic while the media

gateway manages media signal. The media gateway controller manages traffic between signalling gateway and

media gateway while multipoint control unit is the part that establishes conference calls between three or more

people for converged voice, video and data conference. It provides audio or combination of audio, video and data

conference depending on the strength of participating endpoint (Greg, 2004).

Call processor is software running on end user equipment operating system that helps in call set up, call

monitoring, user authorisation, signal coordination and bandwidth control. It translates a phone number to IP address.

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The server enables the establishment of call and support for other features in the system. The Session initiation server

allows the user to forward calls to different location in the VoIP network.

2.2 VoIP Protocols and Standard

There are many protocols and standards for VoIP network (H.323, SIP, MGCP, RTSP, RSVP, SDP etc) but only the

most commonly used ones are discussed.

H.323 protocol

H.323 protocol was developed by International Telecommunication Union and Telephony (ITU-T) in 1997 based on

Real-time Transport Control protocol (RTCP) for sending voice, video and data over IP-based network (Rakesh,

point registrations. The functions of Gatekeeper are:

• Translate alias address to transport address

• Deny or grant access based on call authorisation, source and destination address.

• Call signalling with end point terminals

• Control the number of terminals permitted at a time in H.323 protocol (bandwidth management)

• Maintain the list of ongoing call H.323 calls to determine the busy terminals for bandwidth management.

• Rejection of call from terminal due to authorisation failure in the use of H.225 signalling (Call

authorisation).

Another important component of H.323 protocol is the multipoint control unit (MCU). It acts as a bridge that enables

two or more terminals and Gateway to participate in a multipoint conferencing. MCU is made up of Multipoint

• Q.931 for all signalling, call control and setup.

• Registration Admission status (RAS) is used for communicating with H.323, endpoint and gatekeeper. It

provides interaction between H.323 and the gatekeeper.

The table 1 lists the use of protocol stacks in audio, video and data packet, and their transport protocols.

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The stages in audio connection between terminal, gateway and gatekeeper in H.323 protocol are:

• Registration of terminal and Gatekeeper registration (H255-RAS).

• Call routing between the terminal and gatekeeper (H.225-RAS and H.225-Q.931).

• Initial communication and capability exchange

• Establishing audio communication (open logical channel)(H.245)

• Audio transmission (RTP/RTCP).

H.323 follows the following steps to establish connections between clients (Mona &, Nirmala, 2002). The two

endpoints will register with the gatekeeper.

• Terminal A will start Registration admission status message exchanged by initiating call to the gatekeeper.

• The information provided by gatekeeper is used by Terminal A to contact B.

The problems of H.323 protocol are lack of flexibility, high connection setup latencies implementation difficulties.

Figure 2 and 3 shows the architecture of H.323 and the connection procedures.

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Session Initiation Protocol (SIP)

Session Initiation Protocol (SIP) was developed by Internet Engineering Task Force (IETF) designed to initiate and

terminate VoIP session with one or more participant (Rakesh, 2000), (Mona &, Nirmala, 2002). It is an ASCII-based

peer to peer application protocol that initiates, modifies, creates and terminates interactive multimedia

communication session between users. Because of the flexibility of SIP, it is used for audio, video and data packet

transmission and communication. Session Initiation Protocol is similar to hypertext transfer protocol (http) made up

SIP consists of User Agent and Network server. User Agent is the endpoint that acts on behalf of the user and maybe

client or server. The client is called the user agent client and helps to initiate SIP request while the server known as

user agent server receives the request, process it and returns the responses on behalf of the user. Network servers

include registration server, proxy server and redirect server. Registration server is used for uploading current location

of user, proxy server receives the request forward it to the next hop while redirect server on receiving request,

determines the next hop and returns the address of the next hop server to the client instead of forwarding the request.

SIP call Flow

The six steps for establishing communication in SIP protocol are:

• User registration, initiation and location.

SIP Messages

SIP defines several messages for communicating with the client and SIP server (Rakesh, 2000). Some of the

messages are listed

• INVITE- used to initiate a call by inviting user SIP session call.

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• CANCEL -to cancel a pending request by client.

• BYE- indicates termination of initiated call.

• REGISTER-provide information about user location to the SIP server.

• OPTION-used to get information about call capabilities.

• INFO-process out-of-bound information.

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protocol claim it is complex and has overhead hence it is inefficient for VoIP system. It also lacks extensibilities and

scalability required of VoIP signalling protocol. But SIP reuses most of its header field, encoding rules, error code

and authentication mechanism of hypertext transfer protocol (HTTP). Table 2 summarises the differences between

SIP and H.323 protocols

Table 2 Comparison SIP of and H.323

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2.3 VoIP compression Algorithms

Codec (compression –decompression) is used to convert analog signal (video, voice) to digital data or convert digital

data stream to analog signal. This helps to provide compression capability to save network bandwidth. There are

different audio codec available for voice application (Hersent, 2011). They include:

G.711: approved in 1965 and is the simplest way of digitizing analog signal. The algorithm uses Pulse Code

Modulation with less than 1% acceptable packet loss factor. The encoded audio stream of G.711 is 64kbit/s, so is

worst in terms of bandwidth but the best in quality among the entire scheme.

G.722 was approved in 1988 and provides higher quality digital coding at 7 KHz of audio spectrum at only 48, 56, or

G.723.1 was approved in 1995 for use in H.323 communication and UMTS 99 video cell phones. It uses frame

length of 30ms and needs a workload of 7.5ms in 64kbt/s or 5.3kbit/s operation modes. The algorithm is not

designed for music and its difficult to be used in a fax and modem signal transmission. The International

Telecommunication Union and Telephony recommends it for use in narrow band video conferencing and 3G

wireless multimedia devices.

G.726 was approved in 1990 and uses Adaptive differential Code modulation (ADPCM) techniques to encode G.711

bit stream in words of 2, 3, or 4 bits resulting in bit rate of 16, 24, 32 or 64kbits/s .

G.728 uses low delay, codec executed linear prediction (LD CELP) coding techniques with a mean opinion score

(DTMF) signalling tones reliably. Listed in table 3 are properties of common voice codec schemes.

Codec Bit Rate Payload Packets per

Seconds (pps)

Quality bandwidth Sample

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• VoIP network integrates voice and data: VoIP integrate both Voice, video and data together to save

management and operational manpower and effective use of communication links between network sites.

• Widespread availability of Internet protocol (IP): IP network is readily available all over the world, with

people having access to PC linked to internet. Furthermore, availability of gateways to/from PSTN allows

calls to use VoIP for voice and video calls (Mona & Nirmala, 2002).

• Reduce the cost of Ownership: VoIP integrates data and voice communication traffic into a single network

thereby reducing the cost of infrastructural ownership and maintenance redundancies. It brings different

network elements together such as call server, application server and client server (Bhogal et al, 2004).

• Efficient utilisation of network resource: VoIP network improves the network bandwidth efficiency and

2.5 Challenges Associated with VoIP Implementation

There are number of challenges hampering the implementation of VoIP network for everyday use . Because the

Internet was not designed to transfer real time data, network equipment drop most packets leading to queues that

cause jitter in packet transfer. Routing in IP network is time consuming compared to switching network. Delay and

packet loss in network affect quality of service (QoS). Efforts are on the way to reduce these challenges which

include the use of Reservation protocol (RSVP), design of high priority queues for real time traffic and the use of

multiprotocol label switching (MPLS) to speed up packets through routing points. Another problem of the Internet is

that it is not managed by a single centralised operator that can coordinate the flow of calls, but made up of different

networks and service providers which makes it difficult for providing real time traffics.

Traditional telephones operate on internal 48 volt internal power supply. Therefore it can still continue to work even

if there is power outage. So, they need to have backup power system using Uninterruptible power supply (UPS) and

proper power back up assessment done regularly to ensure there is sufficient power for the VoIP system.

The use of softphones software is another challenge of VoIP implementation. These softphones are vulnerable to

worms, virus, Trojan horses and spyware attack and can compromise the security of the network. Computers

connected to the Internet can be attacked if a user visits a compromised website even if he/she did not open the

website. Traditional PSTN telephone can provide emergency calls because it is tied to physical location and

emergency service provider can easily track caller’s location to the emergency dispatch office. But an individual

using VoIP network can be from anywhere around the globe, thereby making emergency call location difficult

because the caller’s location and identity can’t be traced easily. Even if the VoIP service provider can provide for

2.6 VoIP Quality of Service (QOS)

Quality of service is the ability to control and measure data transmission rate or throughput and error rate and

provides better, predictable result to selected network traffic over different underlying technologies. Voice and video