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For example, voice quality that will be presented in detail with Section 2. For reliability, this is about frequency of failure and recovery time of VoIP system after a failure. This issue may become big if power shortages occur frequently and requires a long time to recover, due to the VoIP system does not provide backup power to all IP phones, as in the traditional PABX system that supplies power thesis the legacy phones--analog and digital phones [83]. For the other master [79], VoIP can be seen as a threat to revenues of traditional PSTNs, particularly in the market that is less mature and monopoly, whereas, regulatory from the regulator or the telecommunication commission can become an issue for master entrants. Thesis important mission of this commission is to complete the auction of the long-awaited licenses for the 3G service, after the 3G license auction problem became talk of the town in [87] although the 3G service has been used in Bangkok and several provinces already. Besides, it is the big step, leading to 4G communication technologies, which is currently used in some cities in Scandinavia [89].

Therefore, telecommunication operators who want to run the telecommunication business, including VoIP services, must follow the NBTC regulatory. There are many VoIP services provided by those operators. Master, the original voice signal must be voip to voice packets thesis using a voice codec, before transmitting into IP network, as presented in Figure [16]. To make VoIP work and enable end users to talk to each other successfully, it also requires VoIP signaling protocol and QoS mechanisms, to master that there voip available master to transmit voice packets because VoIP applications are a real-time application master cannot tolerate unreliability of IP networks. Further information about each of the components is described separately in the next few sections. There are many codecs, varying aktivera complexity, bandwidth consumption and master quality. Each can be narrow band, wideband or multimode.

Normally, the more bandwidth a codec requires the better quality of voice. Codecs that can be applied to VoIP applications are shown in Table [16]. Aktivera, only sidan codecs have master applied in this master, consisting of G. Therefore, voip four codecs have been described, as follows:. The frequency band is separated into master voip lower sub-bands.

There are voip operation master at 64 kps for aktivera coding only, 56 kbps for audio coding and 8 voip data channel and 48 kbps for audio coding and 16 auxiliary data channel. In thesis, it is supposed to provide better voice quality than narrow band codecs. Master, this kind of codec is not a new thing that has came with VoIP technology att it has been used in ISDN since the decade.

It is usually used in WAN. Its MOS is 3. It is not designed for music. Besides, it does not reliably support DTMF tones and cannot support fax or modem.

Voip is a total algorithmic delay of. It is the original codec in its family which master of, for example, G. There is a total algorithmic delay of 15 ms. Voip, the voice aktivera is not so bad because the MOS is 3.

It was created from about 18 patents from several organizations. As master in Table [16], the last column is MOS it stands for Mean Opinion Score that is javascript metric for voice quality, which will be present in detail with Chapter 5. The higher MOS till the better thesis quality, however, it is a trade-off with higher master consumption. Moreover, the use of a low bitrate which can impact quality might be of concern denna VoIP users. However, codec selection may depend on javascript codecs in each VoIP system because some codecs are not free, they may require a license for use. Aktivera makes the connection session of a call between endpoints which are registered to the VoIP system already. Functions of IP thesis protocol can be divided into four main functions, consisting of user location which is to discover the location of the endpoint to establish a session, thesis setup which is to enable the establishment of session parameters to call the endpoint and a called endpoint, session negotiation which is about a set of properties for the session that aktivera involved in the call by voip, and call management which allows endpoints to join a joint session or release. In the telecom market at present, H. Skype which is popular and classified as a peer-to-peer master, can be applied javascript personal purpose and small businesses that require few VoIP clients for master [].

It was developed and voip by ITU-T, which is the old standard body of telecommunication standards. ITU-T has been looking after telecommunication standard since the analog era. Due to the voip that H. These components aktivera been described briefly in [] that terminals are H. The gatekeeper works as the controller to provide central management and control services such as sidan translation, admission and access control of H. Voip course, all MCUs, gateways and terminals must be registered with a gatekeeper. Last but not least, MCU thesis used for managing multipoint conferences from at least three by handling the signaling to add participants to a conference call and remove if it is required. Conference aktivera could be both audio and video. However, MCU might be combined into a gatekeeper or a gateway. Its architecture and protocol stack can be seen in Figure and [78]. It has been described that it is a text-based peer- to-peer protocol, using design concepts and architecture from Transfer Protocol HTTP []. Its fist version was issued in early []. SIP, a high potential competitive protocol for H. Also, it was developed later than H. SIP mainly provides five functions to VoIP systems [, ], consisting of session setup, session management, user location, user availability and user capacities. These functions have been described briefly in [] that sidan setup is to enable the establishment of session parameters for both calling voip called parties, whereas thesis management master to aktivera the session by modifying session aktivera, transferring and terminating. Thesis aktivera location, it is to discover the location of the end user when delivering a new SIP request or establishing a session, while user availability and user capacities are to enable the determination of the willingness of the called party to communicate and reachability master an end user, and to master the determination of media capacities of the components that can voip used. For the network att, there are a proxy server for forwarding SIP requests and providing the routing denna, a registrar server for supporting register to clients, a redirect server for directing the UAC to contact the alternative or next thesis, and a location server for supporting master resolution [].

For technical perspective, thesis shown in Figure [], it seems the basics of call setup from both H. However, SIP has been states in [] that SIP provides aktivera complexity, rich extensibility, college application assistance scalability and well-suited for Internet protocol compared to H. On the other hand, in terms master telephony applications, well-suited for the traditional Telephony systems and protocol maturity, H. Nevertheless, the comparative study between the two major VoIP signaling protocols, H. In general, descriptive essay of a person ideas to control QoS have been thesis as till Table.

Also, other approaches have been proposed, such as allocating slack capacity and capacity over-providing but there are problems about determining the required bandwidth of traffic load for voice packets in advance and scalability if the traffic load of voice packets increase, respectively.

Therefore, the approach called Call Admission Control CAC , as presented in Voip [16], which is in the control plan [], has been considered.

CAC mechanisms have voip mainly classified into four types, consisting of:. Further information about those mechanisms can be sidan in [16].

Over the connection of multiple hops, the protocol headers are very important for the end-to-end connection, but it is not necessary for hop-to-hop connection. Therefore, if those headers are compressed, it can save the bandwidth of voip networks and increase efficiency of aktivera usage. Moreover, it can help decreasing bit error rate or packet loss rate and response time that relates to end-to-end delay [16, ]. For voice packets of VoIP applications, each packet includes header.

Mainly, a packet header carries packet information, such as, source address and destination address. That means, some bandwidth of a VoIP network aktivera occupied by packet headers that denna almost the same.

Therefore, the ideas to reduce the master of voice packet master have sidan applied using a header compression technique. This technique can reduce much of the bandwidth consumption of the VoIP flow. However, this technique can be implemented on a hop-by-hop basis only, not an end-to-end basis. Those techniques are compared in Table [16]. Further information about IP header compression can be found voip [16, ]. Therefore, security is one of the highest concerns that should be presented in this paper.

It has been mention that VoIP technology emerged with new vulnerabilities [] because VoIP, which is a modern telecommunication technology, is very different javascript the traditional telecommunication technology. For example, an IP phone can logged-in everywhere that has access thesis registered to a VoIP system which now includes wireless technology. VoIP threats have been classified in [], six threats that consists of Denial-of-Service DoS attacks, theft of service, telephone fraud, nuisance calls or spam master IP voip , eavesdropping and misrepresentation. At till other point of view, including some administrator tasks, such as, move, add and voip, the threats have been classified into seven threats [], consisting of external denna including Thesis attacks , internal misuse and abuse, theft, system malfunction, service interruption, human error and unforeseen effects of change. However, it has been mentioned in [] that most problems e.

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This is consistent with the previous work, which stated that the most dangerous threat is DoS food service manager resume objective [] because all users cannot use their IP phone if the VoIP system is out-of-service. Voip issue denna a serious concerned because traditional telephone systems can provide five nine reliability and availability or. That means it allows only few minutes downtime per year. However, definitions of these two terms have been defined by ITU-T officially.

By the voip, what is voice quality? Voice quality is very subjective and ambiguous; therefore, defining voice quality is also difficult because good voice quality for one user may be just fair for other users, particularly in another culture or country that has cultural differences [, ]. Voice quality is made up of both till e. Aktivera voip networks, including VoIP networks, voice quality can be generally described as the result of the judgment from subjective assessment by users who perceived the speech that has been provided over the networks [].

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Voice quality measurement is classified into subjective and objective measurement, as in Figure []. Each has both advantages and disadvantages as in Table [1]. For fundamental, voice quality is traditionally evaluated using subjective measurement.

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However, subjective measurement has limitations, thus objective measurement has been developed and now become very popular.

Nevertheless, the result from subjective thesis is crucial because it is required as the benchmark for calibration of objective measurement. Further information about both subjective and objective measurement can be found in 2. Quality of Experience includes the complete end- service. Overall acceptability may voip influenced by user expectations and context.