Voice over ip interoperability thesis


With step towards the fourth generation communication networks, integrated networks are coming into operation. Also voice over IP is expected to be a low cost communication medium. The voice codecs are big constraints which affect the quality of the voice in a network.

Hence, before real time deployment of VoIP over a network it is necessary to evaluate the voice performance over varying networks for various codecs. WLANs [1] are mostly designed for private wired LANs and have been enormously successful for data traffic but voice traffic differs fundamentally from data traffic in its sensitivity to delay and loss [2].

The IEEE Incapability of providing differentiation and prioritization based upon traffic type results in providing satisfactory performance for best-effort traffic only, but inferior support for QoS requirements posed by real time traffic. The PCF mode enables the polled stations to transmit data without contending for the channel. This results in excessive delay and poor performance of VoIP 2 Unlike a typical IEEE IEEE It supports real time applications like VoIP or streaming applications but wastes bandwidth during the off periods.

This requires variable-size data grants at a minimum guaranteed rate. The nrtPS is similar to the rtPS but allows contention based polling. Data streams, such as Web browsing, that do not require a minimum service-level guarantee is supported by BE service. BE connections are never polled but receive resources through contention.


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Since, WiMAX is expected to create the opportunity to successfully penetrate the commercial barrier by providing higher bandwidth, establishing wireless commons becomes an important factor. Also, bandwidth crunch and network integration are some of the major technical and social challenges regarding the future of the community-based Wi-Fi networks [10]. Instead of global Internet connectivity, many current applications and businesses are expected to be better utilized by using the localized Wi-Fi constellation.

With a step towards the next generation, it is expected that an integrated network as shown in Figure 1. VoIP has been widely accepted for its cost effectiveness and easy implementation. A VoIP system consists of three indispensable components, namely 1 codec, 2 packetizer, and 3 playout buffer. Analog voice signals are compressed, and encoded into digital voice streams by the codecs. The output digital voice streams are then packed into constant-bit-rate CBR voice packets by the packetizer.

A two way conversation is very sensitive to packet delay jitter but can tolerate certain degree of packet loss. Hence a playout buffer is used at the receiver end to smooth the speech by removing the delay jitter. Perceived voice with zero jitter, high MOS and low packet end-to-end delay is considered to be the best.

This process is called encoding and the converse is called decoding and both are performed by voice codecs [11]. With bandwidth utilization becoming a huge concern, voice compression techniques are used [11] to reduce bandwidth consumption. Voice compression by a codec adds an additional overhead of 4 Thus, a codec is expected to provide good voice quality even after compression, with minimum delay. Table 1. It has a fixed bit rate of 64kbps.

For example, during a call using G. So, G. TABLE 1. Packet Creation based on voice activity. Moreover, recently voice codecs are developed to detect talk-spurt [15] and silence lengths [15] within a conversation. Silence in a communication period leads to packetization of the background noise and sending it over the network. This causes bandwidth wastage. With silence suppression during the silence period, the codec does not send data as shown in Figure 1.

This decreases channel utilisation and thereby saves bandwidth. Voice communication is noise sensitive. Noise causes the signal to reach the destination with a lead or lag in the time period. This deviation is called jitter.

Lead causes negative jitter and lag causes positive jitter and both degrade the voice quality. The time taken by voice to be transmitted from the mouth of the sender to the ear of the receiver is called packet end-to-end delay. The packet end-to-end delay should be very less for voice communication. Perceived voice quality is typically estimated by the subjective mean opinion score MOS , an arithmetic average of opinion score. MOS of a particular codec is the average mark given by a panel of auditors listening to several recorded samples.

It ranges from 1 unacceptable to 5 excellent. It depends on delay and packet dropped by the network.

The E-model, an analytical model defined in ITU-T recommendation, provides a framework for an objective on-line quality estimation based on network performance measurements like delay and loss and application level factors like low bit rate codecs. The result of the E-model is the calculation of the R-factor best case worst case 0 [5].

The advantage factor A compensates for the above impairments under various user conditions. A is 10 for mobile telephony but 0 for VoIP [5]. R0 is considered to be Typically originating and terminating parts would respectively be the in-house portions of an enterprise network. The part lying in Internet or some public WAN i.

The packet flow is assumed to be from left to right. For flows in the opposite direction, the terminology of originating and terminating is interchanged. Unless stated otherwise, in view of delay calculations, originating and termination parts are regarded to be identical. On the originating side, the analog voice signal is digitized into pulse code modulation PCM signals by the voice codecs. Then the PCM samples are compressed and converted into packet format, thus ready to be sent across the net. For some network configurations, the edge router may also perform codec and compression functions.

In the subsections below, we attempt to describe various kinds of delays together with their associated formulation in units of milliseconds ms.


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The block size used by the G. The present packet is checked while the previous one is being compressed. Note that such an operation is not possible at the very beginning of transmission.

Introduction to Voice Over IP

Thus, a fixed delay of 5 ms, called look-ahead delay, occurs in this section of packet voice transmission [16]-[17]. This type of codec has an 8 kbps modulation rate, hence producing a 10 ms delay for encoding the packet [17]-[24].

To support a good quality voice call, the packetization delay should be less than 30 ms [17]. The packetization process commences after storing the packets into a buffer. As long as the buffering time remains below 10 ms, the compression and buffering periods will overlap and there will be no additional delays introduced at the buffer [17]. Otherwise, the buffering time would exceed the compression time, and the remaining period would add to the total delay created during the buffering operation.

The sources of these delays are explained below: o Switching Delay: The packets wait for 10 ms at each switch in the originating network and 1 ms in the core network [17]. The waiting time of each packet will be 8 Also, these queuing delays will also vary depending on the transmitted IP packet size. Because of these complications, data queuing delay is not easy to formulate.


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voice over ip interoperability thesis Voice over ip interoperability thesis
voice over ip interoperability thesis Voice over ip interoperability thesis
voice over ip interoperability thesis Voice over ip interoperability thesis
voice over ip interoperability thesis Voice over ip interoperability thesis
voice over ip interoperability thesis Voice over ip interoperability thesis
voice over ip interoperability thesis Voice over ip interoperability thesis
voice over ip interoperability thesis Voice over ip interoperability thesis

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