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Voice over Internet Protocol Applications

Introduction:

Voice over Internet Protocol (VoIP) is one of the fastest growing applications for the internet today. It enables people to use the internet as the transmission medium for telephone calls by sending voice data in packets using IP rather than by traditional PSTN (Public Switched Telephone Network).

For QoS in VoIP, the factor which affects the quality of service are delay, delay variation, packet loss, QoS control mechanism, network security and reliability, providing bandwidth, voice compression, echo suppression and jitter on the perceived conversational quality.

We discuss here mainly factor affecting quality of voice. These are

• Packet loss (when some packets never reach their destination, which can cause dropped calls)

• Latency (when voice packets are delayed)

• Jitter (when packets are sent and received, but with timing variations)

We also discuss some points on VoIP over wireless link and some of the benefits of VoIP such as cost reduction, flexibility, advance features and simplified infrastructure.

Basic end to end connection in VoIP:

VoIP refers to communications service that is transported over Internet, rather than the public switched telephone network. The basic steps involved in originating an Internet telephone call are conversion of the analog voice signal to digital format and compression/translation of the signal into Internet protocol (IP) packets for transmission over the Internet; the process is reversed at the receiving end.

Fig. 1 shows a business in which a PBX is connected to VoIP gateway as well as to the local telephone company central office. The VoIP gateway allows telephone calls to be completed through the IP network. Local calls can still be completed through the telephone company as in the past. The business may use the IP network to make all calls between its VoIP gateway connected sites or it may choose to split the traffic between the IP network and the

PSTN based on a least-cost routing algorithm configured in the PBX.

An alternative VoIP implementation uses IP phones and does not rely on a standard PBX. As in Fig. 2 IP phones are connected to a LAN. Voice calls can be made locally over the LAN. The IP phones include codec that digitize and encode (as well as decode) the speech. Calls between different sites can be made over the wide area IP network. Proxy servers perform IP phone registration and coordinate call signaling, especially between sites. Connections to the PSTN can be made through VoIP gateways.

FACTOR AFFECTING VOIP SERVICE

A large number of factors are involved in making a high-quality VoIP call. These factors include the speech codec, packetization, packet loss, delay, delay variation, and the network architecture to provide QoS. Here we discuss only packet loss, delay -which leads to Echo and jitter.

Standards bodies like ITU are continuously addressing issue of voice quality and how to measure voice quality, and have already derived two important recommendations: P.800 (MOS) and P.861 (PSQM). A method Mean Opinion Score (MOS) of voice quality involves recording several pre-selected voice samples over the desired transmission media and then playing them back to a mixed group of men and women under controlled conditions. The scores given by this group are then weighed to give a single MOS score ranging between 1 (worst) and 5 (best). A MOS of 4 is considered "toll-quality" voice.

P.861 Perceptual Speech Quality Measurement (PSQM) tries to automate this process by defining an algorithm through which a computer can derive scores that have a close correlation to the MOS scores

Packet loss:

Packet Loss may occur when connection speeds are compromised by temporary ISP problems, network congestion, or heavy bandwidth usage. Packet loss starts to be a real problem when the percentage of the lost packets exceeds a certain threshold (roughly 5% of the packets), or when packet losses are grouped together in large packet bursts. In those situations, even the best CODECs will be unable to hide the packet loss from the user, resulting in degraded voice quality. Thus, it is important to know both the percentage of lost packets, as well as whether these losses are grouped into packet bursts.

Fig. 3 shows some MOS results from Voice Quality Assessment

(VQA) laboratory.

The data are collapsed over a variety of PLC algorithms and are based on the G.711 (64 kbps) coder and a 20 ms packet size. Considering all the qualifying factors, we believe that VoIP networks must hold packet loss below 1 percent in order to deliver PSTN equivalent voice quality. Because the voice signal is sent as packets on a VoIPnetwork, they may travel different routes to get to destination. At the receiver a packet might arrive very late, corrupted or simply might not arrive. Packet Loss Concealment(PLC) is a technique to mask the effects ofpacket lossinVoIPcommunications

Latency:

The latency is also known as delay VoIP delay or latency is characterized as the amount of time it takes for speech to exit the speaker's mouth and reach the listener's ear.

Three types of delay are inherent in today's telephony networks: propagation delay, serialization delay, and handling delay. Propagation delay is caused by the length a signal must travel via light in fiber or electrical impulse in copper-based networks. Handling delay—also called processing delay—defines many different causes of delay (actual packetization, compression, and packet switching) and is caused by devices that forward the frame through the network. Serialization delay is the amount of time it takes to actually place a bit or byte onto an interface.

150 mSec is specified in ITU-T G.114 recommendation as the maximum desired one-way latency to achieve high-quality voice. Beyond that round-trip latency, callers start feeling uneasy holding a two-way conversation and usually end up talking over each other. The graph in Fig 4 shows how for a specific network configuration the quality degrades with increasing delay. The graph also shows how this degradation is further affected by the presence of echo.

When there is an enough delay which result in an echo that is loud enough to be perceptible to the speaker (usually around 30 ms and above) that the quality of a voice call becomes problematic. There are two possibilities to avoid or solve this annoying problem by echo suppression and echo cancellation.

Jitter:

Jitter is a variation in packet transit delay caused by queuing, contention and serialization effects on the path through the network. In general, higher levels of jitter are more likely to occur on either slow or heavily congested links. If packets do not arrive in precise time interval result in a poor voice quality. A jitter buffer is designed to remove the effects of jitter from the decoded voice stream, buffering each arriving packet for a short interval before playing it out. This substitute additional delay and packet loss (discarded late packets) for jitter. A fixed jitter buffer maintains a constant size whereas an adaptive jitter buffer has the capability of adjusting its size dynamically in order to optimize the delay/discard tradeoff. It is expected that the increasing use of “QoS” control mechanisms such as class based queuing, bandwidth reservation and of higher speed links such as 100 Mbps Ethernet, E3/T3 and SDH will reduce the incidence of jitter related problems at some stage in the future, however jitter will remain a problem for some time to come.

VoIP Calls over Wireless Links Using H.323 Protocol

Currently, the voice and data traffic are treated separately by the GSM (Global Special Mobile) and GPRS (General Packet Radio Service) wireless access technologies. To converge the voice and data networks, the signaling for VoIP should also be able to migrate into the existing infrastructure. The International Telecommunications Union has defined H.323 as the key protocol that implements this migration to today's PSTN signaling domain. It providesaudio-visualcommunication sessions on any packet network. The H.323 standard addresses call signaling and control, multimedia transport and control, and bandwidth control for point-to-point and multi-point conferences. To overcome some limitations of H.323, additional new protocols have been proposed recently to implement VoIP services through gateway decomposition. They include SIP (Session Initiation Protocol), MGCP (Media Gateway Control Protocol) and H.248. However FER (Frame Error Rate) is quite high in air-link condition due to that VoIP call set-up performance can degrade significantly. However there are important research works going on to make wireless VoIP services successful.

BENEFITS:

There are several benefits to using this technology. Such as

1. Cost Reduction: VoIP exchange is based on software rather than hardware, it is easier to alter and maintain. It is easily scalable and organization no longer need separate cabling for your telephone system.

2. Flexibility: You're able to take your VoIP adapters anywhere and use your number anywhere, require just an Internet connection.

3. Advance Features: It has ability to integrate your computer applications such as email, fax, web conferencing and video phone with your telephone. It also offer features such as Voice Mail, Call Forwarding, Call Waiting, Caller ID, Call Block, Call Return and Do Not Disturb.

Conclusion:

There are many technical challenges and factors that need to be resolved for its successful commercial deployment. Significant steps have been taken over the last few years to achieve higher quality systems such as improvements in codec, compression method, different protocols use for prioritize traffic and reliability. Again it depends on what users actually experience and hence how these choices impact on the perceived quality of a solution. Although their benefits overweigh its quality issues.

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