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Introduction
Companies and organizations around the world want to reduce rising communication costs. The consolidation of separate voice and data networks offers an opportunity for significant reduction in communication costs. Accordingly, the challenge of integrating voice and data networks is becoming a rising priority for network managers. Since data traffic is growing much faster than telephone traffic, a need has been identified to transport voice over data networks, as opposed to the transmission of data over voice networks. This brought about the rise for Voice over IP (VoIP). VoIP has become especially attractive given the low-cost, flat rate pricing of the public Internet. Many components will have to be designed to accommodate voice over data networks, such as the access gateways that link the data and the telephony networks among others. Applications that offer Voice over IP services will have to include a comprehensive technology set that reduces the impairments caused by sending voice over data networks that were not designed to handle it. An important factor to be considered by network designers is the problem of Quality of Service (QoS). This is because IP is a best effort service and therefore provides no guarantees on delivery and data integrity. Voice processing will need to handle greater and variable delays, jitter, and cancel echoes that will be introduced from the telephony side. It will also have to include an appropriate algorithm to mask the gaps caused by dropped packets due to congestion on the network. A protocol needs to be implemented which guarantees bandwidth for the duration of a session and also better compression technologies need to be put in place. An understanding of how to handle call set up translation for different types of networks, connections, and internetworking is essential for competent handling of every call. A Voice over IP application meets the challenges of combining legacy voice networks by allowing both voice and signaling information to be transported over IP.
Advantages and Benefits of Voice over IP
One of the main reasons and probably the most significant interest in the race to send voice over IP is the cost advantage that this process offers organizations due to the flat rate, low cost of Internet traffic. Generally the benefits of technology can be divided into three categories:
1)Cost reduction:- Although reducing long distance telephone costs is always a popular topic and provides a good reason to introduce VoIP, the actual savings over a long term are still under scrutiny and debate. These savings from lower prices are however, based on avoiding telephony access charges and settlement fees, rather than actually reducing resource costs. The sharing of equipment and operations costs across both data and voice users can also improve network efficiency, since excess bandwidth on one network can be used by the other, thereby creating economies of scale.
2)Simplification:- An integrated infrastructure that supports all forms of communication allows more standardization and reduces the total equipment complement. The economies of putting all forms of traffic over an IP based network will pull companies in this direction, simply because IP will act as the unifying agent regardless of the underlying architecture. This combined infrastructure can support dynamic bandwidth optimization and a fault tolerant design.
3)Consolidation:- People are the most significant cost elements in a network, so any opportunity to combine operations and eliminate points of failure and to consolidate accounting systems would be beneficial. In the enterprise, SNMP based management with the appropriate MIB structures can be provided for both voice and date services using VoIP. Universal use of the IP protocol for all applications will reduce complexity and provide more flexibility.
Transmitting Voice over Internet Protocol is also beneficial for the following reasons:-
1)Since the Internet is a packet switched or "connectionless" network, the individual packets of each voice signal travel over separate network paths for reassembly in the proper sequence at their ultimate destinations. This makes for a more efficient use of network resources and more reliability than the circuit switched PSTN.
2)Private voice networks require n (n-1) access links. Whereas private data networks require only ‘n’ access links.
3)Voice has per-minute distance sensitive charge, whereas data on the other hand has flat time-sensitive charges.
4)Data transmission has no 64 kbps bandwidth limitation, which means that we can provide high fidelity voice transmissions very easily.
Applications implementing VoIP
1) PSTN gateways: Interconnection of the Internet to the PSTN can be accomplished using a gateway. A PC-based telephone for example would have access to the public network by calling a gateway point close to the destination.
2) nternet-aware telephones: Ordinary telephones can be enhanced to serve as an Internet access device as well as providing normal telephony. Inter-office trunking over the corporate intranet: replacement of tie trunks between company owned PBX’s using an Intranet link would provide for economies of scale and help to consolidate network facilities.
3) Remote access from a branch office: A small office could gain access to corporate voice, data, and facsimile services using the companies Intranet services.
4) Voice calls from a mobile PC via the Internet: Calls to an office can be achieved using a multimedia PC that is connected via the Internet. One example would be using the Internet to call from a hotel instead of using expensive hotel telephones.
5) Internet call center access: Access to call center facilities via the Internet is emerging as a valuable adjunct to electronic commerce applications. Internet call center access would enable a customer who has questions about a product being offered over the Internet to access customer service agents online.
Voice Over IP Design and Development Challenges
The goal of VoIP developers is to add telephone calling capabilities (both voice transfer and signaling) to IP based networks and interconnect these to the public telephone network and to private voice networks, in such a way as to maintain current voice standards and preserve the features everyone expects from the telephone. VoIP development needs to take place in five specific areas:
1)Voice quality should be comparable to what is available using the PSTN, even over networks having variable levels of QoS.
2)The underlying IP network must meet strict performance requirements and criteria including minimizing call refusals, network latency, packet loss, and disconnects. This is required even when there is heavy congestion in the network or when resources have to be shared among multiple users.
3)Call control (the actual signaling) should be done transparent to the user in such a way that they should be unaware of what technology is actually implementing the service.
PSTN/ VoIP service internetworking (and equipment interoperability) involves gateways between the voice and data network environments.
4)System management, security, addressing, and accounting must be provided, preferably consolidated with the PSTN operation support systems.
Quality of Service Issues in IP Networks
The advantages of reduced cost and bandwidth savings of carrying voice over data networks are associated with some Quality of Service (QoS) issues unique to packet networks. Delivering quality voice signals from one point to another cannot be considered successful unless the quality of the delivered signal satisfies the recipient. Providing a level of quality that at least equals the PSTN (this is usually referred to as "toll quality voice") is viewed as a basic requirement. Although QoS usually refers to the fidelity of the transmitted voice and facsimile document it can also be applied to network availability, telephone feature availability, and scalability. Many factors have been identified that play a big role in determining the quality of service. They are as follows:
1) Delay
Two problems that result from high end-to-end delay in a voice network is echo and talk over lap. Echo is caused by signal reflections of the speaker’s voice from the far end telephone equipment back into the speaker’s ear. Echo becomes a problem when the round trip delay exceeds 50 milliseconds. Since echo is perceived as a significant quality problem, Voice over IP systems have to address the need for echo control and implement means for echo cancellation. Talkers overlap is the problem of one caller stepping on the other talker’s speech. This becomes significant if the one-way delay becomes greater than 250 milliseconds. Delay can be subdivided into two sub-components. They can be fixed delay components as well as variable delay components. Fixed delay components include propagation, serialization, and processing. The variable delay components include the queuing delay, jitter buffers as well as variable packet sizes.
2)Jitter (Delay Variability)
Jitter is the variation in inter-packet arrival time as introduced by the variable transmission delay over the network. Removing jitter requires collecting packets and holding them long enough to allow the slowest packets to arrive in time to be played in the correct sequence, which in turn causes additional delay. The conflicting goals of minimizing delay and removing jitter has led to the development of various schemes to adapt the jitter buffer size to match the time varying requirements of network jitter removal.
3) Packet Loss
IP networks cannot provide a guarantee that packets will be delivered at all, much less in order. Packets will be dropped under peak loads and during periods of network congestion. But due to the time sensitivity of voice transmissions, however the normal TCP- based retransmission schemes are not suitable. Approaches that compensate for packet loss have to be developed to overcome this problem.
4) Bandwidth Availability
Bandwidth is the portion of the network that is available to an application to transfer information on the network. The level of reliability and sound quality that is acceptable among users has not yet been reached and this is primarily because of bandwidth limitations and this also leads to packet loss. In voice communications, packet loss shows up in the form of gaps or periods of silence in the conversation, thus leading to a "clipped speech" effect that is unsatisfactory for most users and unacceptable in business communications.
Proposed solutions for problems associated in sending Voice over IP
Maintenance of acceptable voice quality levels despite inevitable variations in network performance is achieved using a variety of techniques. These techniques and solutions to problems that have been detailed above with regard to transmission of voice over IP are as follows.
1) One of the main problems of a very big end-to-end delay is the problem of echoes. The ITU standard G.165 defines performance requirements for echo cancellers. The way the echo cancellers work is that when the echoes are generated from the telephone network toward the packet network, the echo canceller compares the voice data received from the IP network to the voice data that is being transmitted to the IP network. The echo from the telephone network is removed by a digital filter on the transmit path to the IP network.
2) The approach to remove jitter involves counting the number of packets that arrive late and create a ratio of these packets to the number of packets that are successfully processed. This ration is then in turn used to adjust the jitter buffer to target a predetermined allowable late packet ratio. This approach works best with networks with highly variable packet and inter-arrival intervals such as IP.
3) Lost packets are a big problem in networks. Some schemes called lost packet compensation schemes used by voice over IP to overcome the problem of lost packets are as under.
a) Interpolate for lost speech packets by replaying the last packet received during the interval when the last packet was supposed to be played out. This works well when the incidence of lost frames is infrequent. It does not work very well for bursty loss of packets.
b) Another way is to send redundant information at the expense of bandwidth utilization. The basic approach replicates and sends the n’th packet of voice information along with the (n+1)th packet. This method has the advantage of being able to exactly correct for the lost packet. However, this approach uses more bandwidth and creates greater delay.
c) An alternative approach is to develop an algorithm in the digital signal processor that detects missing packets, and then replays the last successfully received packet at a decreased volume in order to fill the gaps.
d) Another problem is that of Out of Order Packets. When an out of order condition is detected in the network, the missing packet is replaced by its predecessor, as if it was lost. 6. Software Support to enable Voice over IP