Voice Over Internet Protocol(VoIP)

Transcription

1 Voice Over Internet Protocol(VoIP) By Asad Niazi Last Revised on: March 29 th, 2004 SFWR 4C03 Major Project Instructor: Dr. Kartik Krishnan

2 1. Introduction The telecommunications companies around the world are trying to put a stop to rising communication costs. The integration of packet and voice networks is one of the alternatives that could significantly reduce these costs. A Voice over packet (VOP) application provides such service by allowing both data and voice information to be sent over the IP network. Voice over IP or Voice over Packet is comprised of several interconnected processes that convert voice signals into packet streams. These packets are then sent over the packet network concurrently with other IP traffic and eventually converted back to voice near their destination. VoIP packets can also carry with them fax information and modem data, providing Voice, Fax, and Data/Modem transmission over all types of data networks including IP and ATM networks. RTP (Real Time Protocol) is used to provide QoS (Quality of Service) by managing and delivering the packets on time. Internet telephony service providers (ITSPs) can improve their services by using efficient private networks. This type of communication provides rich benefits to all classes of users. Just imagine what would it be like to have the long distance calls cost as much as the local call. However it s not just the reduced cost but also the other benefits which the users will enjoy. For example, the web-enabled call centers, collaborative white boarding, remote telecommuting and personal productivity services etc. The early implementations provided low cost PC to PC voice communication; however, the technologies are improving and will hopefully provide regular telephone services in the near future. 2. A typical VoIP Call: The general two way VoIP phone call consists of the following sequence: 1. The user picks up the handset which sends an off-hook signal to the signaling application running on a router. 2. The session application provides dial tone and waits for the destination telephone number 3. The user dials the telephone number and the session application keeps accumulating them until it matches a configured destination pattern. 4. The telephone number is mapped to an IP host via the dial plan mapper. 5. The session application then runs the H.323 session protocol to establish a two way transmission channel. 6. The special coder-decoder compression schemes (CODECs) are used at both the ends for packet formation and compression. 7. When either end of the call hangs up, the session ends. The line becomes idle and

3 waits for the next off-hook signal. 3. VoIP Architecture VOP Applications require software and hardware modules which can be configured to provide flexible and scalable solutions. For this purpose, a large number of new Application Programming Interfaces (APIs) have been defined for application developing. A VOP application provides the Internetworking Function (IWF) making it possible to send voice and data together on packet networks. There are two basic types of information to be handled by the VoIP software i.e. voice and signaling information. The responsibility of the software can be divided into four major areas: 3.1 Voice Packet Software Module It performs echo cancellation, voice compression, voice activity detection, jitter removal, clock synchronization, and voice packetization. 3.2 Telephony Signaling Gateway Software Module This software module manipulates basic connection set-up signals. It detects these signals and translates them into state changes for example off-hook, on-hook, trunk seizure etc. 3.3 Packet Protocol Module It converts telephony signaling protocols into packet signaling protocols in order to set up connections over the packet network. Some header information is also added to these packets before transmitting them over the packet network. 3.4 Network Management Module This module manages other VoIP modules and provides interface for their configuration. 4. Performance Issues The reduced cost does come with some performance and quality control issues which need to be addressed. Some of them are discussed below: 4.1 Delay The delay is due to the nature of IP networks. It causes two problems: echo and 'talker overlap'. Echo is produced by the reflection of speaker s voice from a far end telephone equipment. Echo could become significant if the round trip delay is more than 50 milliseconds. Talker overlap happens when one talker steps on other talker's speech. It could become significant if the one-way delay is more than 250 milliseconds. Therefore, delay reduction is a basic requirement for the functionality of VoIP networks. The major

4 sources of delay are slow accumulation of frames and slow processing. There are two types of delays: propagation delay and handling delay. Propagation delay is due to slow networking medium and handling delay is caused by the hardware devices used for handling voice information. The latter could have a significant impact on voice quality in a packet network. Echo is removed by using an echo canceller which compares the voice data packets received from the network and the ones being sent. A digital filter is used to remove the echo from the hybrid network. 4.2 Lost Packets Lost packets can seriously jeopardize the conversation. Data packets can be retransmitted; however, compensating for lost voice packets is not easy. One way to overcome this problem is to interpolate the missing slot by the previous (n-1) packet. This makes it look continuous but if packets are lost frequently the speech becomes deteriorated. Another approach is to send current (n) packet with the next (n+1) packet. This approach costs much bandwidth and causes delay. A hybrid approach sends some information with n+1th packet and therefore uses less bandwidth. 4.3 Jitter Jitter refers to a problem caused by packets with variable delay times. To remove the jitter, data is buffered at the receiving end and then played at a constant rate. Two new protocols have been specially developed for this purpose, namely the RTP(Real-time Transport Protocol) and RTCP(RTP Control Protocol). The RTP is used to transport the digitized samples of real-time information. RTCP provides information on the quality of the transmission link. 5. New Security Issues: What would you do if you find out that your telephone has been hacked, or your country code is blocked, or your telephone is spreading worms and viruses? All this can happen since the packet network is being used which is quite vulnerable. All these issues have to be dealt with in order to make it safe using a VoIP telephone. To overcome these threats, phone companies undertake several measures. For example, the users are not allowed to have easy passwords, the account might be locked out on multiple wrong password entries or making too many log distance calls. However, many phone companies are not using the best practices which may cause serious problems in the future. 6. Conclusion: The VOIP technology is still going through its developmental stages. Several companies are investing heavily in research and development activities related to VoIP. Hopefully, the problems involving delay, jitter, unreliable delivery and security issues will soon be resolved, providing proper communication. Moreover, software and hardware is being developed for advanced VoIP features. Once fully developed and implemented, this

5 technology could well be a revolution in telecommunications. References: Hensell, Lesley "The New Security Risk of VoIP" E-commerce Times. 02 Oct < "Voice Over Packet White Paper" Texas Instruments. January 1998 Version 2.2 SPEY005 < McKinney, Jenny "VoIP Technology Overview" < "VoIP Gateway Solutions : Overview" Texas Instruments. < &path=templatedata/cm/level1/data/bband_voipgateways_ovw> "Voice Over IP (VoIP) : Overview" Texas Instruments. < NJYP3QC1JAVRVQQ?templateId=975 &path=templatedata/cm/level1/data/bband_voip_ovw> "VoIP" SearchNetworking.com Definitions <