What is Voice over IP?


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What is Voice over IP?

  1. 1. Technical papers | What is voice over IP? What is Voice over IP? Voice over IP, or Voice over Internet Protocol to use the full title, is simply a means of making telephone calls over a data network instead of over the traditional analogue public switched telephone network (PSTN). The term VoIP describes the use of the Internet Protocol (IP) to transfer speech between two or more sites. Inherent in the term is the management of the protocol. In general, this means that the voice information is encoded into discrete digital packets and then transferred across an IP- based network. There are many advantages to this method of telephony, primarily the cost savings that can be made by avoiding the use of the traditional PSTN. In addition to cost savings, the digital nature of VoIP allows easy administration, the implementation of additional services such as voicemail, and a reduction in the physical cabling required for new installations. However, due to the distributed nature of the internet the Quality of Service (QoS) can often suffer, and there are still a number of technical issues affecting the widespread adoption of VoIP. Brief history The term VoIP was first used by the VoIP Forum (a group of major companies including Cisco, Vocalec and 3Com) to promote and develop the use of the International Telecommunications Union’s (ITU) H.323 protocol. The forum also worked on the standardisation of directory services and the use of touch-tone standards for accessing voicemail. The ability to transfer voice over the internet, rather than the PSTN was first made possible in February 1995 when Vocaltec released its Internet Phone software. This software was designed to run on a standard personal computer (PC) equipped with a sound card, speakers, microphone, and modem. The software encoded and compressed the voice signal, converting it into IP packets that were then transmitted over the internet. However this PC-to-PC internet telephony only worked if both parties were using the Internet Phone software. Inherent in this technology were a number of severe limitations. For example, no formalised ringing protocols meant that it was necessary for both parties to pre-arrange the time of the call in order to be available to make the final connections themselves. Users also had to contend with the lack of any kind of directory service, poor quality, and frequent delays. Internet telephony has made a number of important advances since 1995. Many software developers now offer PC telephony software but, more importantly, gateway servers are emerging to act as an interface between the internet and the PSTN. Equipped with voice- processing cards, these gateway servers enable users to communicate via standard telephones over great distances without using the ‘long distance’ telephone network. How does VoIP work? When a traditional call is made using the PSTN, the analogue lines are kept open between the two callers for the entire duration of the call; this is called a circuit. This small segment of the network is exclusively used for this call and will be unavailable to any other users until the call has finished. VoIP does not require such dedicated circuits as it translates voice signals into packets of digital data. These packets are then transmitted over Ethernet or wireless networks, with no part of the network used exclusively by callers. © Becta 2004 Valid at September 2004 page 1 of 7 Review at December 2004
  2. 2. Becta | Technical paper | What is voice over IP? VoIP requires the use of Codecs (COder-DECoders). These can be software, or hardware such as microphones, IP telephones or other similar devices. They are required to convert analogue signals (what we say and hear) into digital (for transmission over the network) and back into analogue. Figure 1 (below) shows how schools can combine their existing voice and data networks to carry both voice and data traffic. In the example shown, the LEA provides schools A and B with a broadband connection that is used to carry both data and VoIP traffic. The schools incur no call costs when calling the LEA or each other as all calls are passed along existing connections. School C has not yet converted to VoIP and therefore runs two separate networks, one for voice and one for data. The LEA also has a VoIP gateway server that allows it to send and receive calls on the PSTN. Should school A or B wish to call outside the VoIP service area the calls pass through the gateway and out over the PSTN. Similarly, if an external site wishes to call school A or B the calls are routed into the VoIP gateway at the LEA and over the network to the school. Figure 1 – a typical H.323 implementation of VoIP with other networks If part of the network uses the PSTN, then after a call is initiated it goes over the local PSTN to the nearest gateway server. This digitises the analogue voice signal, compresses it into IP © Becta 2004 Valid at September 2004 page 2 of 7 Review at December 2004
  3. 3. Becta | Technical paper | What is voice over IP? packets, and moves it onto the IP network for transport to the gateway server at the receiving end. This server converts the digital IP signal back to analogue and completes the call via the local PSTN. Figure 2 – the relationship between IP and PSTN systems Figure 2 shows the relationship between VoIP, PSTN, the internet and the gateways. It is possible to route calls using VoIP over existing infrastructure as long as the necessary gateways are in place. Today, VoIP networks operate by translating telephone numbers to IP addresses and placing an H.323 or Session Initiation Protocol (SIP) call to the device. Should the route between the callers involve all IP networks with no parts PSTN, then no gateways are required to translate the digital data into analogue. The easiest way to start using VoIP is to install a software client onto several networked PCs. These clients are common and there is an abundance of software [http://www.iptelephony.org/GIP/vendors/client-phones/] that allows people to talk to each other providing they have a network connection, microphone and speakers. Person A can call person B simply by typing the IP address of person B’s PC into the software. The voice data between the two PCs is carried over the network along with normal data traffic. This is a relatively simple and cheap way for people to communicate using an existing network infrastructure. The next step up from this basic software implementation is to use IP telephones. These devices look similar to standard telephones and plug straight into the existing data network. They contain all the software and equipment needed to establish and receive VoIP calls; they behave as a normal telephone does, except that they transmit and receive data over computer networks and © Becta 2004 Valid at September 2004 page 3 of 7 Review at December 2004
  4. 4. Becta | Technical paper | What is voice over IP? not the PSTN. Such devices are normally completely independent and are not centrally controlled by a public branch exchange (PBX) or server that would control a normal analogue system. Larger, more complex VoIP networks include many IP telephones linked to one central server that controls functions such as voicemail, extension numbers and caller groups. These larger, wide area networks (WANs) often cover many geographically dispersed sites. Such networks use hardware ‘bridges’ to transfer data between sites in the WAN and these bridges can also convey VoIP traffic. Where is VoIP being used? The main application of VoIP is currently within intranets. As voice communications are dependent upon continuous time-based audio (speech), it is imperative that the bandwidth across which the IP packets are transferred is stable and that packet loss is kept to an absolute minimum. Any packet loss during a VoIP call will result in the speech being transferred sounding ‘clipped’ and/or unintelligible. In addition, delays to the routing of the packets will cause momentary pauses and delays in the received speech, reducing the quality of the call. Consequently local conditions can adversely affect VoIP calls, especially when they are routed across the internet. Therefore VoIP is best suited to use across local area networks (LANs) and wide area networks (WANs) where routing via the internet is not required. VoIP in practice Australia’s Academic and Research Network (AARNet) [http://www.aarnet.edu.au/services/voip/index.html] has implemented VoIP telephony throughout its 37 member organisations. As long distance telephony is common in Australia, VoIP presents significant cost savings – up to 70 per cent over PSTN solutions. The advantages and disadvantages of VoIP VoIP offers a number of potential advantages and benefits to its users: • reduced operating cost, including lower telecom charges • flexibility • reduced infrastructure • integrated services and greater user features. The major advantage of VoIP is cost saving (although do not expect immediate cost savings due to the upgrading of existing infrastructure and systems implementation). In addition to the financial savings, there are a number of additional benefits; it is innately scalable, allowing the quick and easy addition of new terminals and connections. The voice data can be easily processed using a standard PC, allowing call monitoring, voicemail, and a range of other features to be implemented. A single network topology can be put in place, requiring only network cabling to be installed in a building rather than both data and traditional telephone cabling. VoIP systems can also be configured using standard networking tools such as SNMP (simple network management protocol). A side benefit of VoIP is that many of the steps to full implementation are the same as the steps required to implement video conferencing and similar technologies. Whilst VoIP is not strictly speaking a subset of video conferencing there are many similarities, and a large number of products on the market implement both technologies in a single package. © Becta 2004 Valid at September 2004 page 4 of 7 Review at December 2004
  5. 5. Becta | Technical paper | What is voice over IP? However, there are drawbacks: • the network has to be fast, reliable and offer high quality of service • compatibility with existing firewalls and security devices may cause problems. (Firewalls need to be H.323 and possibly SIP compliant depending on the VoIP solution.) Are there any VoIP standards? IP packets are unreliable and use various re-transmission techniques, hence the need for standards to achieve an acceptable level of service. When considering implementing VoIP there are a number of standards that decision makers should examine. • H.323 is the protocol ratified in 1996 by the ITU. As previously discussed, this has become the protocol of choice for the transfer of voice information across the internet. The H.323 standard has also become somewhat of an ‘umbrella’ for a number of other protocols that add extra facilities. • SIP is becoming a more attractive option even though H.323 is the most established protocol. SIPs ‘lighter’ system use in connection and transfer standards means that a number of VoIP system designers are opting to use SIP. The compatibility of SIP devices makes the implementation of this protocol an easier proposition, allowing IT staff to choose equipment from a number of separate manufacturers and still expect that it will be compatible with existing equipment. As these two standards are quite different, interoperability between H.323 and SIP is only possible via a mediator or gateway. What should be considered when purchasing a VoIP solution? • Here are some, but by no means all, of the considerations you should discuss with your supplier when they recommend a VoIP solution. Choose a VoIP supplier who will asses your current structure and decide on the best solution for your needs. This may possibly be a part-PSTN/part-VoIP solution. Asses the extra features that come with VoIP such as phone grouping, instant messaging, instant messaging (IM) and answerphone services. • Ensure that your network can handle the voice traffic load. Look at your existing call usage such as volume, peek usage times and duration of calls. You will need this information when deciding if your current network is able to cope with the expected rise in bandwidth requirements. • Establish a test procedure. Check for voice quality, gaps in conversations, latency, echo and consistency. Make calls internally and externally at various times of the day to ensure the network can cope even at its busiest times. • Obtain high quality, preferably hardware, Codecs. These will play the most crucial part in how the VoIP ‘feels’ to its users. Generally, the higher quality the Codec the better the voice quality. • Administration of VoIP calls should be included, and call logs should be produced automatically in an easy-to-read format. More detailed logging should also be available. • Training should be comprehensive and thorough for both administrators and users. What type of equipment will be needed? The H.323 standard specifies four types of device that together form the core of a VoIP network: © Becta 2004 Valid at September 2004 page 5 of 7 Review at December 2004
  6. 6. Becta | Technical paper | What is voice over IP? • Terminals/IP telephones are the real-time bi-directional devices capable of broadcasting and receiving H.323 traffic. • Gateways perform a translation role, converting H.323 traffic to other protocols and vice-versa. Thus a gateway is often employed to provide communication between a H.323 terminal and the PSTN. • Gatekeepers provide additional services to a VoIP system such as address controlling, authentication, billing, accounting and bandwidth management. • Multipoint control units (MCUs) provide connectivity for three or more terminals allowing conference calling. Whilst gateways, gatekeepers and MCUs are logically separate components of the H.323 standard they are often implemented in a single physical unit. Fully functional IP telephones are available, as are hardware ‘gateways’ which usually perform gateway, gatekeeper and MCU functions. Software solutions are also available for the majority of functions, from simple internet-capable ‘soft terminals’ to sophisticated software gateways, gatekeepers and MCUs. SIP implementations require a substantially different array of equipment to H.323: • A SIP-compliant end device, such as a SIP telephone, is required and this can be hardware or software. • A SIP registrar/location server accepts registration requests from end points and maintains a whereabouts list of all its registered users. • SIP proxy servers stand between the end points and the location server. These allow calls to be redirected to end points and to temporary contact addresses. In order for VoIP to perform at acceptable levels, the network bandwidth must be great enough to cope with the number of calls anticipated. Whilst the load of voice traffic is not high enough to cause concern for the majority of LANs, care must be applied to bandwidth calculations for distributed network use of VoIP. As all voice communication is converted to IP traffic via the VoIP protocols and hardware, there is no need for network architecture to be altered other than to accommodate VoIP equipment. The future As the take up of broadband increases and IP networks become more reliable, the use of VoIP will grow. The normal course of replacing existing analogue communications with digital solutions will continue, and the integration of the two separate networks into one will become more common. However, some major obstacles will have to be overcome if VoIP is to succeed as a complete replacement to PSTN. • Quality of service – the expectations in the communications market are that voice communications should have a 100 per cent uptime and that voice quality should be almost perfect. • Security – it is possible to tap almost any form of communication be it analogue or digital, and VoIP manufactures will have to ensure their solutions are secure from all but the most sophisticated attacks. • Charging – at the moment providers using the PSTN charge by distance called, but with VoIP the model may change to one where telecommunication providers can start reserving bandwidth and charge users accordingly. The existing telecommunications providers may start replacing the national analogue networks with digital equipment and this may lead to a gradual replacement of equipment inside homes and schools. So-called ‘soft switches’ are currently in development, these switch voice traffic © Becta 2004 Valid at September 2004 page 6 of 7 Review at December 2004
  7. 7. Becta | Technical paper | What is voice over IP? using software allowing all the old functionality found in a large exchange to be achieved whilst reducing overall costs and increasing control, billing and other services. Other sources of information The International Engineering Consortium (IEC) VoIP Tutorial (PDF) [http://www.conferzone.com/resource/wp/VOIP_whitepaper.pdf] VoIP Watch — news site [http://www.voipwatch.com/index.php3] Voice over IP and IP telephony: reference links [http://www.cis.ohio-state.edu/~jain/refs/ref_voip.htm] The International Telecommunications Union [http://www.itu.int/home/index.html] The SIP Centre [http://www.sipcenter.com/] Palomar College VoIP Project Journal [http://www.palomar.edu/nibblesandbits/PC_VoIP_Project.htm] © Becta 2004 Valid at September 2004 page 7 of 7 Review at December 2004