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	Content • General • Quality of Service (QoS) • Making an IP call • Links for more information General Internet telephony (called as well Voice over IP - VoIP) refers to familiar telephony services - voice and fax, which are transported via the Internet, rather than via the PSTN or ISDN. The basic steps involved in 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 without the need of establishing a circuit switched connection as it is in the case of ISDN and PSTN. This type of connection is called packet switched. Of course at the receiving end the signal needs to be converted back to analog. Initially the Internet telephony was introduced (in early 1995) as Internet Phone software designed to run on a 486/33-MHz (or higher) personal computer (PC) equipped with a sound card, speakers, microphone, and modem. Calls were to be conducted over a normal Internet connection and QoS was not possible. This was PC -to-PC service that required both PCs to be equipped with the proper software. The idea developed rapidly and as next step so called IP telephony Gateway servers were introduced, which allowed standard telephones to be used. Basically these gateways convert back an forth voice to be carried on IP and PSTN or ISDN networks, i.e. a call can be originated on a standard telephone connected to a PSTN or an ISDN network, then at a gateway switched to an IP network, and then again yet another gateway back to PSTN or ISDN where it can reach another standard telephone. On the standardisation side in May 1996, the International Telecommunications Union (ITU) ratified the H.323 specification, which defines how voice, data, and video traffic will be transported over IP–based local area networks and addresses the core Internet-telephony applications by defining how delay-sensitive traffic, (i.e., voice and video), gets priority transport to ensure real-time communications service over the Internet. There are two major protocols being used for VoIP. Both protocols define ways for devices to connect to each other using VoIP. Also, they include specifications for audio codecs. A codec, which stands for coder-decoder, converts an audio signal into a compressed digital form for transmission and back into an uncompressed audio signal for replay. • The H.323, a standard created by the International Telecommunications Union (ITU) is a comprehensive and very complex protocol. It provides specifications for real-time, interactive videoconferencing, data sharing and audio applications such as IP telephony. H.323 incorporates many individual protocols that have been developed for specific applications. • The Session Initiation Protocol (SIP) created by the Internet Engineering Task Force (IETF) is an alternative to the H.323. SIP is a much more streamlined protocol, developed specifically for IP telephony. Smaller and more efficient than H.323, SIP takes advantage of existing protocols to handle certain parts of the process. For example, Media Gateway Control Protocol (MGCP) is used by SIP to establish a gateway connecting to the PSTN system. Quality of Service (QoS) The main issue related to VoIP is the so-called Quality of Service (QoS) issue. This has to do with the nature of the Internet. Not using circuit switched connection from one side provides better utilisation of the network resources (circuit switched connections are established at the beginning of the call and freed when the call is finished being occupied even if there is no voice traffic, e.g. when the parties involved are silenced, whereas packet switched networks "open" the connection just long enough to send a small chunk of data, called packet), but from another, it cannot itself guarantee that the voice packet will be delivered in time and in sequence (packets in Internet normally travel over road that cannot be pre defined - for example IP packages carrying portions of a picture you want to download from a web site in Germany to Switzerland may come to you via different paths - some via US, others via Japan) and without losses (loss of some packets introduces gaps or periods of silence in the conversation, leading to a clipped-speech effect). Delay occurs during the processing of the packets as well. Another issue is the limitation of the available bandwidth - Internet connections may well be crowded on certain legs of the Internet network if there is no sufficient bandwidth provided. QoS is what aims to guarantee that VoIP provides a circuit switched fixed line quality of speech. Consequently, being based on IP, a VoIP service may be provided on any carrier that can provide IP and QoS, e.g. xDSL, Cable, LAN, DECT, and specially dedicated IP networks. Making an IP call There are a number of ways that you might talk to someone using VoIP. What you may need is one or more from the following: a computer, an IP phone, a standard (non-IP) phone, and a network connection that in one or another way can handle IP. If you've got a computer or a telephone, you can use at least one of these methods without buying any new equipment: • Computer-to-computer - This is certainly the easiest way to use VoIP but hardly the best from quality point of view. All you need is the software (free or very low-cost one is easily available), a microphone, speakers, a sound card and an Internet connection, preferably a fast one like you would get through a cable or DSL modem. Except for your normal monthly ISP fee, there is usually no charge for computer-to-computer calls, no matter the distance. • Computer-to-telephone - This method allows you to call anyone (who has a phone) from your computer. Like computer-to-computer calling, it requires a software client. In addition it requires a service provider (this may be the one that provides you with the software) who provides you with the access from the Internet to the fixed network, and which may require a small per-minute charge. • Telephone-to-telephone - Through the use of IP gateways, you can connect directly with any other standard telephone in the world. What you need is a service provider which gateway you must call in using whatever POTS you have and provide the called party number - your service provider will connect you through their IP-based network which via another gateway close to the destination will switch the call possibly to another POTS. As far as the IP phones are concerned, there are basically two types: one that connects directly to an IP network and the other that connects to a computer and used to replace the computer's microphone and speaker. Whereas the latter is used just to facilitate your VoIP experience when using a PC, the former needs a network connection that utilises the IP protocol. Such IP phones in fact comprise part of the hardware and software that the computers use for Internet telephony - basically they do not utilise modems in the sense computers do. Examples for networks that can be used are business LANs or IP based PBX; Cable or xDSL are the most likely candidates to bring the IP telephony in your home (apart of the normal way of using computer). Still being not perfect VoIP has a number of advantages compared to the traditional networks. For example: • Packet switching allows several telephone calls to occupy the amount of space occupied by only one in a circuit-switched network. Using PSTN, that 10-minute phone call consumed 10 full minutes of transmission time at a cost of 128 Kbps. With VoIP, that same call may have occupied only 3.5 minutes of transmission time at a cost of 64 Kbps, leaving another 64 Kbps free for that 3.5 minutes, plus an additional 128 Kbps for the remaining 6.5 minutes. Based on this simple estimate, another three or four calls could easily fit into the space used by a single call under the conventional system. And this example doesn't even factor in the use of data compression, which further reduces the size of each call. • Utilising IP, allows for introduction of new supplementary services, e.g. Advanced messaging, Directories, Web access, etc. Links for more information IPxStream http://www.iptelephony.org/ IP Telephony online http://www.internet-telephony.org/index.html Internet Telephony Consortium http://itel.mit.edu/ SIP Forum http://www.sipforum.org/ VoIP Council http://www.voipcouncil.org/ To find out more about the extensive swissvoice IP products portfolio click here.