Voip – An Insight Into A Progressing Technology


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Voip – An Insight Into A Progressing Technology

  1. 1. VOIP – AN INSIGHT INTO A PROGRESSING TECHNOLOGY David Tarrant Researcher in Intelligence, Agents and Multimedia School of Electronics and Computer Science University of Southampton, SO17 1BJ. dt302@ecs.soton.ac.uk www.ecs.soton.ac.uk/~dt302 ABSTRACT In this report we address the many developing technologies behind VoIP and which ones are now Technology behind the Voice over Internet Protocol leading the field into the next era of telecoms (VoIP) has been accelerating rapidly over the past communications. Beginning with a quick look at the few years however is still not ready for full older circuit switching networks such as the deployment. VoIP provides a Peer 2 Peer (P2P) link worldwide PSTN and how VoIP has been developed between users for making voice calls over the based on this technology and what improvements internet and servers are beginning to appear in all have been made. An outline of the technology areas of the community, providing different levels of behind VoIP and different uses is then drawn before service to users. However these services can be looking at the supporting server and client based on different protocols and standards making applications. We look at problems that have arisen interoperability between them difficult. This paper during the construction of VoIP and outline what provides an overview on the different standards and actions are being taken to resolve these. A case protocols used in VoIP asking: Where will these study from the research currently taking place at the take us in the future? What advantages does VoIP University of Southampton looks at some of the have to offer over current telephony? And what issues that have arisen and shows where implications are there for existing telecoms commercial interest is being focused. This will companies? enable a strong outlook to be drawn on where the future of the VoIP technology lies and how users Keywords can get involved. VoIP, SIP, H.323, Telephony. 2. BRIEF HISTORY OF TELEPHONY 1. INTRODUCTION COMMUNICATIONS Voice over IP (VoIP), otherwise known as IP Traditional telephone networks are connected via telephony, is the delivery of voice information over copper core to a circuit switched network. When a packet switched networks. This means sending user dials a number this causes a route to be set up voice information in digital form in discrete packets based on that unique number to another phone rather than in the traditional circuit-committed somewhere in the world. This global public switch protocols of the public switched telephone network telephone network (PSTN) is divided into individual (PSTN). A major advantage of VoIP is that it can local switch areas, which combined together make avoid the tolls charged by ordinary telephone up a Local Access and Transport Area (LATA). service by utilising fixed charge IP network services Each country can be dived into many geographic such as broadband. Recent development with LATAs that are controlled by local exchange carriers technologies such as SIP and hardware supporting (LEC). Depending on origin and destination, a call this standard has resulted in the production of a may traverse many switches and across multiple number of commercially marketed SIP handsets, LATAs to be established. Once established a both for wired and wireless networks, removing the need for a PC or laptop running a software handset, National Call or “softphone”. A subscription to a local server from a SIP handset or softphone provides you with all the normal telephony features including voice and fax, 0 2 380 595656 as well as opportunities for text and video services. LEC Type Geographic LATA Permission to make digital or hard copies of all or part of this (Southampton) International Call Unique Telephone No. work for personal or classroom use is granted without fee provided that copies are not made or distributed for profit or commercial advantage and that copies bear this notice and the full citation on the first page. To copy otherwise, to republish, to 00 44 2 380 595656 post on servers or to redistribute to lists, requires prior specific Country permission. Unique Telephone No. th LEC Type 4 Annual Multimedia Systems Conference, Electronics and Geographic LATA Computer Science, University of Southampton (Southampton) © 2004 Electronics and Computer Science, University of Southampton Figure 1 – Breakdown of UK phone number.
  2. 2. singular connection is thus achieved between the congested and quickest route. Figure 2 outlines this two clients via this route and voice packets can now process and shows two clients connecting over the be transmitted. Figure 1 shows a break down of internet. phone number into its many parts showing how a call would be established between 2 clients. Call CLIENT DISCOVERY costs can be implied by simply finding out which VoIP SERVERS LATAs have been traversed and applying charges appropriately [5]. INTERNET 2.1. Telephony into VoIP VoIP is based heavily on the already existent structure of the worldwide PSTN, however the active environment is the internet and thus VoIP has VoIP CALL – DIRECT CONNECTION been tuned to use existing network protocols where CLIENT 1 CLIENT 2 available. Like the PSTN network a user will be connected to a local exchange (server) which in turn Figure 2 – VoIP Call example (discovery and connection) is connected to other servers around the world. This decentralized architecture is ideal for end to These servers are able to communicate freely with end connections of only two users, however each other in order to find and connect users [5]. connection and management of conference calls VoIP has two main deployment methods based becomes more of a challenge. Making multipoint upon protocols from different developers. The ITU-T calls involves using IP multicast to transmit data to recommendation H.323 [3] follows a client server many users, which means that users must be able architecture much like the worldwide PSTN. Clients to transmit and receive multicast packets at their interact both for data transport and control with a location. small number of servers which coordinate and control the session. The IETF recommends the Session Initiation Protocol (SIP) [4] which is a highly decentralized architecture where servers are only used to locate users. A peer to peer link over the internet can then be established between the users without the need for an expensive powerful server. 3. THE SESSION INITIATION PROTOCOL (SIP) SIP (Session initiation protocol) is an Internet standard specified by the Internet Engineering Task Force (IETF) in RFC 2543 [4]. SIP is used to initiate, manage, and terminate interactive sessions between one or more users on the Internet. SIP borrows heavily from HTTP and the e-mail protocol Figure 3 – Structure of SIP SMTP, providing scalability, extensibility, flexibility, and capabilities for creation of new services. As a While servers are required to carry out some of the result SIP is increasingly used for Internet telephony more complex SIP features such as transcoding, it signalling, in gateways, PC phones, softswitches, is possible set up point to point or multicast and softphones, however is not limited to Internet conference calls without the need for a server. SIP telephony and can be used to initiate and manage has been designed specifically to allow clients to any type of session, including video, interactive make use of IP packets for both control and data games, and text chat. SIP takes advantage of the transport within calls. underlying technology of the internet, harnessing A generic SIP call involves a SIP User Agent (UA) this where possible so as to decentralize any locating a user on a registrar server (VoIP server) dependencies on the SIP server. A good example of and then issuing an invitation to them via a proxy which is how users are connected over a SIP server making use of any redirect servers where network: Unlike PSTN once the two users are appropriate. A successful SIP invitation consists of located the call is not connected via the servers or two messages: INVITE followed by an ACK. The the route taken in order to find the users. The INVITE message contains a session description internet already contains a route optimisation from the UA containing information on which type of framework at the packet level and thus users are media to caller wishes to use and can accept for the connected direct to each other using a peer to peer call. Media types, often referred to as codecs link. By default packets will traverse the least
  3. 3. included many such as GSM and the ITU codecs, these being those which establish and negotiate the some of which are already in use on mobile phone calling properties and media transmission types in networks and in other commercial voice use. H.450 defines a generic functional protocol on applications. This capability enables SIP to take full top of H.225 for all supplementary services and advantage of current technology and be integrated provides the only abstraction layer to H.323 where where possible. extra services can be harnessed in a call. Being based heavily on the SMTP and HTTP protocols, SIP adopts many of the methods and usability from these. None more so than user location, as well as each user having a unique number, users can also have a user@host.domain address which is simply aliased to their number. Finally SIP also provides a Session Announcement Protocol (SAP) and the Session Description Protocol (SDP) which support the establishment of multiparty conferencing sessions. SDP defines the description of multimedia sessions, while SAP enables periodic multicasting of information about active sessions. Together these enable third party users to join an already established session within a given time frame. Figure 4 – Structure of H.323 Figure 3 depicts the overall IETF SIP protocol suite H.323 can have a number of servers to perform and the many extensions available, with space left different tasks depending on the scenario. Typically for many more which are being worked upon by the an H.323 gatekeeper (GK) performs many of tasks IETF as internet drafts. SIP itself simply provides a equivalent to the SIP proxy server providing address small number of text based messages to be translation, RAS control, call redirection and exchanged in separate transactions between the resource management. H.323 can also create SIP peer entities. The session itself is described at decentralised point to point links to users for use in two levels. The SIP protocol contains the parties’ calls however the gatekeeper hander the addresses and protocol processing features (media initialisation and termination of the session rather types), with the body containing SDP which is a than the individual clients. In decentralised calling structured, text-based media description format. mode each client must also act as multipoint Since the message body is transparent to SIP any processor and be able to process media streams, type of SDP can be transferred, thus not limiting SIP including multicast. to use in VoIP, but opening it up to use in any To enable advanced features such as conference session based application. SIP extensions such as calling in H.323 further servers must be used to event notification (RFC 3265), session update (RFC establish and manage connections between multiple 3311), call transfer and call holding can then be users. A multipoint controller (MC) establishes an applied to complete the SIP core framework and H.245 control connection to each user for use in VoIP. negotiation of media communication types. The 4. H.323 multipoint processor (MP) is then able to decode and retransmit the streams as required. The H.323 H.323 defines system aspect requirements for MC component is responsible for selecting unicast multimedia communication systems over a packet or multicast media transmission and for choosing switching network. This includes registration, network/transport addresses. admission and status (RAS or RTP/RTCP) control, call setup as defined in H.225.0 and call setup and To establish a call using H.323 the II.323 call signalling as defined in H.245. H.225.0 defines an signalling procedure has to be carried out to alias type for carrying any standard Uniform establish valid H.245 connections via the Resource Locator (URL) [11]. H.323 version 4 [3] gatekeeper. The II.323 call signalling procedure introduced an H.323-specific URL, which may be begins when an originating H.323 client issues an used to resolve the address of a network entity to admission request (ARQ) to local gatekeeper in its which H.323 calls may be directed. Like SIP H.323 domain. When the corresponding confirmation also supports many audio and video codecs for use message (ACF) is received the call setup procedure in calls, as well as real time media transport continues with a SETUP and CONNECTION protocols (RTP and RTCP). message exchange. Upon successful establishment of a call the clients follow the H.245 capability Figure 4 depicts the structure of the H.323 protocol exchange procedure to open media channels which suite showing the compulsory objects in dark tan, both clients are able to support. In later versions (3
  4. 4. upwards) of H.323 clients are able to reduce gives a comparison between the many codecs with signalling overhead by using the Fast Connection bit rate (quality) and bandwidth required for each to procedure. This “FastStart” procedure is included as be used. With the operation of VoIP on a large scale an element in the SETUP message sent on call being over the internet a dedicated service for voice establishment. The “FastStart” element carries the calls is not possible unlike the old PSTN system. proposed media channel description defining the This can lead to other factors affecting the quality of media capabilities of the origin of the call allowing service as well as the limitations on ADSL upload media communication to begin after one round-trip speed. This introduces many problems such as message exchange instead of three. delay, packet loss, bandwidth limitations and echo. With a normal ADSL connection (512:256) users are 5. COMPARISON BETWEEN H.323 AND likely to experience latencies of between 80ms and SIP 400ms on a call, at around 200ms the flow of conversation becomes distorted. This is mainly due Both SIP and H.323 protocols can be used to most residential ADSL only supplying minimal efficiently to connect client to client calls using any upload bandwidth thus limiting the user’s range of media codecs. By placing a suitable proxy capabilities. All this is likely to change firstly when server in the middle is it also possible to perform a SDSL becomes more widely available and then as SIP to H.323 client call. Problems arise when BT themselves switch to IP based networks over the complex functions such conference calls are next 5 years, known as 21CN [6][7]. attempted. H.323 is based on a centralized server that uses a set of tightly integrated protocols to 7. VOIP SERVER PACKAGES control sessions and users connections. In contrast, SIP is often without a server, and its control There are many different VoIP server packages mechanisms are much more loosely coupled and available each with their advantages over the other depend a lot more on the client technology. SIP and each of their own complexity. At the time of clients are able to join and leave a conference by writing Asterisk1 was the most popular server using UDP signalling without the need for package providing built-in support for both H.323 centralised control. A central server in SIP is able to and SIP with functionality which can be harnessed provide easier location of clients as well as after only about 20 minutes for setup. SIP Express centralised session announcement (SAP) using the Router (SER)2 from iptel.org provides a proxy/router Session Description Protocol (SDP). SIP provides a for SIP sessions with an optional extensions module far more abstract protocol than H.323 able to be for the construction of a VoIP server. To set up a used outside of systems such as VoIP, such as that PSTN like system in SER would take a lot longer of multicast and unicast video streaming. H.323 is than the 20minutes of Asterisk, however SER based around only a few recommendations and provides its own scripting language for complete thus becomes easier to pick up and use for user control of functionality. Finally the last major developers. A centralized architecture such as that server package is VOCAL3, a much more of H.323 is more preferable to government services commercial solution with greater support for which can still keep and eye on usage and enable a business and enterprise users. VOCAL is an open line to be tapped, this being a requirement of any source project primarily designed as a SIP ITU phone network. As SIP is not specifically softswitch, however includes translator plug-ins for designed for telephony it does not have to comply support of H.323 endpoints. The following section with this ruling as yet [8]. provides a more in depth look at the technology of the 3 main server technologies and their 6. VOIP CODECS differences. VoIP codecs are used to convert an analogue voice Being the most popular at the time of writing signal into a digitally encoded version for Asterisk provides high levels of built in functionality transmission over the internet. The same codec is which is easy to manage. Its ability to be able to then used for the opposite purpose at its handle complex dialling plans and a wide range of destination. Codecs vary greatly in sound quality, voice, fax, text and video codecs for direct bandwidth required and computational interaction with users is able to seamlessly provide requirements. Each server, program, gateway, etc switchboard, voicemail and operator support on the typically supports several different codecs use of server. Asterisk supports both the H.323 and SIP which use is negotiated upon initialization of the call. standards and most popular codecs, those used for Server codec support is only required if the server is interaction with the user are shown in Figure 5. able to interact with the client in operations such as Asterisk also provides the proprietary IAX (Inter- switchboards and voicemail. Both SIP and H.323 Asterisk eXchange) protocol to enable the contain abstraction layers supporting a set of 1 Asterisk – http://www.asterisk.org standard codecs defined by the ITU for voice calls, 2 SER – http://www.iptel.org/ser being more built into H.323 than SIP. Appendix A 3 VOCAL – http://www.vovida.org/
  5. 5. interconnection of multiple Asterisk servers, with the Codec Asterisk SER VOCAL ability to forward communications between servers G. 711 Y Y or use one server as backup for another. IAX G.723.1 Y supplies a facility for VoIP with the same G.726 Y functionality as an LEC within a PSTN’s LATA. G.729 Y Asterisk provides a complete worldwide solution GSM 06.10 Y Y such as that currently provided by the existing LPC10e Y PSTN network and is able to seamlessly interface iLBC Y with the PSTN network by making use of specifically Speex Y designed hardware. By deploying this hardware in the correct fashion Asterisk is able to fulfil many of Figure 5 –Server support for codecs. the ITU and FCC regulations i.e. enabling users to be able to contact the emergency services from any Finally VOCAL from Vovida provides a much more handset. This is achieved by adding a simple structured enterprise solution which is designed as extension rule to the system to forward the a SIP softswitch however translators are available to appropriate numbers onto the correct end users. allow interoperability with H.323 and MGCP Asterisk provides both a structured number endpoints. The aim of the VOCAL project is to identification as well as the newer provide a SIP based replacement to the PBX/PSTN user@host.domain identification which is stored in a without necessarily providing any extra functionality simple xml type extensions file which Asterisk reads such as that of Asterisk and SER. Vocal is designed upon startup. Being designed to run on the UNIX to run on a distributed architecture of servers platform Asterisk contains modules which can be providing redundancy to handle downtime and high plugged in to enable user, extensions and calling volumes of usage [2]. Not providing any functionality profiles to be managed by a web interface system such as voicemail at the server end means VOCAL and harness the UNIX server capabilities. The one does not need any support for codecs, services downside of the current implementation of Asterisk such as voicemail and call holding are left to the (version 1.0) is the lack of support for IPv6 which individual clients rather than being supplied would be required for large scale networks and for centrally. This makes VOCAL the much more users currently behind a NAT (Network Address extensible system however from a users point a Translation) or Firewall. view is not the easiest solution. Being an open source project VOCAL also has to be installed from SIP Express Router (SER) is a high performance, source as can both Asterisk and SER (which both configurable, VoIP server supporting only SIP have more binary support however). VOCAL is a clients and services. SER uses a full scripting very formidable package with the source code language for its configuration, cutting down on the coming in at 78.1Mb, as opposed to 9.8Mb for SER number of individual configuration files and or 37Mb for Asterisk. improving scalability. This comes at the expense of requiring operators to learn a new language and to 8. VOIP CLIENTS mimic all the functionality which comes built into Asterisk would require a very large learning curve. Clients and phones for use in VoIP networks are To help Iptel the founders of SER provide many pre- now available as both hardware and software built modules for plugging into SER including solutions; both being able to carry out the same interface modules, accounting support and functionality of registering with a VoIP server to voicemail. SERs primary intended use is as a SIP enable calls to be made to other clients. Wireless proxy/router however also provides features to act handsets and mobile DECT technology phones are as a registrar and redirect server. As a proxy/router also now available which are simply plugged into a SER is designed to act as a standalone server and net cable at the base station rather than a phone provides no functionality for direct communication cable. Although this technology is now becoming with other servers. Redirection is built in, but this more widely available, users are still reluctant to buy does not guarantee the user a connection and they into the hardware field as upgrade opportunities are could just end up on a redirection loop or chain of limited without cost. Many solutions are at present servers. Unlike Asterisk, SER has inbuilt support for reflecting the server technology they are designed IPv6 as well as IPv4, and can listen for connections to connect to, the more advanced containing on ports under both protocols concurrently. IPv6 answering machines and call holding ideal for use capability provides greater support for mobile clients with a VOCAL server. Hardware phones are much using the mobility headers of IPv6 and support for more commonly supporting the SIP protocol with those users previously behind a NAT. SER provides H.323 support being relatively hard to find. This is minimal extra interaction with users and support reflected in servers such as VOCAL where a voicemail using a minimal set of base codecs as translations library for H.323 is provided as an add- shown in Figure 5. on rather than a built-in.
  6. 6. Software solutions are much more variable in their as local deployment as a primary telephony service. implementation and usage with upgrades being The project is looking at the many types of server released at a constant rate to keep up with the technology and using both hardware and software changing field and server technology. As with most clients to connect across a WAN with the additional software many open source and free solutions can involvement of some willing volunteers from the be found which install on any platform. Examples of Southampton Open Wireless Network (SOWN)6. commercially produced clients include Windows Currently the system is operating on a single Messenger1 for Windows and SJPhone2 for Asterisk server based in the main campus building Windows and Linux. Neither of these handling all users and calling profiles. A SER server implementations currently supports IPv6, and is also available on the same machine and is acting Windows Messenger lacks DTMF (dial tone) as a proxy to Asterisk to provide IPv6 support for generation facilities which prevent its use with testing with those clients which are able to use this Voicemail and other touch-tone operated services. protocol. With SER acting as a proxy all extensions Free and open source implementations include and user authentication is passed directly through to KPhone3 and LinPhone4, both available for Linux. Asterisk by rewriting the incoming host port and KPhone uses the KDE Qt library, while LinPhone translating the packets for use by Asterisk. This has a GNOME GTK-based graphical interface. This solution operates without problems for calls placed overview provides only a small selection from the between IPv4 users and IPv6 users providing both ever expanding field where more companies are clients are using the same IP protocol. However if getting involved on a daily basis. A recommended each client is using a different protocol the system site to keep up with the latest is www.voip-info.org will fail on direct connection of the call due to the which provides both listings of clients and server different connection types. To enable IPv4-IPv6 technology as well as useful guides to aid along the calls to be connected a further proxy has to be way. provided to translate the packet headers to enable each client to understand the data. This RTP-Proxy As with the server technology the more popular has to have support for both IPv4 and IPv6 traffic on clients, software and hardware, mainly support the the given network and the call between the clients IPv4 protocol with limited support for IPv6. At the should be routed through this proxy. Effectively an time of writing IPv6 support in physical handsets RTP-Proxy operates as a false client to which both was still commercially unavailable. KPhone has real clients send their information thinking that it is been patched to support IPv6 using SER as the their actual endpoint. The proxy then handles the server, but this version has now become traffic between the clients. This idea is good in superseded and no longer works with most recent prospect however research has shown not many of release of SER and the Linux Kernel. LinPhone has the client software packages currently support the built in support for IPv6 and is being developed with RTP-Proxy redirect information and still try to this in mind, however this software implementation connect the call directly [10]. This research is still on is still at beta and contains many bugs [9]. going under the 6NET project being run worldwide One of the most popular clients available currently including a section at the University. operates on the Skype system which provides pure VoIP using SIP (IPv4 only) through their own 10. CONCLUSION software package. This has been designed with VoIP technology and distribution is on the increase similar usage to MSN Messenger where a client is currently with many companies and institutes able to have a phonebook of users who can be seen carrying out research in the area to further enhance to be online or offline. This idea is now being this field, with the aim to provide the complete expanded to include answering machine features solution. The two main protocols of SIP and H.323 and also to support various hardware phones now vary dramatically in their construction and usage of becoming available with support for the Skype each has to be considered carefully. With the recent system (www.skype.com). ruling by the FCC that SIP is not a specific telephony protocol we now have a dramatic 9. VOIP IN OPERATION difference in the market between the two. In this section a brief overview is offered as to the Governments are now backing the use of H.323 due current state of research being performed at the to the legislation existing providing the ability to University of Southampton into VoIP. This research monitor and control the networks usage. While is being carried out in the Intelligence, Agents and 4 Windows Messenger - Multimedia Group5 within the School of Electronics http://www.microsoft.com/windows/messenger/ and Computer Science and is focused on network 5 SJPhone – http://www.sjlabs.com interoperability between IPv4 and IPv6 both over a 6 KPhone – http://www.iptel.org/products/kphone/ wired and wireless medium. The school regards 7 LinPhone – http://www.linphone.org VoIP as an appropriate technology for roaming 8 IAM Research Group – http://www.iam.ecs.soton.ac.uk academics (using mobility provided by IPv6) as well 9 SOWN – http://www.sown.org.uk
  7. 7. computing companies such as Cisco are backing REFERENCES SIP due to the involvement of SIP with the IETF and [1] Ho J. Hu J and Steenkiste P., Voice over IP: A other developing internet technologies. conference gateway supporting interoperability While H.323 is based on a client server architecture between SIP and H.323. Proceedings of the with heavy reliance on the servers to provide all ninth ACM conference on Multimedia. October functionality, SIP provides a much more distributed 2001. and flexible architecture. Multipoint conference calls [2] Dang G. Jennings C. and Kelly D., Procatical in H.323 are controlled centrally by a server while VoIP using Vocal, O’Reilly 1-14 (2002). SIP has SAP to announce active sessions over multicast. This does not rule out the use of web [3] ITU-T Recommendation H.323v.5 “Packet services such as the Universal Description, based multimedia communications systems”, Discovery and Integration (UDDI) to enable clients May 2003. to find active sessions in both SIP and H.323. VoIP [4] Rosenberg J. Schulzrinne H. Camarillo G. has great opportunity to provide internal networks Johnston A. Peterson J. Sparks R. Hendley M. within businesses using a series of servers running and E. Schooler. SIP: Session Initiation Asterisk and IAX (or VOCAL). These servers could Protocol, RFC 3261, June 2002. be spread over a wide area with links over the [5] Cordero R. Williston J. Voice over Internet internet between them to provide inexpensive long Protocol: History of Telephone and VoIP distance calls to companies who operate worldwide. service. UNC School of Law. April 2004. To provide full functionally is likely to either involve http://www.unc.edu/courses/2004spring/law/357 complex routing through many NATs or use of the c/001/projects/jennwill/VOIP/history.html last IPv6 protocol which is as yet not fully supported. accessed 17th November 2004. For home users VoIP provides no real advantage [6] 21ST Century Network, BT Wholesale, over the current PSTN unless everyone decides to http://www.btwholesale.com/application?origin= make the switch and/or latency on internet hubPromo.jsp&event=bea.portal.framework.inte connections dropping. Due the varying amount of rnal.refresh&pageid=wide_article_new&nodeId= technology available, users are going to be reluctant navigation/node/data/our_business/hot_topics/2 to buy into a changing market until some standards 1cn last accessed 17th November 2004. are set or are stopped in development. A good [7] Phillips L., BT begins switchover from PSTN to example is the current number of codecs available IP-based network, Digital media news for for voice transport. Europe, June 2004. BT’s announcement to move to VoIP technology on [8] Wirbel L., FCC Commissioners agree on inter- its own networks has provided a major step in the state nature of VoIP, Comms Design, November field by one of the biggest telecoms companies. 2004. There has been much speculation as to what the http://www.commsdesign.com/news/showArticle impact of VoIP would be on companies such as BT .jhtml?articleID=52600160 last accessed 17th however the 21CN ensures BTs investment in the November 2004. future of telecommunications and promises them a major market share. In the near future other [9] The VoIP Wiki, A reference guide to all things telecoms companies will have to change over their VoIP, http://www.voip-info.org, last accessed own systems and by 2009 it is expected the entire 17th November 2004. network will have switched to VoIP. [10] Tarrant D. VoIP and IPv6 – A series of guides to deployment of VoIP on a large scale network, VoIP technology is here to stay however it will be August 2004,http://www.ecs.soton.ac.uk/~dt302 more evident in business use than in the home. last accessed 17th November 2004. Businesses can harness VoIP to manage their own telecoms networks and cut out one of the major [11] International Telecommunication Union, costs, especially in those spread worldwide. Publications: Recommendations: Series H, Telecoms companies are likely to follow the BT http://www.itu.int/rec/recommendation.asp?type example and invest in the field taking us into a new =products&lang=e&parent=T-REC-H, last era of communications. accessed 17th November 2004.
  8. 8. 11. Appendix A Codecs used in VoIP for communication between clients showing bit rate (quality) and bandwidth consumption of each. Standard bit rate sampling Raw Bandwidth Name Description Remarks by (kb/s) rate (kHz) Usage (ADPCM Intel, IMA ADPCM 32 8 var ) DVI Also known as ulaw/alaw, mu-law G.711 ITU-T Pulse code modulation (PCM) 64 8 87.2 Kbps (US, Japan) and A-law (Europe) Subband-codec that divides 16 kHz G.722 ITU-T 7 kHz audio-coding within 64 kbit/s 64 16 * 120 Kbps + band into two subbands, each coded using ADPCM Coding at 24 and 32 kbit/s for hands- G.722.1 ITU-T free operation in systems with low 24/32 16 * 60 Kbps + Variable Frame Size frame loss Dual rate speech coder for multimedia Part of H.324 video conferencing. G.723.1 ITU-T communications transmitting at 5.3 5.3/6.4 8 20.8/21.9 Kbps DSP Group. and 6.3 kbit/s 40, 32, 24, 16 kbit/s adaptive 16/24/32 31.5/47.2/55.2/6 G.726 ITU-T differential pulse code modulation 8 ADPCM; replaces G.721 and G.723. /40 3.4 Kbps (ADPCM) 5-, 4-, 3- and 2-bit/sample embedded G.727 ITU-T adaptive differential pulse code var. ? var ADPCM. Related to G.726. modulation (ADPCM) Coding of speech at 16 kbit/s using CELP. Annex J offers variable-bit G.728 ITU-T low-delay code excited linear 16 8 31.5 Kbps rate operation for DCME. prediction Coding of speech at 8 kbit/s using G.729 ITU-T conjugate-structure algebraic-code- 8 8 31.2 Kbps Low delay (15 ms) excited linear-prediction (CS-ACELP) GSM Regular Pulse Excitation Long-Term ETSI 13 8 30.3 Kbps Used for GSM cellular telephony. 06.10 Predictor (RPE-LTP) 10 coefficients. Also known as FIPS LPC10e US Govt. Linear-predictive codec 2.4 8 7.8 Kbps 1015 iLBC (internet Low Bitrate Codec) Frames are encoded completely iLBC IETF 13.3 8 27.7 Kbps designed for narrow band speech. independently. Speex is based on CELP and is 2.15- Speex N/A designed to compress voice at bitrates 8/16/32 7.4 Kbps + open-source, multirate codec 44.2 ranging from 2 to 44 kbps.