A Study of Voice over Internet Protocol
Voice over Internet Protocol, is an application that enables data packet networks to transport real time voice traffic. VOIP uses the Internet as the transmission network. This paper describes VoIP and its requirements. The paper further discusses various VoIP protocol, security and its market.
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(IJCSIS) International Journal of Computer Science and Information Security, Vol. 8, No. 2, 2010
A Study of Voice over Internet Protocol
Mohsen Gerami
The Faculty of Applied Science of Post and Communications Danesh Blv, Jenah Ave, Azadi Sqr, Tehran, Iran. Postal code: 1391637111
e-mail: [email protected]
Abstract—Voice over Internet Protocol, is an application that enables data packet networks to transport real time voice traffic. VOIP uses the Internet as the transmission network. This paper describes VoIP and its requirements. The paper further discusses various VoIP protocol, security and its market. Keywords: VOIP; H.323; SIP; Security; Market;
VoIP works in a relatively simple way. Each time you make a phone call your voice is converted into a stream of data. Then, rather than being sent over the phone network, this data stream travels over your broadband internet connection. Each data packet is labelled with its destination address (the person you're calling) and moves through the internet in the same way as web pages and file downloads. When they get to their destination, the packets are reassembled and converted back into sound waves. When you have this process happening simultaneously in two directions, you've got a phone call. Most VoIP services also come with an allocated landline phone number which allows other people to call you. In these cases the call will be routed to the nearest handover point (called a POP or point of presence) and then travel over the internet to your VoIP phone or computer [3]. II. REQUIREMENTS FOR VOIP
I.
INTRODUCTION
VoIP, or Voice over IP, is an application that enables data packet networks to transport real time voice traffic. It consists of hardware and software that allows companies and persons to engage in telephone conversations over data networks. As a result, more and more companies have become interested in implementing VoIP [1]. All VOIP services are not built alike. Some allow you to call anyone with a phone, while others restrict your calls to only other clients using the same VOIP service. You can choose between three different ways to set up a VOIP system on your computer. You can use an ATA (analog voice adaptor) which performs the analog-to-digital conversion, and is plugged in to your computer at one end and your telephone at the other. You can use an IP phone, a phone specifically made for use with VOIP. While the IP phone looks exactly like a normal phone, it's got special Ethernet connectors that allow it to be plugged into your router. They're even working on WIFI phones for VOIP that you can take with you to the various internet hotspots popping up all over the world. Finally, you can make VOIP contact with your computer alone. Simply install the VOIP software, make sure you've got a microphone, speakers, an internet connection (high-speed is best, of course), and a sound card, and chat away. One thing many VOIP-users love about it is the cost, or more accurately, the savings. By using VOIP you save yourself one unnecessary bill per month - your phone bill. VOIP charges, much cheaper usually than most people's phone bills, appear on your regular broadband bill [2]. The technology underpinning VoIP was initially developed in the late 1970s, but it took almost 20 years to evolve from a computer novelty into a household service. It's now used by hundreds of thousands of people every day.
Obviously, the most important requirement is a broadband internet connection. Broadband connections are provided by cable companies (digital cable service), telephone companies (DSL, T1, etc.), and radio/microwave broadband internet connections. Currently, satellite (ie., satellite uplink dish) internet connections are not compatible with VOIP equipment because of the proprietary data compression algorithms used in satellite uplink and downlink. Further, the speed of light delay to and from a geosynchronous orbiting satellite would prove to be very annoying people trying to talk. Broadband connection data uplink and downlink speeds of greater than 80 kilobits per second per telephone circuit (while a call is in progress) are generally considered to be the minimum requirement for "decent" voice transmission quality. A "Telephone Adapter" (or "TA," and also known as an "Analog Telephone Adapter" or "ATA") is a piece of hardware that is used to digitize the voice and establish the IP session to the internet phone company’s network switch. While it is possible to use a computer’s microphone and speakers and special software for telephony over the Internet, the obvious limitation that the computer has to be turned on to make or receive phone calls makes this unwieldy. A TA eliminates the need for a computer to be up and running and accepts a standard 4-wire RJ11 telephone cable to
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support either premise telephone wiring or a direct connection of a standard analog telephone. In addition, the TA usually includes a built in "router" that provides firewall isolation for the computers connecting to the Internet, as well as a Local Area Network (LAN) switch or hub. This allows efficient Internet connection sharing. The internet phone company usually provides or rent the TA, or they can be purchased at retail for a reasonable price [4]. The third requirement is a VoIP Service Provider (VSP) also known as an Internet Telephony Service Provider (ITSP). The Provider will supply you with an account and some form of "Telephone Number" [5]. The final requirement is one or two common variety analog telephone handset. Almost all commonly available wireless telephones and most two line phone sets will work with VOIP [4]. A. All together First, confirm that you have the correct template for your paper size. This template has been tailored for output on the US-letter paper size. If you are using A4-sized paper, please close this file and download the file for “MSW A4 format”. The TA is usually a cable or DSL router that connects to the cable company’s or DSL provider’s supplied terminal (or "modem"). The customer’s computer is then connected to the TA, as are the one or two standard (RJ11) telephone cables, which connect to either a wall outlet or a standard analog telephone, depending on whether telephone extensions are present or not. More than one computer usually can be connected to the TA to create a Local Area Network (LAN).
The above diagram illustrates a sample installation. The telephone on the left is directly connected to the TA, while the one on the right is connected to the premise distribution wiring which is connected to the TA via RJ 11 cable to a wall jack. The key to making VOIP work is the correct initialization of the TA to work with the internet phone company and configuring IP addresses on the router to use for the computer or premise LAN connected computers sharing the broadband connection [6]. III. VOIP PROTOCOLS
To deliver voice, two types of VOIP protocol used: H.323 and SIP. H.323 and SIP both support VoIP and multimedia communications. H.323 is an older standard developed by the ITU. A good chunk of it is based on ISDN which comes from the traditional telephony world. H.323 is a binary protocol and is fairly complex in nature. SIP was developed by the Internet Engineering Task Force (IETF) and is text based (similar to HTTP). Much of the infrastructure already in place to support HTTP has been adapted to support SIP. IT managers within businesses are generally more comfortable with SIP because they are used to handling HTTP traffic. SIP is an open standard and solutions based on SIP are highly interoperable. A lot of effort has gone into ensuring interoperability and many manufacturers work together to regularly test to ensure this. Very few manufacturers are working on new H.323 implementations. SIP has become the standard of choice and is being worked on by large companies such as Microsoft and Cisco [7]. A. H.323 Protocol Overview H.323 is a ITU recommendation based on the H.320 family of standards. The current version of the recommendation is version 4 [8]. Initially, the protocol (version 1) was designed to provide signalling for a multimedia conferencing system for LAN environments with no quality of service provisions. However, in is current state, it has evolved into an umbrella of specifications that define the complete architecture and operation of a multimedia conferencing system over a wide area packet network. In contrast to its original scope, it has become a scalable solution that can be interworked with managed large scale networks. A H.323 system provides the necessary signalling and control operations for performing multimedia communications over an underlying packet based network which may not provide a guaranteed quality of service. The actual network interface, the physical network and the transport protocols used on the network are not included in the scope of H.323. A H.323 system comprises of the following entities: Terminals, Gatekeepers, Gateways, Multipoint Controllers, Multipoint Processors and Multipoint Control Units. • Terminals provide the audio/video/data communications capability in point-to-point or multipoint conferences, as well as handling the H.323 signalling issues on behalf of the user. • Gatekeepers provide admission control and address translation services
Figure 1. A sample installation of VOIP.
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• Gateways are needed to provide interworking with terminals using other signalling protocols, such as PSTN terminals, ISDN terminals, SIP terminals, etc. • Multipoint Controllers, Multipoint Processors and Multipoint Control Units provide support for multipoint conferences.
(IJCSIS) International Journal of Computer Science and Information Security, Vol. 8, No. 2, 2010
• A Redirect Server is a SIP server that provides address mapping services. It responds to a SIP request destined to an address with a list of new addresses. A redirect server doesn’t accept calls, doesn’t forward requests nor does it initiate any of its own.
A central aspect of H.323 is the H.323 call. It is defined as the point-to-point multimedia communication between two H.323 endpoints. If the H.323 endpoint communicates with an endpoint which uses a different signalling protocol, then the H.323 call is defined as the call segment between the H.323 entity and the gateway that provides interworking with the foreign network. The H.323 protocol is a tightly coupled family of sub protocols which must all interoperate in order to complete successfully a multimedia call session. The sub protocols are described in ITU recommendations. The main ones are: • H.225: Sub protocol for messages exchanged between H.323 endpoints for setting up and tearing down a call as well as for messages between an H.323 endpoint and its controlling H.323 entity, such as a gatekeeper. • H.245: Sub protocol for messages exchanged between endpoints in order to control the call session, exchange resource capabilities and establish media channels. • H.235: Sub protocol for security and encryption for H.323 terminals. • H.450: Sub protocols for supplementary services, such as Call Transfer, Call Park, Call Waiting etc [9]. B. SIP Protocol Overview SIP, which stands for Session Initiation Protocol, is an IETF application layer control protocol, defined in RFC 2543 [10], for the establishment, modification and termination of multimedia sessions with one or more participants. SIP makes minimal assumptions about the underlying transport and network layer protocol, which can provide either a packet or byte stream service with either reliable or unreliable service. A SIP system is based on a client/server model and is comprised of the following logical entities: • A User Agent (UA) is an application that acts on behalf of the user, both as a client (User Agent Client) and as a server (User Agent Server). As a client it initiates SIP requests and as a server it accepts calls and responds to SIP requests made by other entities. The user agent is usually part of a multimedia terminal whose media capabilities it controls without having any media capabilities of its own. • A Registrar Server is a SIP server that accepts only registration requests issued by user agents. A registrar server never forwards requests. • A Location Server is a server which provides information to a proxy/redirect server about the possible current locations of a user. Usually, this entity is part of the proxy/redirect servers.
• A Proxy Server is a SIP server that acts both as a server to user agents by forwarding SIP requests and as a client to other SIP servers by submitting the forwarded requests to them on behalf of user agents or proxy servers. With the exception of the user agent, which is usually part of a multimedia terminal, the rest of the logical entities (registrar, redirect and proxy servers)a may be combined in a single application. Therefore, a single entity can act either as a proxy or as a redirect server, according to the SIP request, and at the same time accept registration requests. A SIP call is defined as the multimedia conference consisting of all participants invited by a common source. Although not partitioned formally, the SIP system can be viewed as divided into domains each serviced by one redirect/proxy server and one registrar. A user agent has usually a home domain, which is specified by its address, but it can roam and use services in other domains as well, in which case it is considered to be ’visiting’. Otherwise it is considered to be ”at home” [9] C. Related Work- Comparison of two Protocols The authors of Nortel Networks [11] conclude by recommending SIP as their preference for a control protocol. They point out that even though H.323, unlike SIP, has currently more enterprise oriented and campus scale products deployed, SIP provides long term benefits which are related to and affect time to market, extensibility, multi-party service flexibility, ease of interoperability and complexity of development. The Dalgic and Fang [12] concluded that In terms of functionality and services that can be supported, H.323v3 and SIP are very similar. However, supplementary services in H.323 are more rigorously defined and therefore fewer interoperability issues are expected to arise. Furthermore, H.323 has better compatibility among its different versions and better interoperability with the PSTN. The two protocls are comparable in their QoS support (similar call setup delays, no support for resource reservation or class of service (QoS) setting), but H.323v3 will allow signaling of the requested QoS. On the other hand, according to the paper, SIP’s primary advantages are its flexibility to add new features and its relative ease of implementation and debugging. Finally, the authors note that H.323 and SIP are improving themselves by learning from each other, and the differences between them are diminishing with each new version. The Schulzrinne and Rosenberg [13] wrote that SIP provides a similar set of services to H.323, but provides far lower complexity, rich extensibility, and better scalability. They point out that future work is due to more fully evaluate the protocols, and examine quantitative performance metrics to characterize these differences. They also imply that a study
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measuring the processing overhead of SIP and H.323, would be quite useful [14]. IV. SECURITY
Viruses and malware VoIP utilization involving softphones and software are vulnerable to worms, viruses and malware, just like any Internet application. Since these softphone applications run on user systems like PCs and PDAs, they are exposed and vulnerable to malicious code attacks in voice applications. DoS (Denial of Service) A DoS attack is an attack on a network or device denying it of a service or connectivity. It can be done by consuming its bandwidth or overloading the network or the device’s internal resources. In VoIP, DoS attacks can be carried out by flooding a target with unnecessary SIP call-signaling messages, thereby degrading the service. This causes calls to drop prematurely and halts call processing. Why would someone launch a DoS attack? Once the target is denied of the service and ceases operating, the attacker can get remote control of the administrative facilities of the system. SPIT (Spamming over Internet Telephony) If you use email regularly, then you must know what spamming is. Put simply, spamming is actually sending emails to people against their will. These emails consist mainly of online sales calls. Spamming in VoIP is not very common yet, but is starting to be, especially with the emergence of VoIP as an industrial tool. Every VoIP account has an associated IP address. It is easy for spammers to send their messages (voicemails) to thousands of IP addresses. Voicemailing as a result will suffer. With spamming, voicemails will be clogged and more space as well as better voicemail management tools will be required. Moreover, spam messages can carry viruses and spyware along with them. This brings us to another flavor of SPIT, which is phishing over VoIP. Phishing attacks consist of sending a voicemail to a person, masquerading it with information from a party trustworthy to the receiver, like a bank or online paying service, making him think he is safe. The voicemail usually asks for confidential data like passwords or credit card numbers. You can imagine the rest! Call tampering Call tampering is an attack which involves tampering a phone call in progress. For example, the attacker can simply spoil the quality of the call by injecting noise packets in the communication stream. He can also withhold the delivery of packets so that the communication becomes spotty and the participants encounter long periods of silence during the call. Man-in-the-middle attacks VoIP is particularly vulnerable to man-in-the-middle attacks, in which the attacker intercepts call-signaling SIP message traffic and masquerades as the calling party to the called party, or vice versa. Once the attacker has gained this position, he can hijack calls via a redirection server [16].
A future expectation is that long-established security features (i.e., authentication and encryption) will be integrated into VoIP standards. However, today many existing datacentric security technologies can be utilized to enhance security in the VoIP environment. VoIP network security includes voice-packet security, which focuses on application concerns, while IP security focuses on transport or network security. Controlling security at these levels of the VoIP environment may require network re-design and/or re-engineering which will affect the architecture of the network supporting the VoIP environment. Some specific issues need further attention when a VoIP system is deployed. It is important to remember that securing any network is a continual process that requires staying abreast of the latest vulnerabilities that may exist in network infrastructure components, server operating systems, and applications deployed throughout the enterprise [15]. In the early days of VoIP, there was no big concern about security issues related to its use. People were mostly concerned with its cost, functionality and reliability. Now that VoIP is gaining wide acceptance and becoming one of the mainstream communication technologies, security has become a major issue. The security threats cause even more concern when we think that VoIP is in fact replacing the oldest and most secure communication system the world ever known – POTS (Plain Old Telephone System). A. Security Threats in VoIP Service theft can be exemplified by phreaking, which is a type of hacking that steals service from a service provider, or use service while passing the cost to another person. Encryption is not very common in SIP, which controls authentication over VoIP calls, so user credentials are vulnerable to theft. Eavesdropping is how most hackers steal credentials and other information. Through eavesdropping, a third party can obtain names, password and phone numbers, allowing them to gain control over voicemail, calling plan, call forwarding and billing information. This subsequently leads to service theft. Stealing credentials to make calls without paying is not the only reason behind identity theft. Many people do it to get important information like business data. A phreaker can change calling plans and packages and add more credit or make calls using the victim’s account. He can of course as well access confidential elements like voice mail, do personal things like change a call forwarding number. Vishing Vishing is another word for VoIP Phishing, which involves a party calling you faking a trustworthy organization (e.g. your bank) and requesting confidential and often critical information. Here is how you can avoid being a vishing victim.
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What Affects Voice Quality in VoIP Calls Here are the main things that affect voice quality in VoIP and what can be done to maximize quality. Bandwidth Your Internet connection always tops the list of factors affecting voice quality in VoIP conversations. The bandwidth you have for VoIP is the key for voice quality. For instance, if you have dial-up connection, don’t expect great quality. A broadband connection will work right, as long as it is not spotty, and not shared with too many other communication applications. Bandwidth dependency is besides one of the main drawbacks of VoIP. Equipment The VoIP hardware equipment you use can greatly impact on your quality. Poor quality equipment are normally the cheapest ones (but not always!). It is therefore always good to have as much information as possible on an ATA, router or IP phone before investing on it and starting to use it. Read reviews and discuss about it in forums. It might also be that the hardware you choose is the best in the world, but still you get problems - because you are not using hardware that suits your needs. ATA/Router For an ATA/Router, you need to think of the following: • Compression technologies (codecs) supported • Echo cancellation, which is a mechanism for decreasing echo • Firewall and security support Phone frequencies The frequency of your IP phone may cause interference with other VoIP equipment. There are many cases where people using 5.8 GHz phones have been getting voice quality problems. When all troubleshooting tricks failed, changing the phone to one with a lower frequency (e.g. 2.4 GHz) solved the problem. Weather Conditions At times, the voice is terribly distorted by something called static, which is a small 'dirty-weed' static electricity generated on broadband lines due to thunderstorms, heavy rain, strong gusts, electrical impulses etc. This static is not very much noticeable when you surf the net or download files, which is why we don’t complain about it when we use the Internet for data despite it be here; but when you are listening to voice, it becomes disturbing. It is easy to get rid of static: unplug your hardware (ATA, router or phone) and plug it back again. The static will be brought to naught. The effect of weather conditions on your connection is not something you can change. You can have some short-term relief in some cases, but most of the time, it is up to your service provider to do something. At times, changing the cables solves the problem completely, but this can be costly.
Location of your hardware Interference is a poison for voice quality during voice communication. Often, VoIP equipment interfere with each other thus producing noise and other problems. For example, if your ATA is too close to your broadband router, you might experience voice quality problems. This is caused by electrical feedback. Try moving them away from each other to get rid of the garbled calls, echoes, dropped calls etc. Compression: the codec used VoIP transmits voice data packets in a compressed form, so that the load to be transmitted is lighter. The compression software used for this are called codec’s. Some codecs are good while others are less good. Put simply, each codec is designed for a specific use. If a codec is used for a communication need other than that for which it is meant, quality will suffer. [17] V. VOIP MARKET
Telefonica has acquired VoIP provider Jajah, continuing the consolidation trend in the sector. Microsoft already bought Tellme, BT acquired Ribbit, Google took over GrandCentral and KPN is buying out the minority shareholders in iBasis. In addition, eBay sold a majority stake in Skype to Silver Lake Partners, which also has a stake in Avaya, while Skype and Avaya have started talks on working together. Another hardware manufacturer, Nortel, has received an early bid for its VoIP assets, from Genband. The trend shows newcomers slowly but surly losing their independence by joining larger groups. The latest takeover, Telefonica's acquisition of Jajah, is perhaps the most remarkable in that sense, as Jajah was originally set up to avoid the high international tariffs charged by the incumbents. However, Google and Skype are still maintaining their independence. There are still plenty of potential acquisition targets, at various stages of the VoIP value chain: iSkoot, Truphone, Jaxtr, Fring, Nimbuzz, Ooma, Vonage, 8x8 (Packet8), Rebtel, Freshtel, Mobivox, Sipgate, Vyke, Telio, Snapvine and many others. All these VoIP services providers combine infrastructure with services provision for end-users, and the question is what the new owners will do with acquired assets. The choice comes roughly down to wholesale (capacity, platform services, software) and retail (VoIP services for end-users). BT (Ribbit) and KPN (iBasis) are choosing clearly for the wholesale side, and that seems to also be the case with Telefonica (Jajah). Operators as well as large corporations can be offered VoIP services based on the acquired infrastructure, platforms and software. Google and Skype represent, as 'newcomers' on the telephony market, the retail side. As for Jajah, the question is whether Telefonica will maintain the end-user services, or slowly dissolve these in order to protect its own international business. There is a clear reason to keep this side of the Jajah business though: whatever Telefonica is losing on its home market, it can win back abroad by competing with incumbents elsewhere for international business.
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That still leaves the question of what's the future for newcomers such as Google and Skype. Pressure from VoIP will eventually drive all call costs to the price of local calls. After deducting termination fees, there is little left over for the provider. Call services will then become a true commodity business. That's the doomsday scenario for Skype, as it mainly makes money from avoiding high international call tariffs. Skype will need to build up quickly a large customer base, in order to offset the margin erosion. For Google, this is less of a problem, as it already has various services that do not directly earn money. For the incumbents, which are already seeing a sharp decline in their international business, there is a need to move further up the value chain. Companies such as BT and Telefonica can be expected to drive innovation on the telephony market going forward, with especially business customers profiting. Skype may also, in cooperation with Avaya target the business market, in order to exploit new income sources [18]. A. France led VoIP market in Europe in Q309 With more than 15 million subscribers, France is leading the Voice over IP (VoIP) market in Europe. Orange France is the telecom operator in Europe with the highest VoIP subscriber base with 6.580 million subscribers, it represents 17% of the European VoIP market. According to Dataxis Intelligence, in Q309 there were 39.7 million VoIP subscribers. This figure includes Voice over DSL, Cable and Fiber. The split is 76% over DSL, 21.8% over Cable and 2.2% over Fiber [19].
Suite, video, LTE trials and enhanced mobile IM and presence services. The IMS market will continue to be lumpy on a quarterly basis, but we expect continued positive momentum from new deployments from North American cable operators, Class 5 replacement projects in EMEA and VoLTE to contribute to strong annual growth for at least the next five years," forecasts Diane Myers, directing analyst for service provider VoIP and IMS at Infonetics Research. C. IMS MARKET HIGHLIGHTS First, confirm that you have the correct template for your paper size. This template has been tailored for output on the US-letter paper size. If you are using A4-sized paper, please close this file and download the file for “MSW A4 format”. Worldwide IP Multimedia Subsystem (IMS) equipment vendor revenue totaled $426 million in 2009 and is forecast to grow to $1.44 billion in 2014 Following a down 3Q09, the worldwide IMS equipment market posted its strongest quarter to date, jumping 92% sequentially in 4Q09 4Q09 marked the first quarter in which revenue from IMS equipment for mobile networks surpassed that of IMS equipment for fixed-line networks Alcatel-Lucent and Nokia Siemens Networks each posted very strong IMS equipment results in 4Q09 With key operators and vendors forming the OneVoice initiative and transferring the initiative to the GSMA in February 2010, IMS is guaranteed to get its biggest driver from LTE deployments starting in 2012
Figure 2. VoIP market in Europe in Q309- Source : Dataxis Intelligence
B. SERVICE PROVIDER VOIP AND IMS "The worldwide carrier voice over IP equipment market closed the 2009 calendar year down 28%, as expected, ending with its third consecutive quarter of stable revenue in the fourth quarter, led by very strong session border controller sales. Meanwhile, the IMS equipment market ended on a high note with 2009 worldwide revenue up 142% over 2008. The shining star in 4Q09 for IMS sales were deployments for mobile networks, particularly purchases for Rich Communication
Figure 3. Worldwide IMS growth.
D.
SERVICE PROVIDER VOIP MARKET HIGHLIGHTS The worldwide service provider VoIP equipment market dropped 28.2% from 2008 to 2009, to $2.48 billion worldwide
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The migration to native IP sped up during the past 12 months due to the acceleration of TDM access line loss, resulting in significant declines in TDM-related equipment, particularly traditional trunk media gateways and softswitches In 4Q09, service provider VoIP equipment revenue was up slightly, at 2.7% over 3Q09, continuing the period of relative stability that started in 2Q09 Four vendors stood out for growing revenue in 2009: o o o o Metaswitch in trunk media gateways and softswitches Acme Packet in session border controllers Radisys in media servers BroadSoft in voice application servers
demands have to rise. A VOIP user can call any other user, located anywhere in the world, with better voice quality. That is ideal for VOIP service. For next step the equipment will be more acceptable and technology has to present more power and security enhances user’s trust. REFERENCES
[1] [2] [3] [4] [5] [6] [7] [8] http://www.comtest.com/tutorials/VoIP.html http://articles.directorym.com/Importance_Of_VOIP-a971521.html Ian Grayson, 2009, www.cnet.com.au/voip-guide-voice-over-ip-inaustralia-240056481.htm http://www.zoesnet.net/VOIP.htm http://www.thevoipstore.net/VoIP-Requirements.php http://www.zoesnet.net/VOIP.htm http://searchunifiedcommunications.techtarget.com/expert/Knowledgeba seAnswer/0,289625,sid186_gci1069115,00.html I. T. Union, “Packet-based multimedia communication systems,” Telecommunication Standardization Sector of ITU, Geneva, Switzerland, Recommendation H.323, Nov. 17, 2000. Papageorgiou Pavlos ,2001, A Comparison of H.323 vs SIP M. Handley, H. Schulzrinne, E. Schooler, and J. Rosenberg, “SIP: Session Initiation Protocol,” Internet Engineering Task Force, Request For Comments 2543, Mar. 1999, proposed Standard. N. Networks, “A Comparison of H.323v4 and SIP,” 3GPP S2, Tokyo, Japan, Technical Report S2-000505, Jan. 5 2000. I. Dalgic and H. Fang, “Comparison of H.323 and SIP for IP Telephony Signaling,” in Proceedings of SPIE. Multimedia Systems and Applications II, ser. Proceedings of Photonics East, Tescher, Vasudev, Bove, and Derryberry, Eds., vol. 3845. Boston, Massachusetts. USA: The Internationl Society for Optical Engineering (SPIE), Sept. 20-22 1999. H. Schulzrinne and J. Rosenberg, “A Comparison of SIP and H.323 for Internet Telephony,” in Proceedings ofThe 8th International Workshop on Network and Operating Systems Support for Digital Audio and Video (NOSSDAV 98), Cambridge, UK, July 8-10 1998, pp. 83–86. Papageorgiou Pavlos, 2001, A Comparison of H.323 vs SIP, University of Maryland at College Park DISA for the DOD, 2006, Internet Protocol Telephony & Voice Over Internet Protocol, Version 2, Release 2 Nadeem Unuth, http://voip.about.com/od/security/a/SecuThreats.htm Nadeem Unuth, http://voip.about.com/od/voipbasics/a/factorsquality.htm telecompaper, 2009/12/30, http://en.c114.net/583/a472219.html Dataxis Intelligence, 2010, http://dataxisnews.com/?p=11055 Infonetics Research Report Highlights, 2010, http://www2.marketwire.com/mw/mmframe?prid=593499&attachid=11 91085
GENBAND’s pending acquisition of Nortel’s CVAS unit will cause some significant shifts in the vendor landscape, making GENBAND the largest carrier VoIP vendor in terms of overall revenue [20].
[9] [10]
[11] [12]
[13]
[14] [15] [16] [17] [18] [19] [20]
Figure 4. Worldwide Service Provider IMS growth.
VI.
CONCLUSION
VoIP is the voice of the future however many problems still remain.. VoIP phones are significantly less expensive than traditional phone lines. VoIP companies need some methods to make revenue, and they need a unique business plan; therefore
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A Study of Voice over Internet Protocol
Mohsen Gerami
The Faculty of Applied Science of Post and Communications Danesh Blv, Jenah Ave, Azadi Sqr, Tehran, Iran. Postal code: 1391637111
e-mail: [email protected]
Abstract—Voice over Internet Protocol, is an application that enables data packet networks to transport real time voice traffic. VOIP uses the Internet as the transmission network. This paper describes VoIP and its requirements. The paper further discusses various VoIP protocol, security and its market. Keywords: VOIP; H.323; SIP; Security; Market;
VoIP works in a relatively simple way. Each time you make a phone call your voice is converted into a stream of data. Then, rather than being sent over the phone network, this data stream travels over your broadband internet connection. Each data packet is labelled with its destination address (the person you're calling) and moves through the internet in the same way as web pages and file downloads. When they get to their destination, the packets are reassembled and converted back into sound waves. When you have this process happening simultaneously in two directions, you've got a phone call. Most VoIP services also come with an allocated landline phone number which allows other people to call you. In these cases the call will be routed to the nearest handover point (called a POP or point of presence) and then travel over the internet to your VoIP phone or computer [3]. II. REQUIREMENTS FOR VOIP
I.
INTRODUCTION
VoIP, or Voice over IP, is an application that enables data packet networks to transport real time voice traffic. It consists of hardware and software that allows companies and persons to engage in telephone conversations over data networks. As a result, more and more companies have become interested in implementing VoIP [1]. All VOIP services are not built alike. Some allow you to call anyone with a phone, while others restrict your calls to only other clients using the same VOIP service. You can choose between three different ways to set up a VOIP system on your computer. You can use an ATA (analog voice adaptor) which performs the analog-to-digital conversion, and is plugged in to your computer at one end and your telephone at the other. You can use an IP phone, a phone specifically made for use with VOIP. While the IP phone looks exactly like a normal phone, it's got special Ethernet connectors that allow it to be plugged into your router. They're even working on WIFI phones for VOIP that you can take with you to the various internet hotspots popping up all over the world. Finally, you can make VOIP contact with your computer alone. Simply install the VOIP software, make sure you've got a microphone, speakers, an internet connection (high-speed is best, of course), and a sound card, and chat away. One thing many VOIP-users love about it is the cost, or more accurately, the savings. By using VOIP you save yourself one unnecessary bill per month - your phone bill. VOIP charges, much cheaper usually than most people's phone bills, appear on your regular broadband bill [2]. The technology underpinning VoIP was initially developed in the late 1970s, but it took almost 20 years to evolve from a computer novelty into a household service. It's now used by hundreds of thousands of people every day.
Obviously, the most important requirement is a broadband internet connection. Broadband connections are provided by cable companies (digital cable service), telephone companies (DSL, T1, etc.), and radio/microwave broadband internet connections. Currently, satellite (ie., satellite uplink dish) internet connections are not compatible with VOIP equipment because of the proprietary data compression algorithms used in satellite uplink and downlink. Further, the speed of light delay to and from a geosynchronous orbiting satellite would prove to be very annoying people trying to talk. Broadband connection data uplink and downlink speeds of greater than 80 kilobits per second per telephone circuit (while a call is in progress) are generally considered to be the minimum requirement for "decent" voice transmission quality. A "Telephone Adapter" (or "TA," and also known as an "Analog Telephone Adapter" or "ATA") is a piece of hardware that is used to digitize the voice and establish the IP session to the internet phone company’s network switch. While it is possible to use a computer’s microphone and speakers and special software for telephony over the Internet, the obvious limitation that the computer has to be turned on to make or receive phone calls makes this unwieldy. A TA eliminates the need for a computer to be up and running and accepts a standard 4-wire RJ11 telephone cable to
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support either premise telephone wiring or a direct connection of a standard analog telephone. In addition, the TA usually includes a built in "router" that provides firewall isolation for the computers connecting to the Internet, as well as a Local Area Network (LAN) switch or hub. This allows efficient Internet connection sharing. The internet phone company usually provides or rent the TA, or they can be purchased at retail for a reasonable price [4]. The third requirement is a VoIP Service Provider (VSP) also known as an Internet Telephony Service Provider (ITSP). The Provider will supply you with an account and some form of "Telephone Number" [5]. The final requirement is one or two common variety analog telephone handset. Almost all commonly available wireless telephones and most two line phone sets will work with VOIP [4]. A. All together First, confirm that you have the correct template for your paper size. This template has been tailored for output on the US-letter paper size. If you are using A4-sized paper, please close this file and download the file for “MSW A4 format”. The TA is usually a cable or DSL router that connects to the cable company’s or DSL provider’s supplied terminal (or "modem"). The customer’s computer is then connected to the TA, as are the one or two standard (RJ11) telephone cables, which connect to either a wall outlet or a standard analog telephone, depending on whether telephone extensions are present or not. More than one computer usually can be connected to the TA to create a Local Area Network (LAN).
The above diagram illustrates a sample installation. The telephone on the left is directly connected to the TA, while the one on the right is connected to the premise distribution wiring which is connected to the TA via RJ 11 cable to a wall jack. The key to making VOIP work is the correct initialization of the TA to work with the internet phone company and configuring IP addresses on the router to use for the computer or premise LAN connected computers sharing the broadband connection [6]. III. VOIP PROTOCOLS
To deliver voice, two types of VOIP protocol used: H.323 and SIP. H.323 and SIP both support VoIP and multimedia communications. H.323 is an older standard developed by the ITU. A good chunk of it is based on ISDN which comes from the traditional telephony world. H.323 is a binary protocol and is fairly complex in nature. SIP was developed by the Internet Engineering Task Force (IETF) and is text based (similar to HTTP). Much of the infrastructure already in place to support HTTP has been adapted to support SIP. IT managers within businesses are generally more comfortable with SIP because they are used to handling HTTP traffic. SIP is an open standard and solutions based on SIP are highly interoperable. A lot of effort has gone into ensuring interoperability and many manufacturers work together to regularly test to ensure this. Very few manufacturers are working on new H.323 implementations. SIP has become the standard of choice and is being worked on by large companies such as Microsoft and Cisco [7]. A. H.323 Protocol Overview H.323 is a ITU recommendation based on the H.320 family of standards. The current version of the recommendation is version 4 [8]. Initially, the protocol (version 1) was designed to provide signalling for a multimedia conferencing system for LAN environments with no quality of service provisions. However, in is current state, it has evolved into an umbrella of specifications that define the complete architecture and operation of a multimedia conferencing system over a wide area packet network. In contrast to its original scope, it has become a scalable solution that can be interworked with managed large scale networks. A H.323 system provides the necessary signalling and control operations for performing multimedia communications over an underlying packet based network which may not provide a guaranteed quality of service. The actual network interface, the physical network and the transport protocols used on the network are not included in the scope of H.323. A H.323 system comprises of the following entities: Terminals, Gatekeepers, Gateways, Multipoint Controllers, Multipoint Processors and Multipoint Control Units. • Terminals provide the audio/video/data communications capability in point-to-point or multipoint conferences, as well as handling the H.323 signalling issues on behalf of the user. • Gatekeepers provide admission control and address translation services
Figure 1. A sample installation of VOIP.
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• Gateways are needed to provide interworking with terminals using other signalling protocols, such as PSTN terminals, ISDN terminals, SIP terminals, etc. • Multipoint Controllers, Multipoint Processors and Multipoint Control Units provide support for multipoint conferences.
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• A Redirect Server is a SIP server that provides address mapping services. It responds to a SIP request destined to an address with a list of new addresses. A redirect server doesn’t accept calls, doesn’t forward requests nor does it initiate any of its own.
A central aspect of H.323 is the H.323 call. It is defined as the point-to-point multimedia communication between two H.323 endpoints. If the H.323 endpoint communicates with an endpoint which uses a different signalling protocol, then the H.323 call is defined as the call segment between the H.323 entity and the gateway that provides interworking with the foreign network. The H.323 protocol is a tightly coupled family of sub protocols which must all interoperate in order to complete successfully a multimedia call session. The sub protocols are described in ITU recommendations. The main ones are: • H.225: Sub protocol for messages exchanged between H.323 endpoints for setting up and tearing down a call as well as for messages between an H.323 endpoint and its controlling H.323 entity, such as a gatekeeper. • H.245: Sub protocol for messages exchanged between endpoints in order to control the call session, exchange resource capabilities and establish media channels. • H.235: Sub protocol for security and encryption for H.323 terminals. • H.450: Sub protocols for supplementary services, such as Call Transfer, Call Park, Call Waiting etc [9]. B. SIP Protocol Overview SIP, which stands for Session Initiation Protocol, is an IETF application layer control protocol, defined in RFC 2543 [10], for the establishment, modification and termination of multimedia sessions with one or more participants. SIP makes minimal assumptions about the underlying transport and network layer protocol, which can provide either a packet or byte stream service with either reliable or unreliable service. A SIP system is based on a client/server model and is comprised of the following logical entities: • A User Agent (UA) is an application that acts on behalf of the user, both as a client (User Agent Client) and as a server (User Agent Server). As a client it initiates SIP requests and as a server it accepts calls and responds to SIP requests made by other entities. The user agent is usually part of a multimedia terminal whose media capabilities it controls without having any media capabilities of its own. • A Registrar Server is a SIP server that accepts only registration requests issued by user agents. A registrar server never forwards requests. • A Location Server is a server which provides information to a proxy/redirect server about the possible current locations of a user. Usually, this entity is part of the proxy/redirect servers.
• A Proxy Server is a SIP server that acts both as a server to user agents by forwarding SIP requests and as a client to other SIP servers by submitting the forwarded requests to them on behalf of user agents or proxy servers. With the exception of the user agent, which is usually part of a multimedia terminal, the rest of the logical entities (registrar, redirect and proxy servers)a may be combined in a single application. Therefore, a single entity can act either as a proxy or as a redirect server, according to the SIP request, and at the same time accept registration requests. A SIP call is defined as the multimedia conference consisting of all participants invited by a common source. Although not partitioned formally, the SIP system can be viewed as divided into domains each serviced by one redirect/proxy server and one registrar. A user agent has usually a home domain, which is specified by its address, but it can roam and use services in other domains as well, in which case it is considered to be ’visiting’. Otherwise it is considered to be ”at home” [9] C. Related Work- Comparison of two Protocols The authors of Nortel Networks [11] conclude by recommending SIP as their preference for a control protocol. They point out that even though H.323, unlike SIP, has currently more enterprise oriented and campus scale products deployed, SIP provides long term benefits which are related to and affect time to market, extensibility, multi-party service flexibility, ease of interoperability and complexity of development. The Dalgic and Fang [12] concluded that In terms of functionality and services that can be supported, H.323v3 and SIP are very similar. However, supplementary services in H.323 are more rigorously defined and therefore fewer interoperability issues are expected to arise. Furthermore, H.323 has better compatibility among its different versions and better interoperability with the PSTN. The two protocls are comparable in their QoS support (similar call setup delays, no support for resource reservation or class of service (QoS) setting), but H.323v3 will allow signaling of the requested QoS. On the other hand, according to the paper, SIP’s primary advantages are its flexibility to add new features and its relative ease of implementation and debugging. Finally, the authors note that H.323 and SIP are improving themselves by learning from each other, and the differences between them are diminishing with each new version. The Schulzrinne and Rosenberg [13] wrote that SIP provides a similar set of services to H.323, but provides far lower complexity, rich extensibility, and better scalability. They point out that future work is due to more fully evaluate the protocols, and examine quantitative performance metrics to characterize these differences. They also imply that a study
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measuring the processing overhead of SIP and H.323, would be quite useful [14]. IV. SECURITY
Viruses and malware VoIP utilization involving softphones and software are vulnerable to worms, viruses and malware, just like any Internet application. Since these softphone applications run on user systems like PCs and PDAs, they are exposed and vulnerable to malicious code attacks in voice applications. DoS (Denial of Service) A DoS attack is an attack on a network or device denying it of a service or connectivity. It can be done by consuming its bandwidth or overloading the network or the device’s internal resources. In VoIP, DoS attacks can be carried out by flooding a target with unnecessary SIP call-signaling messages, thereby degrading the service. This causes calls to drop prematurely and halts call processing. Why would someone launch a DoS attack? Once the target is denied of the service and ceases operating, the attacker can get remote control of the administrative facilities of the system. SPIT (Spamming over Internet Telephony) If you use email regularly, then you must know what spamming is. Put simply, spamming is actually sending emails to people against their will. These emails consist mainly of online sales calls. Spamming in VoIP is not very common yet, but is starting to be, especially with the emergence of VoIP as an industrial tool. Every VoIP account has an associated IP address. It is easy for spammers to send their messages (voicemails) to thousands of IP addresses. Voicemailing as a result will suffer. With spamming, voicemails will be clogged and more space as well as better voicemail management tools will be required. Moreover, spam messages can carry viruses and spyware along with them. This brings us to another flavor of SPIT, which is phishing over VoIP. Phishing attacks consist of sending a voicemail to a person, masquerading it with information from a party trustworthy to the receiver, like a bank or online paying service, making him think he is safe. The voicemail usually asks for confidential data like passwords or credit card numbers. You can imagine the rest! Call tampering Call tampering is an attack which involves tampering a phone call in progress. For example, the attacker can simply spoil the quality of the call by injecting noise packets in the communication stream. He can also withhold the delivery of packets so that the communication becomes spotty and the participants encounter long periods of silence during the call. Man-in-the-middle attacks VoIP is particularly vulnerable to man-in-the-middle attacks, in which the attacker intercepts call-signaling SIP message traffic and masquerades as the calling party to the called party, or vice versa. Once the attacker has gained this position, he can hijack calls via a redirection server [16].
A future expectation is that long-established security features (i.e., authentication and encryption) will be integrated into VoIP standards. However, today many existing datacentric security technologies can be utilized to enhance security in the VoIP environment. VoIP network security includes voice-packet security, which focuses on application concerns, while IP security focuses on transport or network security. Controlling security at these levels of the VoIP environment may require network re-design and/or re-engineering which will affect the architecture of the network supporting the VoIP environment. Some specific issues need further attention when a VoIP system is deployed. It is important to remember that securing any network is a continual process that requires staying abreast of the latest vulnerabilities that may exist in network infrastructure components, server operating systems, and applications deployed throughout the enterprise [15]. In the early days of VoIP, there was no big concern about security issues related to its use. People were mostly concerned with its cost, functionality and reliability. Now that VoIP is gaining wide acceptance and becoming one of the mainstream communication technologies, security has become a major issue. The security threats cause even more concern when we think that VoIP is in fact replacing the oldest and most secure communication system the world ever known – POTS (Plain Old Telephone System). A. Security Threats in VoIP Service theft can be exemplified by phreaking, which is a type of hacking that steals service from a service provider, or use service while passing the cost to another person. Encryption is not very common in SIP, which controls authentication over VoIP calls, so user credentials are vulnerable to theft. Eavesdropping is how most hackers steal credentials and other information. Through eavesdropping, a third party can obtain names, password and phone numbers, allowing them to gain control over voicemail, calling plan, call forwarding and billing information. This subsequently leads to service theft. Stealing credentials to make calls without paying is not the only reason behind identity theft. Many people do it to get important information like business data. A phreaker can change calling plans and packages and add more credit or make calls using the victim’s account. He can of course as well access confidential elements like voice mail, do personal things like change a call forwarding number. Vishing Vishing is another word for VoIP Phishing, which involves a party calling you faking a trustworthy organization (e.g. your bank) and requesting confidential and often critical information. Here is how you can avoid being a vishing victim.
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What Affects Voice Quality in VoIP Calls Here are the main things that affect voice quality in VoIP and what can be done to maximize quality. Bandwidth Your Internet connection always tops the list of factors affecting voice quality in VoIP conversations. The bandwidth you have for VoIP is the key for voice quality. For instance, if you have dial-up connection, don’t expect great quality. A broadband connection will work right, as long as it is not spotty, and not shared with too many other communication applications. Bandwidth dependency is besides one of the main drawbacks of VoIP. Equipment The VoIP hardware equipment you use can greatly impact on your quality. Poor quality equipment are normally the cheapest ones (but not always!). It is therefore always good to have as much information as possible on an ATA, router or IP phone before investing on it and starting to use it. Read reviews and discuss about it in forums. It might also be that the hardware you choose is the best in the world, but still you get problems - because you are not using hardware that suits your needs. ATA/Router For an ATA/Router, you need to think of the following: • Compression technologies (codecs) supported • Echo cancellation, which is a mechanism for decreasing echo • Firewall and security support Phone frequencies The frequency of your IP phone may cause interference with other VoIP equipment. There are many cases where people using 5.8 GHz phones have been getting voice quality problems. When all troubleshooting tricks failed, changing the phone to one with a lower frequency (e.g. 2.4 GHz) solved the problem. Weather Conditions At times, the voice is terribly distorted by something called static, which is a small 'dirty-weed' static electricity generated on broadband lines due to thunderstorms, heavy rain, strong gusts, electrical impulses etc. This static is not very much noticeable when you surf the net or download files, which is why we don’t complain about it when we use the Internet for data despite it be here; but when you are listening to voice, it becomes disturbing. It is easy to get rid of static: unplug your hardware (ATA, router or phone) and plug it back again. The static will be brought to naught. The effect of weather conditions on your connection is not something you can change. You can have some short-term relief in some cases, but most of the time, it is up to your service provider to do something. At times, changing the cables solves the problem completely, but this can be costly.
Location of your hardware Interference is a poison for voice quality during voice communication. Often, VoIP equipment interfere with each other thus producing noise and other problems. For example, if your ATA is too close to your broadband router, you might experience voice quality problems. This is caused by electrical feedback. Try moving them away from each other to get rid of the garbled calls, echoes, dropped calls etc. Compression: the codec used VoIP transmits voice data packets in a compressed form, so that the load to be transmitted is lighter. The compression software used for this are called codec’s. Some codecs are good while others are less good. Put simply, each codec is designed for a specific use. If a codec is used for a communication need other than that for which it is meant, quality will suffer. [17] V. VOIP MARKET
Telefonica has acquired VoIP provider Jajah, continuing the consolidation trend in the sector. Microsoft already bought Tellme, BT acquired Ribbit, Google took over GrandCentral and KPN is buying out the minority shareholders in iBasis. In addition, eBay sold a majority stake in Skype to Silver Lake Partners, which also has a stake in Avaya, while Skype and Avaya have started talks on working together. Another hardware manufacturer, Nortel, has received an early bid for its VoIP assets, from Genband. The trend shows newcomers slowly but surly losing their independence by joining larger groups. The latest takeover, Telefonica's acquisition of Jajah, is perhaps the most remarkable in that sense, as Jajah was originally set up to avoid the high international tariffs charged by the incumbents. However, Google and Skype are still maintaining their independence. There are still plenty of potential acquisition targets, at various stages of the VoIP value chain: iSkoot, Truphone, Jaxtr, Fring, Nimbuzz, Ooma, Vonage, 8x8 (Packet8), Rebtel, Freshtel, Mobivox, Sipgate, Vyke, Telio, Snapvine and many others. All these VoIP services providers combine infrastructure with services provision for end-users, and the question is what the new owners will do with acquired assets. The choice comes roughly down to wholesale (capacity, platform services, software) and retail (VoIP services for end-users). BT (Ribbit) and KPN (iBasis) are choosing clearly for the wholesale side, and that seems to also be the case with Telefonica (Jajah). Operators as well as large corporations can be offered VoIP services based on the acquired infrastructure, platforms and software. Google and Skype represent, as 'newcomers' on the telephony market, the retail side. As for Jajah, the question is whether Telefonica will maintain the end-user services, or slowly dissolve these in order to protect its own international business. There is a clear reason to keep this side of the Jajah business though: whatever Telefonica is losing on its home market, it can win back abroad by competing with incumbents elsewhere for international business.
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That still leaves the question of what's the future for newcomers such as Google and Skype. Pressure from VoIP will eventually drive all call costs to the price of local calls. After deducting termination fees, there is little left over for the provider. Call services will then become a true commodity business. That's the doomsday scenario for Skype, as it mainly makes money from avoiding high international call tariffs. Skype will need to build up quickly a large customer base, in order to offset the margin erosion. For Google, this is less of a problem, as it already has various services that do not directly earn money. For the incumbents, which are already seeing a sharp decline in their international business, there is a need to move further up the value chain. Companies such as BT and Telefonica can be expected to drive innovation on the telephony market going forward, with especially business customers profiting. Skype may also, in cooperation with Avaya target the business market, in order to exploit new income sources [18]. A. France led VoIP market in Europe in Q309 With more than 15 million subscribers, France is leading the Voice over IP (VoIP) market in Europe. Orange France is the telecom operator in Europe with the highest VoIP subscriber base with 6.580 million subscribers, it represents 17% of the European VoIP market. According to Dataxis Intelligence, in Q309 there were 39.7 million VoIP subscribers. This figure includes Voice over DSL, Cable and Fiber. The split is 76% over DSL, 21.8% over Cable and 2.2% over Fiber [19].
Suite, video, LTE trials and enhanced mobile IM and presence services. The IMS market will continue to be lumpy on a quarterly basis, but we expect continued positive momentum from new deployments from North American cable operators, Class 5 replacement projects in EMEA and VoLTE to contribute to strong annual growth for at least the next five years," forecasts Diane Myers, directing analyst for service provider VoIP and IMS at Infonetics Research. C. IMS MARKET HIGHLIGHTS First, confirm that you have the correct template for your paper size. This template has been tailored for output on the US-letter paper size. If you are using A4-sized paper, please close this file and download the file for “MSW A4 format”. Worldwide IP Multimedia Subsystem (IMS) equipment vendor revenue totaled $426 million in 2009 and is forecast to grow to $1.44 billion in 2014 Following a down 3Q09, the worldwide IMS equipment market posted its strongest quarter to date, jumping 92% sequentially in 4Q09 4Q09 marked the first quarter in which revenue from IMS equipment for mobile networks surpassed that of IMS equipment for fixed-line networks Alcatel-Lucent and Nokia Siemens Networks each posted very strong IMS equipment results in 4Q09 With key operators and vendors forming the OneVoice initiative and transferring the initiative to the GSMA in February 2010, IMS is guaranteed to get its biggest driver from LTE deployments starting in 2012
Figure 2. VoIP market in Europe in Q309- Source : Dataxis Intelligence
B. SERVICE PROVIDER VOIP AND IMS "The worldwide carrier voice over IP equipment market closed the 2009 calendar year down 28%, as expected, ending with its third consecutive quarter of stable revenue in the fourth quarter, led by very strong session border controller sales. Meanwhile, the IMS equipment market ended on a high note with 2009 worldwide revenue up 142% over 2008. The shining star in 4Q09 for IMS sales were deployments for mobile networks, particularly purchases for Rich Communication
Figure 3. Worldwide IMS growth.
D.
SERVICE PROVIDER VOIP MARKET HIGHLIGHTS The worldwide service provider VoIP equipment market dropped 28.2% from 2008 to 2009, to $2.48 billion worldwide
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The migration to native IP sped up during the past 12 months due to the acceleration of TDM access line loss, resulting in significant declines in TDM-related equipment, particularly traditional trunk media gateways and softswitches In 4Q09, service provider VoIP equipment revenue was up slightly, at 2.7% over 3Q09, continuing the period of relative stability that started in 2Q09 Four vendors stood out for growing revenue in 2009: o o o o Metaswitch in trunk media gateways and softswitches Acme Packet in session border controllers Radisys in media servers BroadSoft in voice application servers
demands have to rise. A VOIP user can call any other user, located anywhere in the world, with better voice quality. That is ideal for VOIP service. For next step the equipment will be more acceptable and technology has to present more power and security enhances user’s trust. REFERENCES
[1] [2] [3] [4] [5] [6] [7] [8] http://www.comtest.com/tutorials/VoIP.html http://articles.directorym.com/Importance_Of_VOIP-a971521.html Ian Grayson, 2009, www.cnet.com.au/voip-guide-voice-over-ip-inaustralia-240056481.htm http://www.zoesnet.net/VOIP.htm http://www.thevoipstore.net/VoIP-Requirements.php http://www.zoesnet.net/VOIP.htm http://searchunifiedcommunications.techtarget.com/expert/Knowledgeba seAnswer/0,289625,sid186_gci1069115,00.html I. T. Union, “Packet-based multimedia communication systems,” Telecommunication Standardization Sector of ITU, Geneva, Switzerland, Recommendation H.323, Nov. 17, 2000. Papageorgiou Pavlos ,2001, A Comparison of H.323 vs SIP M. Handley, H. Schulzrinne, E. Schooler, and J. Rosenberg, “SIP: Session Initiation Protocol,” Internet Engineering Task Force, Request For Comments 2543, Mar. 1999, proposed Standard. N. Networks, “A Comparison of H.323v4 and SIP,” 3GPP S2, Tokyo, Japan, Technical Report S2-000505, Jan. 5 2000. I. Dalgic and H. Fang, “Comparison of H.323 and SIP for IP Telephony Signaling,” in Proceedings of SPIE. Multimedia Systems and Applications II, ser. Proceedings of Photonics East, Tescher, Vasudev, Bove, and Derryberry, Eds., vol. 3845. Boston, Massachusetts. USA: The Internationl Society for Optical Engineering (SPIE), Sept. 20-22 1999. H. Schulzrinne and J. Rosenberg, “A Comparison of SIP and H.323 for Internet Telephony,” in Proceedings ofThe 8th International Workshop on Network and Operating Systems Support for Digital Audio and Video (NOSSDAV 98), Cambridge, UK, July 8-10 1998, pp. 83–86. Papageorgiou Pavlos, 2001, A Comparison of H.323 vs SIP, University of Maryland at College Park DISA for the DOD, 2006, Internet Protocol Telephony & Voice Over Internet Protocol, Version 2, Release 2 Nadeem Unuth, http://voip.about.com/od/security/a/SecuThreats.htm Nadeem Unuth, http://voip.about.com/od/voipbasics/a/factorsquality.htm telecompaper, 2009/12/30, http://en.c114.net/583/a472219.html Dataxis Intelligence, 2010, http://dataxisnews.com/?p=11055 Infonetics Research Report Highlights, 2010, http://www2.marketwire.com/mw/mmframe?prid=593499&attachid=11 91085
GENBAND’s pending acquisition of Nortel’s CVAS unit will cause some significant shifts in the vendor landscape, making GENBAND the largest carrier VoIP vendor in terms of overall revenue [20].
[9] [10]
[11] [12]
[13]
[14] [15] [16] [17] [18] [19] [20]
Figure 4. Worldwide Service Provider IMS growth.
VI.
CONCLUSION
VoIP is the voice of the future however many problems still remain.. VoIP phones are significantly less expensive than traditional phone lines. VoIP companies need some methods to make revenue, and they need a unique business plan; therefore
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