Assignment+1+-+VoIP

** WIDE AREA NETWORK (BITS 2323)  ** ** ASSIGNMENT 1:  ** ** VOICE OVER INTERNET PROTOCOL (VoIP)  ** ** PREPARED FOR:   ** ** PUAN KHADIJAH BINTI WAN MOHD GHAZALI  ** ** SUBMISSION DATE:  **
 * NAME MATRIC NUMBER **
 * AHMAD FAIZ BIN MOHD B030810219  **
 * MUHAMMAD ISMAIL BIN IBRAHIM B030810153  **
 *  24 FEBRUARY 2009 **

__________________________________________________________________________________________________________________________ VoIP (pronounced voyp ) is the name of a new communications technology that changes the meaning of the phrase telephone call. VoIP stands for Voice over Internet Protocol and it means “voice transmitted over a computer network”. Internet Protocol (IP) networking is supported by all sorts of networks - corporate, private, public, cable, and even wireless networks. The term of Voice over IP (VoIP) does not refer to a single service but encompasses an entire collection of service that can fill the phone service needs of many different residential and business customers. VoIP can be use by a service provider to optimize its capability to carry many calls. VoIP can be use by a small and large business for their office phone system and also use as a good alternative to the public phone service. People might be using the VoIP and no even know or realize it. Many telephone service providers are starting to use some form of VoIP inside their network because of cost efficiencies can afford them. Many online chat services, such as Xbox, live voice chat, Skype, Yahoo Messenger and so on, rely on VoIP. VoIP have four standards which are: 1.1    ITU-T H.320 Standards for Video Conferencing 1.2    H.323 ITU Standards 1.3    H.324 ITU Standards 1.4    VPIM Technical Specification
 * 1.0   ****  Introduction  **

The main difference between traditional phone systems and VoIP systems is circuit switching versus packet switching. The public telephone network (PSTN) use circuit switching to carry your voice from your phone to person you are calling. This means that while you are in telephone, a connection made end to end through the phone system. This requires resource such as series of wires, switch and connection in the telephone network that are dedicated to the duration of your call. While users are using them, no one else can use them because the end to end is reserve for your conversion. This approach works well but when the resources that are required to carry millions of calls each day coast to coast. At first, each call required a separate set of copper wires, Technology got better and now millions of calls can be carried over fiber-optic cables but even though density improved, the basic principles of circuit switching still apply nowadays where each call consume a channel on a wire end to end for the duration call. Packet switching work differently. Instead of having dedicated connection end to end, packet switching breaks the voice conversion into pieces. Transmit the piece and the reassembles the pieces at the other side back into the voice conversation. Packet switching enables people to share the same resource at the same time. The important difference is that during a traditional phone call, users are using the dedicated circuit for the duration of your call and the transmission is constant. But in the packet switching the pieces of the conversation find their own way through the network and are re-assemble on the other end, which allow many conversations to make place than circuit switching. So lots of other folks can use the same circuit at the same time. Both diagram below show sample transmission using circuit switching and packet switching. ** Figure 2.2 – Packet Switching Sample Diagram**
 * 2.0   ****  VoIP Traffics (Circuit Switching VS Packet Switching)  **
 * Figure 2.1 – Circuit Switching Sample Diagram **

3.1    VoIP Signalling Signalling refers to how central office switch in the phone network communicates between itself and your phone, or to other switches in the network. You need to understand a few important signals. For VoIP to PSTN call transmission, the terminal adapters connect the handset on user’s house and connect to the broadband internet connection. The Terminal Adapter act as translator that converting the signal into VoIP signals by converting the analog signal into digital signal. When user lift the phone, instead of the central office recognizing that the phone is off the book, the terminal translates it to a message sent to the broadband phone provider that you want to place a call. And from that point, the signaling is similar to the PSTN signaling except at each step, the terminal adapter is translating whatever phone handset action into digital messages that are being sent over broadband internet connection to the broadband phone service provider’s softswitch which refer to a central office switch that only receive digital signal. The softswitch takes care of routing the call. Notice that in this case, the call still routed through the PSTN to the person that user a calling to and this done by a gateway between the broadband phone service and the PSTN that totally transparent for all user. For VoIP TO VoIP call transmission, the terminal adapter connects both phone to their broadband internet connections. The caller goes of the hook and dials the destination number. The softswitch that serves the caller routes the call to the other softswitch that serves the called number. The destination terminal adapter gets the digital message for the incoming call and converts it to the ring the other phone. When the called person picks up the phone, again the flurry of digital message are exchengeed between terminal adapters and softswitches on both end to establish the call transmission. Another important function to understand about the terminal adapter is that is converts voice conversation into packets that can be sent over the internet. 3.2    VoIP Carries a Conversation Human speech is made up of analog sound waves which can transmit using straightforward technique. For digital telephony (with VoIP futures), a dedicated circuit does not transmit the voice so human speech must be converting to a digital stream (1 & 0 bits) by the transmitter and then re-created on the receiving end. The analog-to-digital conversion is accomplished by sampling which is the process of taking many instantaneous measurement of an analog signal. To convert the wave of analog signal, the waveform is measured thousand of times per second. For every voltage level, a corresponding combination of 1s and 0s exists, and combination is sent across the digital network. This process of measuring and converting is called sampling. On the receiving end, the combination of 1s and 0s is read and the corresponding voltage is re-created. If enough sample are taken, the original analog signal can be nearly exactly replicated by “connecting the dots” of the instantaneous measurement re-created on the receiving end. The trick for “near except” replication of the original signal is to take the right amount of samples because too few sample can be result in multiple waveforms that could possibly connect the dots. Too many samples can provide fantastic sound quality but it also required much data transmission to be cost effective. The right amount turns out to be twice the rate of the highest frequency in the waveform with pure tone of 1000 Hz. After enough samples are taken, the data in the form of bytes is shoved into packet and sent on its way to the other phone. When the packet reach the other phone, the sampled data re-creates the original waveform which excites a diaphragm/speaker in order to be hear by the other end user. ** Figure 3.1 - General VoIP Service Works**
 * 3.0   ****  How VoIP Works  **

4.1    Lowering Monthly Phone Bill Four primary factors make broadband phone services so inexpensive: 4.1.1 Infrastructure Cost Broadband phone services do not have similar infrastructure cost. Instead these services are made possibly by strategically located VoIP gateways which translate between VoIP and PSTN systems throughout the geographic areas they are serving. Beyond that, when a person is using the internet connection, so they have no additional wiring costs. VoIP on the other hand uses a data network that much less expensive network to set up and maintain. If that wasn’t a good enough deal, the section of the network that is most expensive for the part that goes from the local office to house is owned and therefore maintained by a phone a cable companies. Because the network treats voice the same way that is treats data with unlimited uploading and downloading for a monthly fee. 4.1.2    Transport Cost Broadband phone services use the public internet as the primary transport because the conversion of your voice into packets works just like packets that carry email. It is easy to identify why no additional cost is incurred with a call from one VoIP phone to another. The Internet Service Provider (ISP) simply charges you a monthly fee for the internet connection. This structure does fall apart through if the network runs out of capacity someday because the ISP would have to spend money to upgrade their networks. With the relatively low bandwidth required for VoIP calls because capacity is not an issue today but if the internet videophones take off, we can see costs go up. The real trick to making VoIP worth a darn, however, is that the calls are still cheap when calling PSTN phones anywhere in the country and in some cases even international calls are relatively cheap. 4.1.3    Regulator Compliance Cost Broadband services are not classified by regulators in the same way as standard telephony. Instead, they are classified as data services, and therefore many of the same regulations do not apply because VoIP is not currently subject of the regulations, the recurring operational cost for VoIP provide the less than those of PSTN providers. 4.1.4    Taxes and Fee Broadband phone services are classified as data services and are therefore not subject to many of the same local, state and federal taxes. There are smaller countries that have regulations and fees that are hefty for VoIP. In most places VoIP is free, but if you know that you are going to contact a country with a large fee frequently, you might be better off keeping the landline. 4.2    Flexibility VoIP can facilitate tasks and provide services that may be more difficult to implement using the PSTN. Examples include: 4.2.1    The ability to transmit more than one telephone calls over the same broadband connection. This can make VoIP a simple way to add an extra telephone line to a home or office. 4.2.2    Secure calls using standardized protocols such as  [|Secure Real-time Transport Protocol]. Most of the difficulties of creating a [|secure phone connection] over traditional phone lines, like digitizing and digital transmission, are already in place with VoIP. It is only necessary to [|encrypt] and [|authenticate] the existing data stream. 4.2.3    Location independence. Only an Internet connection is needed to get a connection to a VoIP provider. For instance, call center agents using VoIP phones can work from anywhere with a sufficiently fast and stable Internet connection. 4.2.4    Integration with other services available over the Internet, including video conversation, message or data file exchange in parallel with the conversation, audio conferencing, managing address books, and passing information about whether others are available to interested parties.
 * 4.0   ****  Advantages of VoIP  **

Voice over Internet Protocol is one of the many telephony applications that can be used over Internet Protocol (IP). By operating VoIP protocols over Internet Protocol, IP users can place voice, fax, and modem calls over IP-based networks. VoIP uses a signaling protocol to establish sessions or conversations. Once VoIP endpoints agree on the parameters for a call, all media signals between the end-points are digitized, compressed, packetized, and exchanged as IP datagram. VoIP is a very different application from traditional Internet applications but it is simply another application. The figure below shown about the relationship between VoIP and TCP/IP architecture:
 * 5.0   ****  VoIP Architectures  **

** Figure 5.1 - Relationship between VoIP and TCP/IP architecture

**      5.1   Real-time Transport Protocol – an end-to-end delivery protocol for real-time media applications. 5.2  RTP Control Protocol – Part of the RTP specification that enables call quality monitoring like jitter, packet loss, and latency. 5.3  Session Description Protocol – a protocol that provides the means to specify the characteristics of a media session. 5.4  Codecs for the encoding and decoding of analog signals into digital signals. Today, our telephone conversations are fairly secure due to the parameter and physical security of the PSTN. Internet effectively removes both of these barriers, making eavesdropping and tapping a trivial effort in many cases. Because of this, we must add security measures to protect VoIP in the applications signaling and media protocols. Although there is still the threat of Time Division Multiplexing (TDM) toll fraud in legacy voice system, the security risks for misuse or attack from individuals or groups outside of a company are low. In contrast, with VoIP you inherit all of the risks of an IP network including viruses, targeted denial of services attacks, and the security of packetized voice conversations which are possible to intercept. So, to prevent all security threat goes into VoIP, there is several solution: ** 6.1  ****Locking Down Your System **  6.1.1  Preventing unauthorized access to the network is a smart first step voice security. For an additional layer of protection, in case somebody does gain unauthorized access, organizations can also encrypt voice traffic. Voice and video-enabled VPN (V3PN) technology are available in many routers and security appliances. It will encrypt voice as well as data traffic using IP Security (IPsec) or Advanced Encryption Standard (AES). Encryption is performed in hardware so that firewall performance is not affected.  6.1.2  Many security experts also recommend limiting VoIP data to a single virtual local area network (VLAN). A VLAN will keep voice network traffic hidden from data network users and providing an additional layer of security. The technique can also limit the scope of damage to the VLAN in the event of an attack. An additional side benefit is that a VLAN help prioritize VoIP data over other types of network traffic.  6.1.3  When creating the VLAN, be sure to place its equipment behind separate firewalls. This practice will restrict traffic crossing VLAN boundaries to applicable protocols and prevent viruses and other kinds of malware from spreading from clients to servers.  6.2  **Data and Physical Security **  6.2.1  VoIP transmission needs the packet data encryption to being safeguard. Yet call signaling encryption is important as well to prevent hackers from misdirecting or otherwise interfering with call traffic.  6.2.2  <span style="font-family: 'Calibri','sans-serif'; mso-ansi-language: EN;">A secure gateway that properly configured is a VoIP basic of the system. The gateway will limit system access to authenticated and approved users while keeping hackers safely on the outside. Gateways act as the networks that lie behind them can be protected through the use of a stateful package inspection (SPI) firewall and network address translation (NAT) tools. <span style="font-family: 'Calibri','sans-serif'; mso-bidi-font-family: Calibri; mso-fareast-font-family: Calibri; mso-ansi-language: EN;"> 6.3  **<span style="font-weight: normal; font-family: 'Calibri','sans-serif'; mso-ansi-language: EN; mso-bidi-font-weight: bold;">Eternal Vigilance **<span style="font-family: 'Calibri','sans-serif'; mso-ansi-language: EN;"> <span style="font-family: 'Calibri','sans-serif'; mso-bidi-font-family: Calibri; mso-fareast-font-family: Calibri; mso-ansi-language: EN;"> 6.3.1  <span style="font-family: 'Calibri','sans-serif'; mso-ansi-language: EN;">Be sure to install updates, particularly security patches, as soon as possible. This will prevent hacker attacks. It is also important to disable non-essential operating and application services. <span style="font-family: 'Calibri','sans-serif'; mso-bidi-font-family: Calibri; mso-fareast-font-family: Calibri; mso-ansi-language: EN;"> 6.3.2  <span style="font-family: 'Calibri','sans-serif'; mso-ansi-language: EN;">Ethernet ports are also prime hacker entry points. Using management tools, access to authenticated and pre-approved users and devices can be limited. <span style="font-family: 'Calibri','sans-serif'; mso-bidi-font-family: Calibri; mso-fareast-font-family: Calibri; mso-ansi-language: EN;"> 6.3.3  <span style="font-family: 'Calibri','sans-serif'; mso-ansi-language: EN;">Building redundancy into a VoIP system can help it better withstand hacker attacks as well as equipment failure. Multiple gateways, nodes, routers, servers and power supplies make a system more resilient and reliable.
 * 6.0 VoIP Security  **

**  7.0    ****  VoIP Protocols  ** 7.1 Remote Voice Protocol over IP (RVP/IP) <span style="font-size: 12pt; color: black; line-height: 150%; mso-fareast-font-family: 'Times New Roman';"> It is MCK Communications protocol for transporting digital telephony sessions over packet or circuit based data networks. The protocol is used primarily in MCK's Extender product family, which extends PBX services over Wide Area Networks (WANs). RVP provides facilities for connection establishment and configuration between a client (or remote station set) device and a server (or phone switch) device. RVP/IP uses TCP to transport signaling and control data, and UDP to transport voice data. <span style="font-size: 12pt; color: black; line-height: 150%; mso-fareast-font-family: 'Times New Roman';">7.2 Session Announcement Protocol ( <span style="font-size: 12pt; line-height: 150%; mso-fareast-font-family: 'Times New Roman'; mso-bidi-font-weight: bold;">SAPv2) <span style="font-size: 12pt; color: black; line-height: 150%; mso-fareast-font-family: 'Times New Roman';"> <span style="font-size: 12pt; color: black; line-height: 150%; mso-fareast-font-family: 'Times New Roman';"> SAP is an announcement protocol that is used by session directory clients. A SAP announcer periodically multicasts an announcement packet to a well-known multicast address and port. The announcement is multicast with the same scope as the session it is announcing, ensuring that the recipients of the announcement can also be potential recipients of the session the announcement describes (bandwidth and other such constraints permitting). This is also important for the scalability of the protocol, as it keeps local session announcements local. The following is the format of the SAP data packet. 7.3  Session Description Protocol (SDP) <span style="font-size: 12pt; color: black; line-height: 150%; mso-fareast-font-family: 'Times New Roman';"> The Session Description Protocol (SDP) describes multimedia sessions for the purpose of session announcement, session invitation and other forms of multimedia session initiation. On Internet Multicast backbone (Mbone) a session directory tool is used to advertise multimedia conferences and communicate the conference addresses and conference tool-specific information necessary for participation. The SDP does this. It communicates the existence of a session and conveys sufficient information to enable participation in the session. Many of the SDP messages are sent by periodically multicasting an announcement packet to a well-known multicast address and port using SAP (session announcement protocol). These messages are UDP packets with a SAP header and a text payload. The text payload is the SDP session description. Messages can also be sent using email or the WWW (World Wide Web). The SDP text messages include: 7.3.1  <span style="font-size: 12pt; color: black; line-height: 150%; mso-fareast-font-family: 'Times New Roman';">Session name and purpose 7.3.2  <span style="font-size: 12pt; color: black; line-height: 150%; mso-fareast-font-family: 'Times New Roman';">Time the session is active 7.3.3  <span style="font-size: 12pt; color: black; line-height: 150%; mso-fareast-font-family: 'Times New Roman';">Media comprising the session 7.3.4  <span style="font-size: 12pt; color: black; line-height: 150%; mso-fareast-font-family: 'Times New Roman';">Information to receive the media 7.4   Simple Gateway Control Protocol (SGCP) <span style="font-size: 12pt; color: black; line-height: 150%; mso-fareast-font-family: 'Times New Roman';"> Simple Gateway Control Protocol (SGCP) is used to control telephony gateways from external call control elements. A telephony gateway is a network element that provides conversion between the audio signals carried on telephone circuits and data packets carried over the Internet or over other packet networks. The SGCP assumes a call control architecture where the call control intelligence is outside the gateways and is handled by external call control elements. The SGCP assumes that these call control elements, or Call Agents, will synchronize with each other to send coherent commands to the gateways under their control. 7.5   Session Initiation Protocol (SIP) <span style="font-size: 12pt; color: black; line-height: 150%; mso-fareast-font-family: 'Times New Roman';"> Session Initiation Protocol (SIP) is an application layer control simple signalling protocol for VoIP implementations using the Redirect Mode. SIP is a textual client-server base protocol and provides the necessary protocol mechanisms so that the end user systems and proxy servers can provide different services: 7.5.1  <span style="font-size: 12pt; color: black; line-height: 150%; mso-fareast-font-family: 'Times New Roman';">Call forwarding in several scenarios: no answer, busy, unconditional, address manipulations 7.5.2  <span style="font-size: 12pt; color: black; line-height: 150%; mso-fareast-font-family: 'Times New Roman';">Callee and calling number identification 7.5.3  <span style="font-size: 12pt; color: black; line-height: 150%; mso-fareast-font-family: 'Times New Roman';">Personal mobility 7.5.4  <span style="font-size: 12pt; color: black; line-height: 150%; mso-fareast-font-family: 'Times New Roman';">Caller and callee authentication 7.5.5  <span style="font-size: 12pt; color: black; line-height: 150%; mso-fareast-font-family: 'Times New Roman';">Invitations to multicast conference 7.5.6  <span style="font-size: 12pt; color: black; line-height: 150%; mso-fareast-font-family: 'Times New Roman';">Basic Automatic Call Distribution (ACD) <span style="font-size: 12pt; color: black; line-height: 150%; mso-fareast-font-family: 'Times New Roman';"> SIP addresses (URL) can be embedded in Web pages and therefore can be integrated as part of powerful implementations. It using simple protocol and provides its own reliability mechanism. It creates, modifies and terminates sessions with one or more participants. These sessions include Internet multimedia conferences, Internet telephone calls and multimedia distribution. Members in a session can communicate using multicast or using a mesh of unicast relations or a combination of these. It supports user mobility by proxy and redirecting requests to the user’s current location. Users can register their current location. SIP is not tied to any particular conference control protocol. It is designed to be independent of the lower-layer transport protocol and can be extended with additional capabilities. SIP transparently supports name mapping and redirection services, allowing the implementation of ISDN and Intelligent Network telephony subscriber services. These facilities also enable personal mobility which is based on the use of a unique personal identity. SIP supports five facets of establishing and terminating multimedia communications: 7.5.7  <span style="font-size: 12pt; color: black; line-height: 150%; mso-fareast-font-family: 'Times New Roman';">User location 7.5.8  <span style="font-size: 12pt; color: black; line-height: 150%; mso-fareast-font-family: 'Times New Roman';">User capabilities 7.5.9  <span style="font-size: 12pt; color: black; line-height: 150%; mso-fareast-font-family: 'Times New Roman';">User availability 7.5.10  <span style="font-size: 12pt; color: black; line-height: 150%; mso-fareast-font-family: 'Times New Roman';">Call setup 7.5.11  <span style="font-size: 12pt; color: black; line-height: 150%; mso-fareast-font-family: 'Times New Roman';">Call handling 7.6   Skinny Client Control Protocol (SCCP) Skinny Client Control Protocol (SCCP). Telephony systems are moving to a common wiring plant. The end station of a LAN or IP- based PBX must be simple to use, familiar and relatively cheap. The H.323 recommendations are quite an expensive system. An H.323 proxy can be used to communicate with the Skinny Client using the SCCP. In such a case the telephone is a skinny client over IP, in the context of H.323. A proxy is used for the H.225 and H.245 signalling. The skinny client (i.e. an Ethernet Phone) uses TCP/IP to transmit and receive calls and RTP/UDP/IP to/from a Skinny Client or H.323 terminal for audio. Skinny messages are carried above TCP and use port 2000. The messages consist of Station message ID messages. **  8.0    ****  VoIP Quality of Services  ** Quality of services in VoIP is very important because it act as a tool to make sure the VoIP is success. The term of service is it generally means what is offered to consumers in communication facilities. The term of quality in VoIP is simply able to speak and listen in a clear and continuous voice without interference with other callers. To make it success, it depends on these following factors: 8.1    Data loss 8.2    Consistent delay and characteristics same as jitter 8.3    Latency For the quality of bandwidth, we have to think how to achieve good voice quality with limited and often shared bandwidth. Broadband phone service    enables voice telephone calls to work over your high-speed Internet connection. A broadband phone also known as a VoIP or Internet phone utilizes the same IP network as your Internet service. Hardware adapters connect a standard telephone to the high-speed Internet connection to create a broadband phone. Below are the following three primary type of broadband phone services exist: 9.1    Internet VoIP Phone Services Internet VoIP Phone Services include any service that can provide services over the internet. These services are very popular for providing low cost additional phone line to home. Typically the service has component like phone, terminal adapter, Wireless Router, Broadband modem that connect to broadband provider in order to access the Softswitch PSTN over the internet. In the customer home, it would have a terminal adapter which enables the phone to work with digital VoIP. A high-speed broadband internet access provides the connection to phone provider. While at the provider site, a softswitch typically act as the central office to route calls to and from phones. There is usually a subscriber database and a voice mail server as well as gateway to the public switched telephone network (PSTN) for “off-Net” calls which means the call that originate or terminate on a different network than the broadband phone service provider’s. With internet phone service, customer can assign a phone number and also can dialing a nearly identical to that of the PSTN. Features provided are also similar to PSTN, with the additional features describes the advantage of broadband phone services such as lowering monthly phone bill, phone number flexibility and online call management that already described above. 9.2    Cable VoIP Digital Phone Services Cable VoIP Digital Phone Services are similar to internet VoIP services but the primary difference is a cable provider which typically provides high-speed broadband internet access. Also provide VoIP-based digital phone service. These services are increasingly popular as a replacement for primary PSTN line. Typically the service has the components such as Wireless router, cable modem, phone that connect to PSTNs softswitch over cable provider through the internet. The broadband cable modem usually serves a dual role and also acting as the terminal adapter that convert the phone signal in home to a VoIP digital signal and a high-speed broadband access provides the connection to phone provider. Just like mention before, the internet VoIP provider use the softswitch typically act as the central office to route calls to and from phones. There is usually a subscriber database and a voice mail server as well as gateway to the public switched telephone network (PSTN) for “off-Net” calls which means the call that originate or terminate on a different network than the broadband phone service provider’s. Typically the advantages, services and features were same as internet phone service. The different a cable VoIP service is that high-speed broadband cable provider is also the VoIP provider but the calls are not typically transported across the public internet but instead are routed directly from the broadband cable network to the VoIP switch. Costs are depending on the futures included and whether unlimited local and long-distance calls are included. 9.3    VoIP Chat Services VoIP chat services are different from internet VoIP and cable VoIP services while those services operate similar to PSTN telephony services including being assigned a normal phone number, VoIP chat services work much more like instant messaging (IM) programs and are often referred to as PC-to-PC calling. These services are rapidly growing as a low cost method for international calling and are popular with teenagers for general unlimited talking with each other. Typically service has components such as computer, wireless router, broadband modem that directly connect to broadband provider in order to access PSTN’s softswitch through an internet. In this case, customers have no distinct terminal adapter in home because the terminal adapter function is already provided by a desktop or laptop computer. Another difference is that a computer also usually serves as a phone and it’s generally not a service to which customer would connect to the traditional PSTN phone. High-speed broadband internet access provides the connection to VoIP chat provider. With VoIP chat services, customer are usually not assigned a phone number because typically method of calling each other is to look up the person in the contact list in the computer and just click it to start conversation similar to IM services. With some services and customer can also call PSTN phone number as well. The features provided are typically similar to IM features, not the fancier PSTN features found in internet VoIP and cable VoIP services. Calls are free to other users on the service and customers are charge for “off-Net” calls. ** Figure 9.3 - VoIP Chat Services ** **  10.0    ****  Conclusion  ** As a conclusion, the VoIP data traffic continues to increase and surpass the voice traffic, the convergence and integration of these technologies will not only continue to improve, but also will pave the way for a truly unified and seamless means of communication. Implementing VoIP can provide significant benefits and savings to your company. VoIP calls can be high quality with cheap alternative to long-distance and especially international calling. Currently the volume of calls is very low, but if the technology begins to be widely deployed it is expected the telephone companies will react to protect their investments. So, for the feature of VoIP, it is calls CoIP. CoIP mean a communication over IP that text, voice and video are converging over the same set of protocols used by VoIP and that Instant Messenger will be the application that enables users to seamlessly move between them.
 * 9.0   ****  Inventory Of VoIP   **
 * Figure 9.1 - Internet VoIP Phone Services **
 * Figure 9.2 - Cable VoIP Digital Phone Services **

**  11.0    ****  References  ** http://www.voip-news.com/feature/top-security-threats-2008-012408/ http://www.voip-news.com/sp/snp/1/3i.htm Jim Doherty and Neil Anderson (2006). “Internet Phone Services Simplified” United States of America: Paul Boger Alan B. Johnson and David M. Piscitello (2006). “Understanding Voice over IP Security” London: ARTECH HOUSE Voice over Internet Protocol (VoIP).pdf