ABSTRACT

This paper is the first in a three part series that will ultimately detail the past, present and future of Voice over Internet Protocol (VoIP). The purpose of this paper is to detail the history of VoIP and explore this technology/industry by examining its technological history, cultural history and its economic history. For the sake of brevity, this paper (and the two that follow) will focus on this technology and its place in the business world. The goals for the second paper will be to focus on the present condition of VoIP in the business world and to draw connections to historic events. In a similar structure, the second paper will also focus on the technology, cultural and economic factors that define VoIP in current terms. In addition, the second paper will detail particular legal or ethical issues faced by the industry, the target audience for this industry, and will present the factors leading to an explanation the digital divide. The goals for the third paper will focus on the future possibilities of VoIP in the business world and will draw connections from both the past and present states of the technology/industry. That document will draw together the hypotheses presented by leaders in the industry and will also include my own analysis of the future of the technology. Furthermore, I will include hypothetical events that may affect the technology, our culture and the industrys economics. The use of VoIP by individual consumers was the beginning of a massive move from traditional telephone systems to a form of new media where voice and other forms of digital media could converge with an already established data network. Major advancements in the technology are the result of business development and adoption. This paper focuses on the history of VoIP and how this technology fits into the business setting.

INTRODUCTION
VoIP is a technology that allows telephone calls to be made over computer networks like the Internet. VoIP converts analog voice signals into digital data packets and supports real-time, two-way transmission of conversations using Internet Protocol (IP).

FIG 1: VOIP SYSTEM IMPLEMENTATION

History of voip
Now that the groundwork has been documented, we can examine the short brief of VoIP. From most accounts, VoIP started in February of 1995 by a small company in Israel called Vocaltec, Inc. Their product, InternetPhone, allowed one user to call another user via their computers, a microphone and a set of speakers. Additionally, this application/product only worked if both the caller and the receiver had the same software setup. By 1998 some entrepreneurs started to market PC-to-phone

and phone-to-phone VoIP solutions. The phone calls were marketed as Free nation-wide long distance calls. When the caller would start the call he/she had to listen to advertisements before the call was connected. Another development in 1998 was the hardwares foray into the market. There were three IP Switch manufactures that introduced VoIP switching software as a standard in their routing equipment. By the end of 1998 VoIP calls had yet to total 1% of all voice calls By 2000, VoIP calls accounted for 3% and by 2003 that number had jumped up to 25%.

Impact of VoIP on Networks

Although VoIP is based on packet switching technology and should be a more efficient transport format, in actual fact, it is more inefficient. This is so due to the pervasive use of SIP and Skype protocols, which are bandwidth hungry. VoIP adds significant traffic load and latency to the network, especially if the network was not planned with that application in mind. Network management is also more time consuming. For example, it becomes necessary to differentiate the different user groups. Increased quality of service monitoring is also needed to ensure service level agreements to subscribed customers.

Traditional PSTN Call vs. VoIP

Traditional telephony uses circuit switching technology while VoIP uses packet switching. In circuit-switched networks, network resources are dedicated to the circuit during the entire message, and the entire message follows the same path. In packetswitched networks, the message is broken into packets, each of which can take a different route to the destination, where the packets are recompiled into the original message. As such, packet switching is supposed to be a much more efficient and cost effective way of sending voice messages. VoIP calls have different routing arrangements, such as peer to peer and those set up and maintained by proxy servers. Proxy servers are logical intermediary entities that control 3 and process call requests on behalf of its group of user clients e.g. users on a LAN or within a company. A peer to peer VoIP call is a direct connection between two users. Party A calls party B, party B accepts the call and a VoIP session is established between the two users. Calls via proxy severs are established in two ways. In one scenario, Party A sends a call request for B to the proxy server, the proxy server sends As information to B and Bs information to A, and maintains the connection for the duration of the call.

2.1. Benefits
1. Analysts project that VoIP will reach over 12 millions of broadband-enabled UShouseholds and small businesses by 2008 . The number of IP station lines will grow from 3 millions in 2004 to over 18 millions by the end of 2008 at an annual growth rate of 50%.1 Now many cable operators, telecom companies, and independent VoIP providers have moving toward offering consumer and residential VoIP. The calls will travel over internet instead of traditional phone lines and are much cheaper. Such service works with most cable modems and DSL broadband connections, which in turn is compatible for most home computer networks. Specifically, VoIP offers the following benefits:

2. Reduced Cost VoIP offers efficient use of bandwidth and require fewer longdistance trunks between switches [29]. The traditional circuit-switched networks, or the PSTN, have to dedicate a full-duplex 64Kbps channel for the duration of a single call. In contrast, the VoIP networks only require approximately 14Kbps, as voice compression is employed, and the bandwidth is used only when something has to be transmitted. Because the voice is delivered over the same data channels as any other data, it results in more network traffic over fewer leased lines. In addition, PSTN requires expensive tandem/toll switching capability, while using IP networks can bypass local and long-distance tolls [4]. Therefore, VoIP is likely to be used first in places where cost savings are significant. An example might be a company with highly distributed sites worldwide connected through a private or public IP network. 3. Unique Applications and Telephony Features Digitized data is easy to manipulate and analyze; the possibilities for new or enhanced applications and features are tremendous. For example, IP telephony can be integrated into interactive web pages, where a user clicks an icon to initiate a phone call. Businesses can utilize this feature to promote their products on the Internet. 4. Utilizing Wireless LAN Wireless LAN (WLAN), emerging as a mainstream technology in many organizations, can soon be upgraded to handle both voice and data traffic. While the cost of wireless long-distance services remains high, Wireless VoIP (WVoIP) could provide users the option of connecting to IP networks for long-distance calls. At the Spring VON 2004 conference, many leading handset providers, such as Motorola, Ericsson, Nokia, demonstrated combination wireless VoIP/cell phones. With such devices, users can make call via Wi-Fi (IEEE 802.11b) where WLAN is available, and make call via cellular in other places.

2.2. Challenges
However, several factors hinder VoIP adoption into full swing. Primarily, migrating from traditional reliable PBX system toward a less-proven IP communication system presents unacceptable risks for many telecommunications and data managers. Moreover, many IP telephony standardsin such key areas as signaling, quality of service (QOS), security and power over Ethernet and phone setshave not yet been definitively established Three critical areas need to be addressed before VoIP become widely accepted and integrated: 1. Infrastructure Reliability Since voice and data are all integrated on a data network, high reliability and survivability of the network infrastructure is strictly demanded. 2. Return on Investment Even though cost saving motivates enterprises to deploy VoIP, it is still difficult to evaluate the total cost of ownership (TCO) and return

on investment (ROI) for individual enterprises. Organizations must consider whether they need to upgrade their network infrastructure to meet the bandwidth requirement for IP telephony, and whether IP telephony can bring new beneficial services and features. In addition, VoIP end-user equipment costs more than traditional phones. Rather than jumping to VoIP technology, most enterprises take a wait-and-see approach. 3. Regulatory Environment Voice is now being delivered over unregulated IP networks. VoIP is treated as Internet-based information service as opposed to a communications service, thus VoIP providers dont have to pay state and local taxes. Regulatory and legislative bodies have engaged in the regulatory debate in the United States at both federal and state level. Regulatory and public policy decisions made now will determine how successful IP-based services will shape future voice and data services 4. Process Change IP-telephony will initiate process changes to those businesses that will allow this technology to be properly leveraged. These changes will affect the roles, rules, procedures, and structures pertaining to communications within enterprises as well as between the enterprise and customers. 5. Training To successfully implement advanced voice applications on data network, organizations have to either consult VoIP product vendors or educate their own data network and telephony teams involved in a convergence project. In addition, training is required for end users to ensure successful deployment and user adoption. 6. Consumer Adoption IP phone users expect IP telephony to deliver services of the same quality of PSTN, but at lower cost and more features. However, the cost of end user equipment for IP telephony is still high for individual consumers, and some IP phones still have only limited features

CODEC NEGOTIATION
Voice over IP (VoIP) Codec Negotiation
Codec negotiation allows a Net Performer unit to automatically change the codec that is loaded for a particular channel according to the codec list it receives in a SIP INVITE

FIG 2: CODEC NEGOTIATION message. Codec negotiation is availableon all SIP-enabled NetPerformer products that run V10.2.X.With SIP, when a unit sends an INVITE message to the remote side, theremote channel must be using the same codec for the call to beaccepted. Codec negotiation ensures that this is always the case, withoutrequiring manual reconfiguration of the channel Only the remote side can negotiate the codec. With codec negotiation calls can be set up more easily, as they arenot restricted to channels configured with the same codec as theoriginating channel The negotiation process is carried out rapidly, and does not interferewith the efficiency of the link.

VoIP Protocols
VoIP uses different protocols to setup a call, route packets, and create a conversation over the internet. For call setup, the two most used protocols are H323 and SIP. H323 is based on an ITU standard and has commonality with traditional PSTN. H.323 is the more mature of the two and covers a wider range of services. SIP, which was developed by the Internet Engineering Task Force (IETF), has no commonality with the PSTN format. Though SIP is less defined, it is more flexible, more scalable, and more easily integrated into the internet application; for example, it has better traversal network address translations (NAT) and firewalls. SIP is an application layer control protocol that can establish, modify and terminate media sessions such as internet telephony calls and multimedia connections. It works independently of underlying transport protocols and without dependency on the type of session that is being established. As a result of these advantages, the usage of SIP has surpassed that of H323. However in terms of traditional networks, H323 is more efficient, as its message encoding format is binary while SIP is text based and therefore bulky for networks. Proprietary protocols such as Skype are now widely used. Skype is an encrypted peer to peer protocol. Because of the way it functions, connecting using a randomly generated port and is always on, it is very inefficient in the network and uses a lot of bandwidth. For routing packets across the network, codecs are used to encode and decode both ends of the conversation, so that it can be sent and received across the network. Different codecs have characteristics and bandwidth requirements that can impact network performance. Codecs that employ no compression technology require more bandwidth. Those that use compression, require less bandwidth, but can impact voice quality. For the transfer of voice conversations VoIP usesmainly Real-time Transfer Protocol (RTP). This is with the exception of Skype, which uses proprietary protocols. RTP was originally designed for unidirectional real-time applications, but voice is a bidirectional real-time application. To facilitate the bidirectional nature of the VoIP application, RTP uses transmission header fields, containing data instructions for routing the voice packet. This adds to the packet size, and hence to the bandwidth requirements.

VoIP calls can be made on the Internet using a VoIP service provider and standard computer audio systems. Alternatively, some service providers support VoIP through ordinary telephones that use special adapters to connect to a home computer network. Many VoIP implementations are based on the H.323 technology standard. VoIP offers a substantial cost savings over traditional long distance telephone calls it will alter the pricing and cost structures of traditional telephony. Furthermore, when compared with circuit-switched services (yet another name for legacy networks), IP networks can carry 5 to 10 times the number of voice calls over the same bandwidth. H.323 An ITU Recommendation that defines Packet-based multimedia communications systems. H.323 defines a distributed architecture for creating multimedia applications, including VoIP SIP Defined as IETF RFC 2543. SIP defines a distributed architecture for creating multimedia applications, including VoIP MGCP Defined as IETF RFC 2705. MGCP defines a centralized architecture for creating multimedia applications, including VoIP H.248 An ITU Recommendation that defines Gateway Control Protocol. H.248 is the result of a joint-collaborate with the IETF. H.248 defines a centralized architecture, and is also known as Megaco Megaco Defined as IETF RFC 2885. Megaco defines a centralized architecture H.323 is a recommendation from the ITU Telecommunication Standardization Sector (ITU-T) that defines the protocols to provide audio-visual communication sessions on any packet network. The H.323 standard addresses call signaling and control, multimedia transport and control, and bandwidth control for point-to-point and multi-point conferences. It is widely implemented by voice and videoconferencing equipment manufacturers, is used within various Internet real-time applications such as GnuGK and NetMeeting and is widely deployed worldwide by service providers and enterprises for both voice and

video services over IP networks. It is a part of the ITU-T H.32x series of protocols, which also address multimedia communications over ISDN, the PSTN or SS7, and 3G mobile networks. H.323 call signaling is based on the ITU-T Recommendation Q.931 protocol and is suited for transmitting calls across networks using a mixture of IP, PSTN, ISDN, and QSIG over ISDN. A call model, similar to the ISDN call model, eases the introduction of IP telephony into existing networks of ISDN-based PBX systems, including transitions to IP-based PBXs. Within the context of H.323, an IP-based PBX might be a gatekeeper or other call control element which provides service to telephones or videophones. Such a device may provide or facilitate both basic services and supplementary services, such as call transfer, park, pick-up, and hold. While H.323 excels[citation needed] at providing basic telephony functionality and interoperability, H.323s strength lies in multimedia communication functionality designed specifically for IP networks The first version of H.323 was published by the ITU in November 1996[3] with an emphasis of enabling videoconferencing capabilities over a local area network (LAN), but was quickly adopted by the industry as a means of transmitting voice communication over a variety of IP networks, including WANs and the Internet (see VoIP). Over the years, H.323 has been revised and re-published with enhancements necessary to better-enable both voice and video functionality over packet-switched networks, with each version being backward-compatible with the previous version.[4] Recognizing that H.323 was being used for communication, not only on LANs, but over WANs and within large carrier networks, the title of H.323 was changed when published in 1998.[5] The title, which has since remained unchanged, is "Packet-Based Multimedia Communications Systems." The current version of H.323 was approved in 2009.[6] One strength of H.323 was the relatively early availability of a set of standards, not only defining the basic call model, but also the supplementary services needed to address business communication expectations.[citation needed] H.323 was the first VoIP standard to adopt the Internet Engineering Task Force (IETF) standard Real-time Transport Protocol (RTP) to transport audio and video over IP networks. H.323 is a system specification that describes the use of several ITU-T and IETF protocols. The protocols that comprise the core of almost any H.323 system are: * H.225.0 Registration, Admission and Status (RAS), which is used between an H.323 endpoint and a Gatekeeper to provide address resolution and admission control services. * H.225.0 Call Signaling, which is used between any two H.323 entities in order to establish communication.

* H.245 control protocol for multimedia communication, which describes the messages and procedures used for capability exchange, opening and closing logical channels for audio, video and data, control and indications. * Real-time Transport Protocol (RTP), which is used for sending or receiving multimedia information (voice, video, or text) between any two entities. Many H.323 systems also implement other protocols that are defined in various ITU-T Recommendations to provide supplementary services support or deliver other functionality to the user. Some of those Recommendations are:[citation needed]

FIG 3:H323 PROTOCOL STACK * H.235 series describes security within H.323, including security for both signaling and media. * H.239 describes dual stream use in videoconferencing, usually one for live video, the other for still images. * H.450 series describes various supplementary services. * H.460 series defines optional extensions that might be implemented by an endpoint or a Gatekeeper, including ITU-T Recommendations H.460.17, H.460.18, and H.460.19 for Network address translation (NAT) / Firewall (FW) traversal. In addition to those ITU-T Recommendations, H.323 implements various IETF Request for Comments (RFCs) for media transport and media packetization, including the Realtime Transport Protocol (RTP).Codecs ,H.323 utilizes both ITU-defined codecs and codecs defined outside the ITU. Codecs that are widely implemented by H.323 equipment include: * Audio codecs: G.711, G.729 (including G.729a), G.723.1, G.726, G.722, G.728, Speex * Text codecs: T.140 * Video codecs: H.261, H.263, H.264

All H.323 terminals providing video communications shall be capable of encoding and decoding video according to H.261 QCIF. All H.323 terminals shall have an audio codec and shall be capable of encoding and decoding speech according to ITU-T Rec. G.711. All terminals shall be capable of transmitting and receiving A-law and -law. Support for other audio and video codecs is optional.[6]

Gateways
Gateways are devices that enable communication between H.323 networks and other networks, such as PSTN or ISDN networks. If one party in a conversation is utilizing a terminal that is not an H.323 terminal, then the call must pass through a gateway in order to enable both parties to communicate. Gateways are widely used today in order to enable the legacy PSTN phones to interconnect with the large, international H.323 networks that are presently deployed by services providers. Gateways are also used within the enterprise in order to enable enterprise IP phones to communicate through the service provider to users on the PSTN. Gateways are also used in order to enable videoconferencing devices based on H.320 and H.324 to communicate with H.323 systems. Most of the third generation (3G) mobile networks deployed today utilize the H.324 protocol and are able to communicate with H.323-based terminals in corporate networks through such gateway devices. [edit] Gatekeepers A Gatekeeper is an optional component in the H.323 network that provides a number of services to terminals, gateways, and MCU devices. Those services include endpoint registration, address resolution, admission control, user authentication, and so forth. Of the various functions performed by the gatekeeper, address resolution is the most important as it enables two endpoints to contact each other without either endpoint having to know the IP address of the other endpoint. Gatekeepers may be designed to operate in one of two signaling modes, namely "direct routed" and "gatekeeper routed" mode. Direct routed mode is the most efficient and most widely deployed mode. In this mode, endpoints utilize the RAS protocol in order to learn the IP address of the remote endpoint and a call is established directly with the remote device. In the gatekeeper routed mode, call signaling always passes through the gatekeeper. While the latter requires the gatekeeper to have more processing power, it also gives the gatekeeper complete control over the call and the ability to provide supplementary services on behalf of the endpoints. H.323 endpoints use the RAS protocol to communicate with a gatekeeper. Likewise, gatekeepers use RAS to communicate with other gatekeepers. A collection of endpoints that are registered to a single Gatekeeper in H.323 is referred to as a zone. This collection of devices does not necessarily have to have an associated

physical topology. Rather, a zone may be entirely logical and is arbitrarily defined by the network administrator.

FIG 4: GATEWAY Gatekeepers have the ability to neighbor together so that call resolution can happen between zones. Neighboring facilitates the use of dial plans such as the Global Dialing Scheme. Dial plans facilitate inter-zone dialing so that two endpoints in separate zones can still communicate with each other.

Real-time Transport
In real-time interactive audio/video, people communicate with one another in real time. The Internet phone or voice over IP is an example of this type of application. Video conferencing is another example that allows people to communicate visually and orally. Characteristics Before addressing the protocols used in this class of applications, we discuss some characteristics of real-time audio/video communication. SIP Protocol The Secure Real-time Transport Protocol (or SRTP) defines a profile of RTP (Real-time Transport Protocol), intended to provide encryption, message authentication and integrity, and replay protection to the RTP data in both unicast and multicast applications.

It was developed by a small team of IP protocol and cryptographic experts from Cisco and Ericsson including David Oran, David McGrew, Mark Baugher, Mats Naslund, Elisabetta Carrara, James Black, Karl Norman, and Rolf Blom. It was first published by the IETF in March 2004 as RFC 3711. Since RTP is closely related to RTCP (Real Time Control Protocol) which can be used to control the RTP session, SRTP also has a sister protocol, called Secure RTCP (or SRTCP); SRTCP provides the same security-related features to RTCP, as the ones provided by SRTP to RTP. Utilization of SRTP or SRTCP is optional to the utilization of RTP or RTCP; but even if SRTP/SRTCP are used, all provided features (such as encryption and authentication) are optional and can be separately enabled or disabled. The only exception is the message authentication feature which is indispensably required when using SRTCP. 1 Data flow encryption 2 Authentication, integrity and replay protection 3 Key Derivation 4 SRTP Interoperability 5 External links

Data flow encryption

For encryption and decryption of the data flow (and hence for providing confidentiality of the data flow), SRTP (together with SRTCP) utilizes AES as the default cipher. There are two cipher modes defined which allow the original block cipher AES to be used as a stream cipher:

Segmented Integer Counter Mode

A typical counter mode, which allows random access to any blocks, which is essential for RTP traffic running over unreliable network with possible loss of packets. In the general case, almost any function can be used in the role of "counter", assuming that this function does not repeat for a long number of iterations. But the standard for encryption of RTP data is just a usual integer incremental counter. AES running in this mode is the default encryption algorithm, with a default encryption key length of 128 bits and a default session salt key length of 112 bits.

f8-mode
A variation of output feedback mode, enhanced to be seekable and with an altered initialization function. The default values of the encryption key and salt key are the same

as for AES in Counter Mode. (AES running in this mode has been chosen to be used in UMTS 3G mobile networks.) Besides the AES cipher, SRTP allows the ability to disable encryption outright, using the so called "NULL cipher", which can be assumed as the second supported cipher (or the third supported cipher mode in sum). In fact, the NULL cipher does not perform any encryption (i.e. the encryption algorithm functions as though the key stream contains only zeroes, and copies the input stream to the output stream without any changes). It is mandatory for this cipher mode to be implemented in any SRTP-compatible system. As such, it can be used when the confidentiality guarantees ensured by SRTP are not required, while other SRTP features (such authentication and message integrity) may be used. Though technically SRTP can easily accommodate new encryption algorithms, the SRTP standard states that new encryption algorithms besides those described cannot simply be added in some implementation of SRTP protocol. The only legal way to add a new encryption algorithm, while still claiming the compatibility with SRTP standard, is to publish a new companion standard track RFC which must clearly define the new algorithm.

Authentication, integrity and replay protection

The above-listed encryption algorithms do not secure message integrity themselves, allowing the attacker to either forge the data or at least to replay previously transmitted data. Hence the SRTP standard also provides the means to secure the integrity of data and safety from replay. To authenticate the message and protect its integrity, the HMAC-SHA1 algorithm (defined in RFC 2104) is used, which produces a 160-bit result, which is then truncated to 80 or 32 bits to become the authentication tag appended to the packet. The HMAC is calculated over the packet payload and material from the packet header, including the packet sequence number. To protect against replay attacks, the receiver maintains the indices of previously received messages, compares them with the index of each new received message and admits the new message only if it has not been played (i.e. sent) before. Such an approach heavily relies on the integrity protection being enabled (to make it impossible to spoof message indices).

Key Derivation
A key derivation function is used to derive the different keys used in a crypto context (SRTP and SRTCP encryption keys and salts, SRTP and SRTCP authentication keys) from one single master key in a cryptographically secure way. Thus, the key management protocol needs to exchange only one master key, all the necessary session keys are generated by applying the key derivation function. Periodical application of the key derivation function will result in security benefits. It prevents an attacker from collecting large amounts of ciphertext encrypted with one single session key. Certain attacks are easier to carry out when a large amount of ciphertext is available. Furthermore, multiple applications of the key derivation function provides backwards and forward security in the sense that a compromised session key

does not compromise other session keys derived from the same master key. This means even if an attacker managed to recover a certain session key, he is not able to decrypt messages secured with previous and later session keys derived from the same master key. (Note that, of course, a leaked master key reveals all the session keys derived from it.) SRTP relies on an external key management protocol to set up the initial master key. Two protocols specifically designed to be used with SRTP are ZRTP and MIKEY. There are also other methods to negotiate the SRTP keys. There are several vendors which offer products that use the SDES key exchange method.

SRTP Interoperability

Comparison of VoIP software for phones, servers and applications supporting SRTP RFC 3711, Proposed Standard, The Secure Real-time Transport Protocol (SRTP) RFC 4771, Proposed Standard, Integrity Transform Carrying Roll-Over Counter for the Secure Real-time Transport Protocol (SRTP) RFC 3551, Standard 65, RTP Profile for Audio and Video Conferences with Minimal Control RFC 3550, Standard 64, RTP: A Transport Protocol for Real-Time Applications RFC 2104, Informational, HMAC: Keyed-Hashing for Message Authentication Entry for SRTP in the voip-info.org-Wiki

SIP PROTOCOL
The Session Initiation Protocol (SIP) is an IETF-defined signaling protocol widely used[citation needed] for controlling communication sessions such as voice and video calls over Internet Protocol (IP). The protocol can be used for creating, modifying and terminating two-party (unicast) or multiparty (multicast) sessions. Sessions may consist of one or several media streams. Other SIP applications include video conferencing, streaming multimedia distribution, instant messaging, presence information, file transfer and online games[citation needed]. The SIP protocol is an Application Layer protocol designed to be independent of the underlying Transport Layer; it can run on Transmission Control Protocol (TCP), User Datagram Protocol (UDP), or Stream Control Transmission Protocol (SCTP).[1] It is a text-based protocol, incorporating many elements of the Hypertext Transfer Protocol (HTTP) and the Simple Mail Transfer Protocol (SMTP).

SIP is a text-based protocol with syntax similar to that of HTTP. There are two different types of SIP messages: requests and responses. The first line of a request has a method, defining the nature of the request, and a Request-URI, indicating where the request should be sent.The first line of a response has a response code. For SIP requests,RFC 3261 defines the following methods Main article: List of SIP request methods:REGISTER: Used by a UA to indicate its current IP address and the URLs for which it would like to receive calls. INVITE: Used to establish a media session between user agents. ACK: Confirms reliable message exchanges. CANCEL: Terminates a pending request. BYE: Terminates a session between two users in a conference. OPTIONS: Requests information about the capabilities of a caller, without setting up a call. SIP requests are the codes used by Session Initiation Protocol for communication. To complement them there are SIP Responses, which generally indicate whether this request succeeded or failed, and in the latter case, why it failed.

Request name Description

INVITE: Indicates a client is being invited to participate in a call session. ACK: Confirms that the client has received a final response to an INVITE request. RFC 3261 BYE Terminates a call and can be sent by either the caller or the callee. CANCEL OPTIONS REGISTER PRACK Cancels any pending request. Queries the capabilities of servers. Registers the address listed in the To header field with a SIP server. Provisional acknowledgement.

SUBSCRIBE Subscribes for an Event of Notification from the Notifier. NOTIFY PUBLISH Notify the subscriber of a new Event. Publishes an event to the Server.

INFO Sends mid-session information that does not modify the session state. REFER MESSAGE UPDATE Asks recipient to issue SIP request (call transfer.) Transports instant messages using SIP. Modifies the state of a session without changing the state of the dialog.

SIP RESPONSE
SIP responses are the codes used by Session Initiation Protocol for communication. They complement the SIP Requests, which are used to initiate action such as a phone conversation. Note that the Reason Phrases of the responses listed below are only the recommended examples, and can be replaced with local equivalents without affecting the protocol. LIST OF SIP RESPONSE CODES Provisional (1xx): Request received and being processed. Success (2xx): The action was successfully received, understood, and accepted. Redirection (3xx): Further action needs to be taken (typically by sender) to complete the request. Client Error (4xx): The request contains bad syntax or cannot be fulfilled at the server. Server Error (5xx): The server failed to fulfill an apparently valid request. Global Failure (6xx): The request cannot be fulfilled at any server. SIP makes use of transactions to control the exchanges between participants and deliver messages reliably. The transactions maintain an internal state and make use of timers. Client Transactions send requests and Server Transactions respond to those requests with one-or-more responses. The responses may include zero-or-more Provisional (1xx) responses and one-or-more final (2xx-6xx) responses. Transactions are further categorized as either Invite or Non-Invite. Invite transactions differ in that they can establish a long-running conversation, referred to as a Dialog in

SIP, and so include an acknowledgment (ACK) of any non-failing final response (e.g. 200 OK). Because of these transactional mechanisms, SIP can make use of un-reliable transports such as User Datagram Protocol (UDP).

FIG 5: SESSION INITIATION PROTOCOL If we take the above example, User1s UAC uses an Invite Client Transaction to send the initial INVITE (1) message. If no response is received after a timer controlled wait period the UAC may have chosen to terminate the transaction or retransmit the INVITE.

However, once a response was received, User1 was confident the INVITE was delivered reliably. User1s UAC then must acknowledge the response. On delivery of the ACK (2) both sides of the transaction are complete. And in this case, a Dialog may have been established.

VOIP Advantages
First, we have an expectation that VOIP has many advantages over regular phone service. Low cost. If you have a broadband Internet connection (DSL or cable), you can make PC-to-PC phone calls anywhere in the world for free. If you wish to make a PC-to-phone connection, there's usually a charge for this but probably much cheaper than your regular phone service. You can pay as you go or you can sign up with a VOIP service provider and pay a monthly fee in return for unlimited calls within a certain geographic area. For example, some VOIP services in the United States allow you to call anywhere in North America at no extra charge. Overseas calls are charged at a relatively small rate. 1.Portability. You can take your cheap phone service with you and make and receive phone calls wherever there is a broadband connection simply by signing in to your VOIP account. This makes VOIP as convenient as e-mail. If you are traveling, simply pack a headset and use your laptop plugged into the Internet, or plug a VOIP phone directly into the Internet connection and you can talk to your family or business associates for almost nothing. If you don't have a VOIP phone, use an analog terminal adaptor (ATA) to connect a regular phone to the Internet. Today you can find ATAs that small, portable and inexpensive. A great idea for frequent travelers. 2.Phone-to-phone VOIP is also portable. When you sign up with a VoIP service provider the Internet phone or adaptor that is used with that service is assigned a unique number. This 'phone number' remains valid even if your VoIP service is in Cleveland and you are connected to the Internet in Bangkok. An Internet phone is small and light enough to take with you anywhere. Simply plug it into a broadband connection anywhere in the world and you can make and receive calls just as though you were in your own home or office. VOIP Disadvantages If VOIP is starting to sound really good to you, make sure you understand the following downsides as well. No service during a power outage. During a blackout a regular phone is kept in service by the current supplied through the phone line. This is not possible with IP phones, so when the power goes out, there is no VOIP phone service. One solution to this problem is to use battery backups or power generators to provide electricity.

If you decide to continue subscribing to a regular phone line as an emergency backup, consider that monthly cost cuts into your overall VOIP savings. However, VOIP would still make sense in this case if your home or business made significant long distance calls. Without power VOIP phones are useless, so in case of emergencies during power cuts it can be a major disadvantage. Emergency 911 calls. Another major concern with VOIP involves emergency 911 calls. Traditional phone equipment can trace your location. Emergency calls are diverted to the nearest call center where the operator can see your location in case you can't talk. However, because a voice-over-IP call is essentially a transfer of data between two IP addresses, not physical addresses, with VOIP there is currently no way to determine where your VOIP phone call is originating from. To solve this issue, the E911 standard by law mandates that VOIP service providers pass name and address information to the nearest Public Safety Access Point (PSAP) when 911 is dialed. Not all PSAPs in the country support E911 yet, so you should ask when deciding on a VOIP service provider if E911 is supported in your area.

Features.
Unlike regular phone service which usually charges more for extra features, VOIP comes with a host of advanced communication features. For example, call forwarding, call waiting, voicemail, caller ID and three-way calling are some of the many services included with VOIP telephone service at no extra charge. You can also send data such as pictures and documents at the same time you are talking on the phone. CONCLUSION This paper made a review on VoIP over WLAN, itsadvantages and challenges. As a major concerned issue, QoSwas explained and its contributing factors were stated indetails. Although, VoIP can tolerate packet loss to someextent, it is very sensitive to delay factor. Jitter also plays amain role on voice quality therefore jitter buffer is introducedto smooth the playout of packets. Echo and throughput arealso other factors that affect the quality of voice. Differenttechniques for voice QoS assessment werementioned as well.Furthermore, WLAN is a bandwidth limited network whichmaylimit the number of VoIP calls. Accordingly, this paperdiscussed capacity and its measurement technique as it is base on IEEE 802.11b standard with voice codes and Quality and Reliability. Because VOIP relies on an Internet connection, your VOIP service will be affected by the quality and reliability of your broadband Internet service and sometimes by the limitations of your PC. Poor Internet connections and congestion can result in garbled or distorted voice quality. If you are using your computer at the same time as making a computer VOIP call, you may find that voice quality deteriorates dramatically.

Bad or unreliable internet connections can result in poor voice quality, such as clipping, voice delay, or dropped calls. And emergency 911 service may not work as you expect.

References [1] Overview of the PSTN and Comparisons to Voice over IP,
CH01, White paper, October 2001; [2] H. M. Chong, H. S. Matthews, Comparative Analysis of Traditional Telephone and Voice-over-Internet Protocol (VoIP) Systems, IEEE ISEE 2004; [3] H.323 Technology, Ixia, White Paper, 2004; [4] IETF RFC 3261, SIP: Session Initiation Protocol, June 2002; [5] IETF RFC 3435, Media Gateway Control Protocol (MGCP) Version 1.0, January 2003; [6] Media Gateway Control Protocol (MGCP) Technology, [7]. Forouzan Data Communications and Networking", Mcgraw Hill International Edition, Fourth Edition, International Edition 2007.