E-Business : Voice over IP

Faculty of Computer Science and Automatic Control
Politehnica University of Bucharest
313 Splaiul Independenţei
ROMANIA

Abstract: - Electronic business, commonly referred to as "eBusiness" or "e-business", or an internet business, may be defined as the application of information and communication technologies (ICT) in support of all the activities of business. Commerce constitutes the exchange of products and services between businesses, groups and individuals and can be seen as one of the essential activities of any business. Electronic commerce focuses on the use of ICT to enable the external activities and relationships of the business with individuals, groups and other businesses. Electronic business methods enable companies to link their internal and external data processing systems more efficiently and flexibly, to work more closely with suppliers and partners, and to better satisfy the needs and expectations of their customers.

Key-Words: - VoIP, gatekeeper, endpoint, gateway, softphone, asterisk

1 Introduction
E-business involves business processes spanning the entire value chain: electronic purchasing and supply chain management, processing orders electronically, handling customer service, and cooperating with business partners. Special technical standards for e-business facilitate the exchange of data between companies. E-business software solutions allow the integration of intra and inter firm business processes. E-business can be conducted using the Web, the Internet, intranets, extranets, or some combination of these. Applications for electronic business can be divided into three categories: a) Internal business systems: o customer relationship management o enterprise resource planning o document management systems o human resources management b) Enterprise communication and collaboration: o VoIP o content management system o e-mail o voice mail o Web conferencing o Digital work flows (or business process management) c) Electronic commerce - business-to-business electronic commerce (B2B) or business-to-consumer electronic commerce (B2C): o internet shop o supply chain management o online marketing o offline marketing

1. Introduction to Voice over IP
Voice over IP (VoIP) is a family of technologies, methodologies, communication protocols, and transmission techniques for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. Other terms frequently encountered and often used synonymously with VoIP are IP telephony, Internet telephony, voice over broadband (VoBB), broadband telephony, and broadband phone. Internet telephony refers to communications services—Voice, fax, SMS, and/or voice-messaging applications—that are transported via the Internet, rather than the public switched telephone network (PSTN). The steps involved in originating a VoIP telephone call are signaling and media channel setup, digitization of the analog voice signal, encoding, packetization, and transmission as Internet Protocol (IP) packets over a packet-switched network. On the receiving side, similar steps (usually in the reverse order) such as reception of the IP packets, decoding of the packets and digital-to-analog conversion reproduce the original voice stream. Even though IP Telephony and VoIP are terms that are used interchangeably, they are actually different; IP telephony has to do with digital telephony systems that use IP protocols for voice communication while VoIP is actually a subset of IP Telephony. VoIP is a technology used by IP telephony as a means of transporting phone calls. VoIP systems employ session control protocols to control the set-up and tear-down of calls as well as audio codecs which encode speech allowing transmission over an IP network as digital audio via an audio stream. The choice of codec varies between different implementations of VoIP depending on application requirements and network bandwidth; some implementations rely on narrowband and compressed speech, while others support high fidelity stereo codecs. VoIP is available on many smartphones and Internet devices so that users of portable devices that are not phones may place calls or send SMS text messages over 3G or Wi-Fi.

2. Protocols
Voice over IP has been implemented in various ways using both proprietary and open protocols and standards. Examples of the network protocols used to implement VoIP include: • H.323 • Media Gateway Control Protocol (MGCP) • Session Initiation Protocol (SIP) • Real-time Transport Protocol (RTP) • Session Description Protocol (SDP) • Inter-Asterisk eXchange (IAX)
The H.323 protocol was one of the first VoIP protocols that found widespread implementation for long-distance traffic, as well as local area network services. However, since the development of newer, less complex protocols, such as MGCP and SIP, H.323 deployments are increasingly limited to carrying existing long-haul network traffic. In particular, the Session Initiation Protocol (SIP) has gained widespread VoIP market penetration. 1. SIP
The Session Initiation Protocol (SIP) is an IETF-defined signaling protocol widely used 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. 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). SIP employs design elements similar to the HTTP request/response transaction model. Each transaction consists of a client request that invokes a particular method or function on the server and at least one response. SIP reuses most of the header fields, encoding rules and status codes of HTTP, providing a readable text-based format. For SIP requests, RFC 3261 defines the following 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. • PRACK (Provisional Response Ack): PRACK improves network reliability by adding an acknowledgement system to the provisional Responses (1xx). PRACK is sent in response to provisional response (1xx).
The SIP response types defined in RFC 3261 fall in one of the following categories: • 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. 2. H323
The H.323 system defines several network elements that work together in order to deliver rich multimedia communication capabilities. Those elements are Terminals, Multipoint Control Units (MCUs), Gateways, Gatekeepers, and Border Elements. Collectively, terminals, multipoint control units and gateways are often referred to as endpoints. While not all elements are required, at least two terminals are required in order to enable communication between two people. In most H.323 deployments, a gatekeeper is employed in order to, among other things, facilitate address resolution.H.323 Network Elements:
Terminals in an H.323 network are the most fundamental elements in any H.323 system, as those are the devices that users would normally encounter. They might exist in the form of a simple IP phone or a powerful high-definition videoconferencing system. Inside an H.323 terminal is something referred to as a Protocol stack, which implements the functionality defined by the H.323 system.
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.
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. 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.
3. Advantages
The basic and most important advantage of Voice over IP is the cost. VoIP can be a benefit for reducing communication and infrastructure costs. Examples include: • Routing phone calls over existing data networks to avoid the need for separate voice and data networks. • The ability to transmit more than one telephone call over a single broadband connection. • Secure calls using standardized protocols (such as Secure Real-time Transport Protocol). Most of the difficulties of creating a secure telephone connection over traditional phone lines, such as digitizing and digital transmission, are already in place with VoIP. It is only necessary to encrypt and authenticate the existing data stream.
4. Challenges
With the VoIP technology there are many challenges that have to be dealt with.

1. Quality of service
Communication on the IP network is inherently less reliable in contrast to the circuit-switched public telephone network, as it does not provide a network-based mechanism to ensure that data packets are not lost, and are delivered in sequential order. It is a best-effort network without fundamental Quality of Service (QoS) guarantees. Therefore, VoIP implementations may face problems mitigating latency and jitter. By default, network routers handle traffic on a first-come, first-served basis. Network routers on high volume traffic links may introduce latency that exceeds permissible thresholds for VoIP. Fixed delays cannot be controlled, as they are caused by the physical distance the packets travel; however, latency can be minimized by marking voice packets as being delay-sensitive with methods such as DiffServ. A VoIP packet usually has to wait for the current packet to finish transmission, although it is possible to preempt (abort) a less important packet in mid-transmission, although this is not commonly done, especially on high-speed links where transmission times are very short even for maximum-sized packets. An alternative to preemption on slower links, such as dialup and DSL, is to reduce the maximum transmission time by reducing the maximum transmission unit. But every packet must contain protocol headers, so this increases relative header overhead on every link traversed, not just the bottleneck (usually Internet access) link. Voice, and all other data, travels in packets over IP networks with fixed maximum capacity. This system may be more prone to congestion and DoS attacks than traditional circuit switched systems; a circuit switched system of insufficient capacity will refuse new connections while carrying the remainder without impairment, while the quality of real-time data such as telephone conversations on packet-switched networks degrades dramatically. Fixed delays cannot be controlled as they are caused by the physical distance the packets travel. They are especially problematic when satellite circuits are involved because of the long distance to a geostationary satellite and back; delays of 400–600 ms are typical. When the load on a link grows so quickly that its switches experience queue overflows, congestion results and data packets are lost. This signals a transport protocol like TCP to reduce its transmission rate to alleviate the congestion. But VoIP usually uses UDP not TCP because recovering from congestion through retransmission usually entails too much latency.[13] So QoS mechanisms can avoid the undesirable loss of VoIP packets by immediately transmitting them ahead of any queued bulk traffic on the same link, even when that bulk traffic queue is overflowing. The receiver must resequence IP packets that arrive out of order and recover gracefully when packets arrive too late or not at all. Jitter results from the rapid and random (i.e., unpredictable) changes in queue lengths along a given Internet path due to competition from other users for the same transmission links. VoIP receivers counter jitter by storing incoming packets briefly in a "de-jitter" or "playout" buffer, deliberately increasing latency to improve the chance that each packet will be on hand when it is time for the voice engine to play it. The added delay is thus a compromise between excessive latency and excessive dropout, i.e., momentary audio interruptions. Although jitter is a random variable, it is the sum of several other random variables that are at least somewhat independent: the individual queuing delays of the routers along the Internet path in question. Thus according to the central limit theorem, we can model jitter as a gaussian random variable. This suggests continually estimating the mean delay and its standard deviation and setting the playout delay so that only packets delayed more than several standard deviations above the mean will arrive too late to be useful. In practice, however, the variance in latency of many Internet paths is dominated by a small number (often one) of relatively slow and congested "bottleneck" links. Most Internet backbone links are now so fast (e.g. 10 Gbit/s) that their delays are dominated by the transmission medium (e.g., optical fiber) and the routers driving them do not have enough buffering for queuing delays to be significant. It has been suggested to rely on the packetized nature of media in VoIP communications and transmit the stream of packets from the source phone to the destination phone simultaneously across different routes (multi-path routing). In such a way, temporary failures have less impact on the communication quality. In capillary routing it has been suggested to use at the packet level Fountain codes or particularly raptor codes for transmitting extra redundant packets making the communication more reliable. A number of protocols have been defined to support the reporting of QoS/QoE for VoIP calls. These include RTCP Extended Report (RFC 3611), SIP RTCP Summary Reports, H.460.9 Annex B (for H.323), H.248.30 and MGCP extensions. The RFC 3611 VoIP Metrics block is generated by an IP phone or gateway during a live call and contains information on packet loss rate, packet discard rate (because of jitter), packet loss/discard burst metrics (burst length/density, gap length/density), network delay, end system delay, signal / noise / echo level, Mean Opinion Scores (MOS) and R factors and configuration information related to the jitter buffer.

2. Susceptibility to power failure
Telephones for traditional residential analog service are usually connected directly to telephone company phone lines which provide direct current to power most basic analog handsets independently of locally available power. IP Phones and VoIP telephone adapters connect to routers or cable modems which typically depend on the availability of mains electricity or locally generated power. Some VoIP service providers use customer premise equipment (e.g., cable modems) with battery-backed power supplies to assure uninterrupted service for up to several hours in case of local power failures. Such battery-backed devices typically are designed for use with analog handsets. Some VoIP service providers implement services to route calls to other telephone services of the subscriber, such a cellular phone, in the event that the customer's network device is inaccessible to terminate the call.The susceptibility of phone service to power failures is a common problem even with traditional analog service in areas where many customers purchase modern telephone units that operate with wireless handsets to a base station, or that have other modern phone features, such as built-in voicemail or phone book features. 3. Lack of redundancy
The current separation of the Internet and the PSTN provides a certain amount of redundancy. An Internet outage does not necessarily mean that a voice communication outage will occur simultaneously, allowing individuals to call for emergency services and many businesses to continue to operate normally. In situations where telephone services become completely reliant on the Internet infrastructure, a single-point failure can isolate communities from all communication, including Enhanced 911 and equivalent services in other locales. However, the network design envisioned by DARPA in the early 1980s included a fault tolerant architecture under adverse conditions.

4. Number portability
Local number portability (LNP) and Mobile number portability (MNP) also impact VoIP business. In November 2007, the Federal Communications Commission in the United States released an order extending number portability obligations to interconnected VoIP providers and carriers that support VoIP providers. Number portability is a service that allows a subscriber to select a new telephone carrier without requiring a new number to be issued. Typically, it is the responsibility of the former carrier to "map" the old number to the undisclosed number assigned by the new carrier. This is achieved by maintaining a database of numbers. A dialed number is initially received by the original carrier and quickly rerouted to the new carrier. Multiple porting references must be maintained even if the subscriber returns to the original carrier. The FCC mandates carrier compliance with these consumer-protection stipulations. A voice call originating in the VoIP environment also faces challenges to reach its destination if the number is routed to a mobile phone number on a traditional mobile carrier. VoIP has been identified in the past as a Least Cost Routing (LCR) system, which is based on checking the destination of each telephone call as it is made, and then sending the call via the network that will cost the customer the least. This rating is subject to some debate given the complexity of call routing created by number portability. With GSM number portability now in place, LCR providers can no longer rely on using the network root prefix to determine how to route a call. Instead, they must now determine the actual network of every number before routing the call. Therefore, VoIP solutions also need to handle MNP when routing a voice call. In countries without a central database, like the UK, it might be necessary to query the GSM network about which home network a mobile phone number belongs to. As the popularity of VoIP increases in the enterprise markets because of least cost routing options, it needs to provide a certain level of reliability when handling calls. MNP checks are important to assure that this quality of service is met. By handling MNP lookups before routing a call and by assuring that the voice call will actually work, VoIP service providers are able to offer business subscribers the level of reliability they require.

5. Security
VoIP telephone systems are susceptible to attacks as are any Internet-connected devices. This means that hackers who know about these vulnerabilities (such as insecure passwords) can institute denial-of-service attacks, harvest customer data, record conversations and break into voice mailboxes. Another challenge is routing VoIP traffic through firewalls and network address translators. Private Session Border Controllers are used along with firewalls to enable VoIP calls to and from protected networks. For example, Skype uses a proprietary protocol to route calls through other Skype peers on the network, allowing it to traverse symmetric NATs and firewalls. Other methods to traverse NATs involve using protocols such as STUN or Interactive Connectivity Establishment (ICE). Many consumer VoIP solutions do not support encryption, although having a secure phone is much easier to implement with VoIP than traditional phone lines. As a result, it is relatively easy to eavesdrop on VoIP calls and even change their content. An attacker with a packet sniffer could intercept your VoIP calls if you are not on a secure VLAN. However, physical security of the switches within an enterprise and the facility security provided by ISPs make packet capture less of a problem than originally foreseen. Further research has shown that tapping into a fiber optic network without detection is difficult if not impossible. This means that once a voice packet is within the Internet backbone it is relatively safe from interception. There are open source solutions, such as Wireshark, that facilitate sniffing of VoIP conversations. A modicum of security is afforded by patented audio codecs in proprietary implementations that are not easily available for open source applications; however, such security through obscurity has not proven effective in other fields. Some vendors also use compression, which may make eavesdropping more difficult. However, real security requires encryption and cryptographic authentication which are not widely supported at a consumer level. The existing security standard Secure Real-time Transport Protocol (SRTP) and the new ZRTP protocol are available on Analog Telephone Adapters (ATAs) as well as various softphones. It is possible to use IPsec to secure P2P VoIP by using opportunistic encryption. Skype does not use SRTP, but uses encryption which is transparent to the Skype provider. In 2005, Skype invited a researcher, Dr Tom Berson, to assess the security of the Skype software, and his conclusions are available in a published report. To prevent the above security concerns government and military organizations are using voice over secure IP (VoSIP), secure voice over IP (SVoIP), and secure voice over secure IP (SVoSIP) to protect confidential and classified VoIP communications. Secure voice over IP is accomplished by encrypting VoIP with Type 1 encryption. Secure voice over Secure IP is accomplished by using Type 1 encryption on a classified network, like SIPRNet. Public Secure VoIP is also available with free GNU programs. Caller ID support among VoIP providers varies, but is provided by the majority of VoIP providers.
Many VoIP carriers allow callers to configure arbitrary Caller ID information, thus permitting spoofing attacks. Business grade VoIP equipment and software often makes it easy to modify caller ID information, providing many businesses great flexibility. The Truth in Caller ID Act has been in preparation in the US Congress since 2006, but as of January 2009 still has not been enacted. This bill proposes to make it a crime in the United States to "knowingly transmit misleading or inaccurate caller identification information with the intent to defraud, cause harm, or wrongfully obtain anything of value ...".

1. Voice over IP softwares
There are many softwares that are usefull for voice over IP. I will describe two types of software in the following pages: softphones (client side) and gatekeepers (server side).

2.1 Softphones
A softphone is a software program for making telephone calls over the Internet using a general purpose computer, rather than using dedicated hardware. Often a softphone is designed to behave like a traditional telephone, sometimes appearing as an image of a phone, with a display panel and buttons with which the user can interact. A softphone is usually used with a headset connected to the sound card of the PC, or with a USB phone. To make voice calls over the Internet, a user typically requires the following: • A modern PC with a microphone and speaker, or with a headset, or with USBphone. • Reliable high-speed Internet connectivity like Digital Subscriber Line (DSL), or cable service. • Account with an Internet telephony service provider or IP PBX provider 1. SJphone
SJphone is a softphone that allows you to speak over Internet with any desktops, notebooks, PDAs, stand-alone IP-phones and even with any conventional landline or mobile phones. It supports both SIP and H.323 industry standards, and is fully inter-operable with most major Internet Telephony Service Providers (ITSP) and VoIP software and hardware. For calling over the world you need to open an account with an IP-Telephony service and to get their service profile, or to make their SJphone™ profile yourself. Advanced users may build their own Voice over IP networks using either an H.323 Gatekeeper, SIP proxy, IP-PBX, and other components. They can also connect to conventional telephones using H.323/SIP Gateways (such as Cisco 26xx/36xx/37xx/53xx).
Sjphone features include the following:

SIP Compatibility allows you to speak over Internet

- directly with most other SIP-softphones and stand-alone SIP-phones (such as Cisco 7940) - with other people in your company through your SIP IP-PBX system or Proxies (such as Asterisk, Broadsoft and others)
- with all other people around the world using most SIP ITSP accounts such as FWD (www.fwd.pulver.com)
- with a regular telephone using most SIP Gateways

SIP features:
- TCP support
- TLS tunneling support
- Full support for SRV/NAPTR records and transport layer enhancements
- Concealed Caller ID
- DNS support in the SIP stack

H.323 Compatibility allows you to speak over Internet

- directly with any other H.323-softphones, and stand-alone H.323-phones.
- with other people in your company through your H.323 IP-PBX system, such as Cisco Call Manager
- with all other people around the world through any H.323 ITSP accounts
- with a computer or telephone on a private network through a Gatekeeper
- a regular telephone using most H.323 Gateways

Instant Messaging:

- Jabber: Presence
- Jabber: Simple Chat
- Jabber: vCard Display/Editor
- Jabber: Customizable Rules for Placing Calls from the Roster
- Jabber: full support for TLS transport

Features for Advanced Users:

- Customized service profiles allow you to create your own profiles for calls through your H.323 Gatekeeper, Gateway, or SIP Proxy
- Multisession calls
- 3-way conferencing using different VoIP protocols
- Manual codec selection
- Support for extended H.323 address syntax
- Support for advanced SIP URI syntax
- Support for CallTo URL
- Lost and out-of-order packet indication
- Log window for operation monitoring
- Media Engine statistics
- Per-Call RTP statistics
- QoS/TOS/DiffServ support
- Advanced lost packet recovery, offering better sound quality over a poor connection.
- Optimized processor usage for HT/Multicore processors
- Remote support console
- An SDK for developing third-party VoIP applications

2. X-lite
X–Lite 3.0 is CounterPath’s next-generation softphone client, offering users all the productivity of a traditional telephone with desktop and mobile computer enhancements. From a simple click of a mouse button or tap on the keyboard users can dial, answer, or otherwise manage calls and personal availability. Whether over wired or wireless connections, X-Lite supports a variety of headset devices to augment the modern telephony experience, severing the restrictive tethers of traditional, limited telephone receivers. Designed to work over IP-based systems, X-Lite provides endpoint VoIP solutions that use internet-based telephony servers within an enterprise LAN (Local Area Network) or VoIP service provider network.
Standard Telephone Features
The X-Lite 3.0 softphone has all the standard telephone features, including:
• Two lines
• Call display and Message Waiting Indicator (MWI)
• Speakerphone
• Mute
• Redial
• Hold
• Do not disturb
• Call ignore
• Call history – list of received, missed, dialed and blocked calls
• Call forward
• Call record
• Three-way audio and video conferencing.
Enhanced Features and Functions
The X-Lite 3.0 softphone also supports the following VoIP features and functions:
• Instant messaging and presence using the SIMPLE protocol.
• Managed Contacts list—importing and exporting contacts between X-Lite and other applications.
• Support for Intel® Centrino® Mobile technology, allowing X-Lite to provide more consistent quality of service across both wired and wireless networks using industry standards such as 802.11e.
• “Zero touch” configuration of audio and video devices; no manual setup is required.
• “Zero touch” detection of the bandwidth that a user’s computer can access for communication.
• Acoustic echo cancellation, automatic gain control, voice activity detection.
• Support for the following audio codecs:
Broadvoice-32, Broadvoice-32 FEC, G.711aLaw, G.711uLaw, GSM, iLBC, L16 PCM Wideband.
• Support for the following video codecs:
H.263, H.263 1998.
• Automatic selection of the best codec based on the remote party’s capability, available bandwidth, and network conditions. X-Lite switches codecs during a call in response to changing network conditions.
• Compliance with the RFC 3261 SIP standard.
• STUN and ICE NAT traversal. XTunnels for firewall traversal.
• Support for DTMF (RFC 2833, inband DTMF or SIP INFO messages).

1. Gatekeepers
A gatekeeper is an H.323 entity on the network that provides services such as address translation and network access control for H.323 terminals, gateways, and MCUs. Also, they can provide other services such as bandwidth management, accounting, and dial plans that you can centralize in order to provide scalability. Gatekeepers are logically separated from H.323 endpoints such as terminals and gateways. They are optional in an H.323 network. But if a gatekeeper is present, endpoints must use the services provided.
A typical H323 Gatekeeper call flow for a successful call may look like:

| | | |
Endpoint A Endpoint B 1234 1123 1. Endpoint A dials 1123 from the system. 2. Endpoint A sends ARQ (Admission Request) to the Gatekeeper. 3. Gatekeeper returns ACF (Admission Confirmation) with IP address of endpoint B. 4. Endpoint A sends Q.931 call setup messages to endpoint B. 5. Endpoint B sends the Gatekeeper an ARQ, asking if it can answer call. 6. Gatekeeper returns an ACF with IP address of endpoint A. 7. Endpoint B answers and sends Q.931 call setup messages to endpoint A. 8. IRR sent to Gatekeeper from both endpoints. 9. Either endpoint disconnects the call by sending a DRQ (Disconnect Request) to the Gatekeeper. 10. Gatekeeper sends a DCF (Disconnect Confirmation) to both endpoints.
The gatekeeper allows calls to be placed either: Directly between endpoints (Direct Endpoint Model), or Route the call signaling through itself (Gatekeeper Routed Model). 1. GNUGK Gatekeeper
The GNU Gatekeeper is an open-source project that implements a H.323 gatekeeper. A gatekeeper provides call control services to H.323 endpoints and is an integral part of most telephony or video conferencing installations that are based on the H.323 standard.
According to the H.323 standard, a gatekeeper shall provide the following services:

Address Translation Admissions Control Bandwidth Control Zone Management Call Control Signaling Call Authorization Bandwidth Management Call Management

The GNU Gatekeeper implements all of these functions and a number of additional features. For the protocol handling and encoding and decoding of the messages, it uses the H323Plus library. H.323 is an international standard published by the ITU. It is a communications standard for audio, video, and data over the Internet.
GnuGk contains a rich set of features, including: • Cross-platform, including Linux, Windows, Mac OS X, Solaris, FreeBSD, OpenBSD and NetBSD. • A policy-based flexible routing mechanism. • Calling and called numbers rewriting, including CLI rewriting. • Full H.323 proxy, including RTP/RTCP media channels and T.120 data channels. • NAT traversal using a number of protocols, including H.460.18 and H.460.19. • LDAP directory support (H.350) • Call retry / failover. • Clustering support by neighbors, parent/child, alternates GK. • TCP status port for monitoring and external call routing. • H.235 security. • Accounting and call authorization via SQL database, radius • ENUM support

2. Asterisk
Asterisk is software that turns an ordinary computer into a communications server. Asterisk powers IP PBX systems, VoIP gateways, conference servers and more. It is used by small businesses, large businesses, call centers, carriers and governments worldwide. Asterisk is free and open source.
Asterisk-based telephony solutions offer a rich and flexible feature set. Asterisk offers both classical PBX functionality and advanced features, and interoperates with traditional standards-based telephony systems and Voice over IP systems. Asterisk offers the advanced features that are often associated with large, high end (and high cost) proprietary PBXs.

Call Features:

ADSI On-Screen Menu System
Alarm Receiver
Append Message
Authentication
Automated Attendant
Blacklists
Blind Transfer
Call Detail Records
Call Forward on Busy
Call Forward on No Answer
Call Forward Variable
Call Monitoring
Call Parking
Call Queuing
Call Recording
Call Retrieval
Call Routing (DID & ANI)
Call Snooping
Call Transfer
Call Waiting
Caller ID
Caller ID Blocking
Caller ID on Call Waiting
Calling Cards
Conference Bridging
Database Store / Retrieve
Database Integration
Dial by Name
Direct Inward System Access
Distinctive Ring
Distributed Universal Number
2. Personal improvements
The problem with the existing most common used free gatekeepers (gnugk and asterisk) is that they both lack a friendly GUI. GNUGK has a GUI built in JAVA which is very slow and inefficient. Until recent years GNUGK lacked an user interface completely. On the other hand asterisk has a fully featured interface built in PHP, that is very responsive, hard to configure and does not have the possibility to visualize the calls that are in progress.
This defect or lack in functionality leads to poor monitorization for the person/persons that is/are supervising the network. For this reason I’ve decided to personally build an user interface that is fast, efficient, highly customizable that enables you to detect the slightest problem in less than a minute.

4 Conclusion
Voice over IP is quickly becoming readily available across much of the world, however many problems still remain. For the time being transmission networks involve too much latency or drop too many packets, this effects quality of service sometimes severely deteriorating the quality of the call. Also VOIP contains many security risks, sending out packets that any person may intercept. Although VOIP may offer cheaper solutions for many the PSTN offers a high QoS and greater security that makes up for its higher prices. It is my belief that the telephone market will continue to be dominated by the PSTN until quality of service and security issues can be addressed.

References:
[1] Tomi Yletyinen - “The Quality of voice over IP” http://www.netlab.tkk.fi/tutkimus/ipana/paperit/tomidt.pdf [2] Voip Benefits article http://www.voipresource.net/VoIP-benefits.htm [3] Anshul Kundaje şi Geetali Bhatia - „Voice over IP”http://www1.cs.columbia.edu/~abk2001/voip.pdf

[4] H.323 article

http://en.wikipedia.org/wiki/H.323

[5] Session Initiation Protocol article

http://en.wikipedia.org/wiki/Session_Initiation_Protocol

[6] „Voice quality in converging telephony and IP networks” article

http://www.highbeam.com/doc/1G1-66157325.html

[7] Documentaţie tehnologii web

http://www.w3schools.com/

[8] „Modern ways of communication - VoIP” article

http://www.articlesbase.com/voip-articles/modern-ways-of-communication-voip-53631.html