VOIP-Voice Over Internet Protocol
VOIP:VOICE OVER INTERNET PROTOCOL
Problem Definition and Description:
Introduction: Voice over Internet Protocol (VoIP) is a general term for a family of transmission technologies for delivery of voice communications over IP networks such as the Internet or other packet-switched networks. Other terms frequently encountered and synonymous with VOIP are IP telephony, Internet telephony, voice over broadband (VOBB), broadband telephony and broadband phone. Internet telephony refers to communications services — voice, facsimile, and/or voice-messaging applications — that are transported via the Internet, rather than the public switched telephone network (PSTN). The basic steps involved in originating an Internet telephone call are conversion of the analog voice signal to digital format and compression/translation of the signal into Internet protocol (IP) packets for transmission over the Internet; the process is reversed at the receiving end.
VOIP systems employ session control protocols to control the set-up and tear-down of calls as well as audio codec’s which encode speech allowing transmission over an IP network as digital audio via an audio stream. Codec use is varied between different implementations of VOIP (and often a range of codec’s are used); some implementations rely on narrowband and compressed speech, while others support high fidelity stereo codec.
There are three common methods of connecting to VOIP service providers:
A typical analog telephone adapter (ATA) for connecting an analog phone to a VOIP provider
* An Analog Telephone Adapter (ATA) may be connected between an IP network (such as a broadband connection) and an existing telephone jack in order to provide service nearly indistinguishable from PSTN providers on all the other telephone jacks in the residence. This type of service, which is fixed to one location, is generally offered by broadband Internet providers such as cable companies and telephone companies as a cheaper flat-rate traditional phone service.
* Dedicated VOIP phones are phones that allow VOIP calls without the use of a computer. Instead they connect directly to the IP network (using technologies such as Wi-Fi or Ethernet). In order to connect to the PSTN they usually require service from a VOIP service provider; most people therefore will use them in conjunction with a paid service plan.
* A soft phone (also known as an Internet phone or Digital phone) is a piece of software that can be installed on a computer that allows VOIP calling without dedicated hardware. It is becoming increasingly common for telecommunications providers to use VOIP telephony over dedicated and public IP networks to connect switching stations and to interconnect with other telephony network providers; this is often referred to as "IP backhaul"."Dual mode" telephone sets, which allow for the seamless handover between a cellular network and a Wi-Fi network, are expected to help VOIP become more popular.
Phones such as the NEC N900iL, many of the Nokia E-series and several other Wi-Fi enabled mobile phones have SIP clients built into the firmware. Such clients operate independently of the mobile phone network (however some operators choose to remove the client from subsidized handsets). Some operators such as Vodafone actively try to block VOIP traffic from their network. Others, like T-Mobile, have refused to interconnect with VOIP-enabled networks as was seen in the legal case between T-Mobile and True phone, which ultimately was settled in the UK High Court in favor of the VOIP carrier.
Because of the bandwidth efficiency and low costs that VOIP technology can provide, businesses are gradually beginning to migrate from traditional copper-wire telephone systems to VOIP systems to reduce their monthly phone costs.
VOIP solutions aimed at businesses have evolved into "unified communications" services that treat all communications—phone calls, faxes, voice mail, e-mail, Web conferences and more—as discrete units that can all be delivered via any means and to any handset, including cell phones. Two kinds of competitors are competing in this space: one set is focused on VOIP for medium to large enterprises, while another is targeting the small-to-medium business (SMB) market.
VOIP runs both voice and data communications over a single network, which can significantly reduce infrastructure, costs.
Voice over Internet Protocol (VoIP) is a method for making phone calls over the Internet or using private networks. Traditional phone calls must travel over a series of switches and circuits owned by the telephone companies, which control the process and the charges. By using VoIP, both businesses and individuals can enjoy a substantial cost savings, especially while making long-distance calls.
VoIP at a technical level is a communications method that uses the competing standards Session Initiation Protocol (SIP) 1 and H.323,2 both of which are widely deployed. The two standards deal with the routing of voice conversations over the Internet, or IP-based networks. The standards define protocols that are derived from traditional phone systems. Signaling protocols replace the traditional private branch exchange (PBX) functions and are carried out by server-based IP PBXs with application software. Examples of this software include Cisco Call Manager, Nortel Call Pilot, and Asterisk. The second type, media protocols, defines the protocols used between two endpoints or VoIP phone devices.
Examples include the Cisco 7900 series phones or a VoIP wireless phone. Vulnerabilities in VoIP have been found in the signaling and media protocols, the call management software, and in the VoIP phone devices themselves.
At its simplest, Voice over IP is the transport of voice using the Internet Protocol (IP), however this broad term hides a multitude of deployments and functionality and it is useful to look in more detail at what VoIP is being used for today. Currently the following types of VoIP applications are in use:
Ø Private users who are using voice over IP for end to end phone calls over the public internet. These users typically trade quality, features and reliability for the fact that the service is very low cost andare generally happy with the service. Although globally the numbers of users taking advantage of this technology is large the density of such users is very low and when compared with the PSTN the call volumes are negligible.
Ø Business users on private networks provided by telecom and datacom providers. These services offer relatively high quality and reliability and are feature rich but come at a price. When compared with the PSTN the call volumes supported by these services are small, however such services are nonetheless commercially successful.
Ø IP trunking solutions used by long haul voice providers. Typically these offerings use private IP networks to connect islands of the PSTN together, e.g. a low cost way of calling the USA from the UK. Customers access these services using traditional black phones but the voice is carried over an IP network.
Although these voice over IP deployments have been successful and each will continue to have its place in the future they have not yet faced the issue of how the wider PSTN could be migrated to an end-to-end voice over IP infrastructure. Providing a voice over IP solution that will scale to PSTN call volumes, offer PSTN call quality and equivalent services, as well as supporting new and innovative services is a significant challenge.
This white paper addresses the central question of how would a carrier deploy a voice over IP service that offered end to end VoIP whilst offering PSTN equivalence. It considers the issues faced by such a carrier andhow the end to end VoIP service fits into the legacy PSTN infrastructure that will undoubtedly remain for a good many years.
Java Media Framework: The Java Media Framework (JMF) is a Java library that enables audio, video and other time-based media to be added to Java applications and applets. This optional package, which can capture, play, stream, and transcode multiple media formats, extends the Java Platform, Standard Edition (Java SE) and allows development of cross-platform multimedia applications.
An initial, playback-only version of JMF was developed by Sun Microsystems, Silicon Graphics, and Intel, and released as JMF 1.0 in 1997. JMF 2.0, developed by Sun and IBM, came out in 1999 and added capture, streaming, pluggable codec’s, and transcoding. JMF is branded as part of Sun's "Desktop" technology of J2SE opposed to the Java server-side and client-side application frameworks. The notable exceptions are Java applets and Java Web Start, which have access to the full JMF in the web browser's or applet viewer’s underlying JRE.
JMF 2.0 originally shipped with an MP3 decoder and encoder. This was removed in 2002, and a new MP3 playback-only plug-in was posted in 2004.
JMF binaries are available under a custom license and the source is available under the SCSL.
The current version ships with four JAR (file format) files, and shell scripts to launch four JMF-based applications:
* JMStudio - A simple player GUI
* JMFRegistry - A GUI for managing the JMF "registry," which manages preferences, plug-ins, etc.
* JMFCustomizer - Used for creating a JAR file that contains only the classes needed by a specific JMF application, which allows developers to ship a smaller application.
JMF is available in an all-Java version and as platform-specific "performance packs", which can contain native-code players for the platform, and/or hooks into a multimedia engine specific to that platform. JMF 2.0 offers performance packs for Linux, Solaris and Windows (on SPARC).
JMF abstracts the media it works with into Data Sources (for media being read into JMF) and Data Sinks (for data being exported out). It does not afford the developer significant access to the particulars of any given format; rather, media is represented as sources (themselves obtained from URL's) that can be read in and played, processed, and exported (though not all codec’s support processing and transcoding).
A Manager class offers static methods that are the primary point-of-contact with JMF for applications.
Literature Survey:Voice over Internet Protocol (VoIP) is a general term for a family of transmission technologies for delivery of voice communications over IP networks.
Ø 1974 — The Institute of Electrical and Electronic Engineers (IEEE) published a paper titled "A Protocol for Packet Network Interconnection."
Ø 1981 — IPv4 is described in RFC 791. * 1985 — The National Science Foundation commissions the creation of NSFNET.
Ø 1995 — Vocal Tec releases the first commercial Internet phone software.
Ø 1996 —ITU-T begins development of standards for the transmission and signaling of voice communications over Internet Protocol networks with the H.323 standard.US telecommunication companies petition the US Congress to ban Internet phone technology.
Ø 1997 — Level 3 began development of its first soft switch, a term they coined in 1998.
Ø 1999 —The Session Initiation Protocol (SIP) specification RFC 2543 is released.Mark Spencer of Digium develops the first open source Private branch exchange (PBX) software (Asterisk).
Ø 2004 — Commercial VOIP service providers proliferate.
Ø 2005 — OpenSER (later Kamailio and OpenSIPS) SIP proxy server is forked from the SIP Express Router.
Ø 2006 — Free SWITCH open source software is released.
Voice over Internet Protocol (VoIP) is a technology for communicating using “Internet protocol” instead of traditional analog systems. Some VoIP services need only a regular phone connection, while others allow you to make telephone calls using an Internet connection instead. Some VoIP services may allow you only to call other people using the same service, but others may allow you to call any telephone number - including local, long distance, wireless, and international numbers.
The FCC has worked to create an environment promoting competition and innovation to benefit consumers and, where necessary, has acted to ensure that VoIP providers comply with important public safety requirements and public policy goals.
For example, due to reports that some VoIP subscribers were unable to access 911 emergency services, in June 2005 the FCC imposed 911 obligations on providers of “interconnected” VoIP services – VoIP services that allow users generally to make calls to and receive calls from the regular telephone network. (You should know, however, that 911 calls using VoIP are handled differently than 911 calls using your regular telephone service.
The FCC requires interconnected VoIP providers and telephone companies that obtain numbers for them to comply with Local Number Portability (LNP) rules. Beginning in late summer 2010, VoIP providers, as well as wireless and wireline providers, must shorten the porting period for “simple” ports from the current four days to one business day. The new deadline applies to all simple ports – including “intermodal” ports such as wireline to wireless, wireless to wireline, wireline or wireless to VoIP, or any other combination. Simple ports generally do not involve more than one line or more complex adjustments to telephone switching equipment. VoIP providers must also contribute to funds established to share LNP and numbering administration costs among all telecommunications providers benefiting from these services.
More than 30 years ago Internet didn't exist. Interactive communications were only made by telephone at PSTN line cost.
Data exchange was expansive (for a long distance) and no one had been thinking to video interactions (there was only television that is not interactive, as known).
Few years ago we saw appearing some interesting things: PCs to large masses, new technologies to communicate like cellular phones and finally the great net: Internet; people begun to communicate with new services like email, chat, etc. and business reborned with the web allowing people buy with a "click" to destination Many years ago we discovered that sending a signal to a remote destination could have be done also in a digital fashion: before sending it we have to digitalize it with an ADC (analog to digital converter), transmit it, and at the end transform it again in analog format with DAC (digital to analog converter) to use it.
VoIP works like that, digitalizing voice in data packets, sending them and reconverting them in voice at destination.
Digital format can be better controlled: we can compress it, route it, convert it to a new better format, and so on; also we saw that digital signal is more noise tolerant than the analog one (see GSM vs. TACS).
TCP/IP networks are made of IP packets containing a header (to control communication) and a payload to transport data: VoIP use it to go across the network and come.
Not really. H.323 standard recommendations were developed for IP network video transport in 1996
•In 1998 significant revisions were made to standardize the use of voice codec’s.
•This “standardization” prompted growth in the VoIP developers arena and the hope for a “new” revolutionary product.
•VoIP has been around for many years in proprietary formats with gateways and “piggy backing” on existing data networks.
•An example of legacy VoIP products are the MICOM Communications Corporation’s Network Integration products.
McAfee Labs first observed an increase in VoIP vulnerabilities during the end of 2006 and that trend has continued through today. We can credit part of this increase to better tools for finding VoIP vulnerabilities, yet this upward trend should be largely attributed to the growing number of VoIP installations.
According to a report by Infonetics Research, overall enterprise telephony grew more than 8 percent in the second and third quarters of 2008. Vendors Cisco, Avaya, and Nortel have been consistently in the top three of enterprise vendor deployment. Recently, Cisco took the top spot overall with growth of 19 percent in the third quarter of 2008.
Avaya grew 10 percent, with Nortel completing the top three.3 it’s hardly surprising that products from these three vendors have the majority of known VoIP vulnerabilities.
In 2001, we first saw VOMIT (Voice Over Misconfigured Internet Telephones).4 This tool takes the network traffic dump of a Cisco IP phone conversation and converts it to a file that can be played on
Ordinary sound players. The tool supported only H.323/G.711, or Cisco IP phones, although tools such as VoIPong work for SIP and the media transport protocol/RTP.5
The example uses WireShark (formerly Ethereal), an open-source network-analysis tool (or sniffer).
Eavesdropping attacks can occur because the media transport protocol that carries the conversation lacks encryption in many default configurations. This is the case when using RTP as the media transport
layer. For a superior solution, you should use secure RTP (SRTP), which provides both encryption
MPLS defines label-switched paths, which are simple uni-directional forwarding paths constructed by wrapping ATM, IP, and other transport protocols packets in MPLS frames. MPLS identifies each frame with a label. The ingress label edge router (LER) provisions the labels and distributes them to label switching routers (LSR) using a signaling protocol such as LDP or Resource Reservation Protocol-traffic engineering (RSVP-TE) prior to enabling transport across the path. The label distribution process involves an automated sequence of resource requests and acknowledgements that create a path between two points in the network. When using RSVP-TE, QoS parameters may be specified as a requirement to each LSR. When acknowledged, these QoS parameters represent an agreement to provide that level of QoS, or FEC, to packets forwarded along the path.
Service providers can construct customized LSPs that support specific application requirements.
Network managers can design LSPs to minimize the number of hops, meet certain bandwidth requirements, support precise performance requirements, bypass potential points of congestion, direct traffic away from the default path selected by the IGP, or simply force traffic across certain links or nodes in the network.
An important benefit of the label-swapping forwarding algorithm is its ability to take any type of user traffic, associate it with an FEC, and map the FEC to an LSP that has been specifically designed to satisfy the FEC’s requirements. Adding DSCP support to the MPLS network allows the network to populate a single LSP with multiple FECs.
The MPLS-TE approach also enables network administrator to provision MPLS fast reroute (FRR) paths for LSPs and associated backup paths while minimizing physical LSR overlap between primary and backup paths. FRR limits path outage times to milliseconds by pre-negotiating resource borrowing from LSR neighbors and localizing the event signaling that implements the FRR operation.
Deploying technologies based on label-swapping forwarding techniques offers network administrators precise control over traffic flow in their networks. This unprecedented level of control results in a network that operates more efficiently and provides more predictable service.
More than 30 years ago Internet didn't exist. Interactive communications were only made by telephone at PSTN line cost.
Few years ago we saw appearing some interesting things: PCs to large masses, new technologies to communicate like cellular phones and finally the great net: Internet; people begun to communicate with new
services like email, chat, etc. and business reborned with the web allowing people buy with a "click".Most wired voice communications were carried over the Public Switched Telephone Network (PSTN), which relies on switches to establish a dedicated circuit between a source and a destination to carry an analog voice signal. More recently, Voice over Internet Protocol (VoIP) was developed as a means for enabling speech communication using digital, packet-based, Internet Protocol (1P) networks such as the Internet. A principle advantage of IP is its efficient bandwidth utilization. VoIP may also be advantageous where it is beneficial to carry related voice and data communications over the same channel, to bypass tolls associated with the PSTN, to interface communications originating with Plain Old Telephone Service (POTS) with applications on the Internet, or for other reasons. As discussed in this specification, the problems and solutions related to VoIP may also apply to Facsimile over Internet Protocol (FoIP)
Scope and Major objective of Project:
Scope: The main aim of this project is to represent the working of the basic use of the internet protocol through computers and talking devices, dealing with the issues which occur during the transmission of voice over internet.
We cannot know what the future is, but we can try to image it with many computers, Internet almost everywhere at high speed and people talking (audio and video) in a real time fashion. We only need to know
what will be the means to do this: UMTS, VoIP (with video extension) or other? Anyway we can notice that Internet has grown very much in the last years, it is free (at least as international means) and could be the right communication media for future.
1. It provides a global interaction with issues related with voice transmission.
2. All companies can make use of the software to resolve their queries and use the service in more efficient way.
3. It can be used in defense services.
4. All educational institutions can make use of this software to improve their academic knowledge.
5. The government offices can also make use of this software .
Major Objective:The use of Internet Protocol for transmission of voice over the internet and show how this phenomena works is the basic objective of this project.
Our basic aim is to improve the current condition of the software such as:
Ø Operational cost: VOIP can be a benefit for reducing communication and infrastructure costs.
Ø Flexibility: VOIP can facilitate tasks and provide services that may be more difficult to implement using the PSTN.
Ø The ability to transmit more than one telephone call over a single broadband connection without the need to add extra lines.
Integration with other services available over the Internet, including video conversation, message or data file exchange during the conversation, audio conferencing, managing address books, and passing information about whether other people are available to interested parties.
Problem Analysis and Complete:
software requirement specification: About existing condition of the software application: The problem which we face today is as follow:
1. Quality of service:By default, IP routers handle traffic on a first-come, first-served basis. When a packet is routed to a link where another packet is already being sent, the router holds it on a queue. Should additional traffic arrive faster than the queued traffic can be sent, the queue will grow. If VOIP packets have to wait their turn in a long queue, intolerable latency may result.One way to avoid this problem is to simply ensure that the links are fast enough so that queues never build even in the worst case. This usually requires additional mechanisms to limit the amount of traffic entering the network, and for voice traffic this is usually done by limiting the number of simultaneous calls. Another approach is to use quality-of-service (QoS) mechanisms such as Diffserv to give priority to VOIP packets and other latency-sensitive traffic so they can "jump the line" and be transmitted ahead of any bulk data packets already in the queue. This can work quite well when voice constitutes a relatively small fraction of the total network load, as it usually does in today's Internet.
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., cablemodems) 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.
The susceptibility of phone service to power failures is a common problem even with traditional analog service in areas where many customers purchase modern handset units that operate wirelessly to a base station, or that have other modern phone features, such as built-in voicemail or phone book features.
3. Lack of redundancy: With the current separation of the Internet and the PSTN, a certain amount of redundancy is provided. 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.
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.
5.Security: 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 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.
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 to 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.The Voice VPN solution provides secure voice for enterprise VOIP networks by applying IPSec encryption to the digitized voice stream.
6. Support for other telephony devices: Another challenge for VOIP implementations is the proper handling of outgoing calls from other telephony devices such as DVR boxes, satellite television receivers, alarm systems, conventional modems and other similar devices that depend on access to a PSTN telephone line for some or all of their functionality. These types of calls sometimes complete without any problems, but in other cases they fail. If VOIP and cellular substitution becomes very popular, some ancillary equipment makers may be forced to redesign equipment, because it would no longer be possible to assume a conventional PSTN telephone line would be available in consumer's homes.As the popularity of VOIP grows, and PSTN users switch to VOIP in increasing numbers, governments are becoming more interested in regulating VOIP in a manner similar to PSTN services.
Another legal issue that the US Congress is debating concerns changes to the Foreign Intelligence Surveillance Act. The issue in question is calls between Americans and foreigners. The National Security Agency (NSA) isn't authorized to tap Americans' conversations without a warrant—but the Internet, and specifically VOIP doesn't draw as clear a line to the location of a caller or a call's recipient as the traditional phone system does.
As VOIP's low cost and flexibility convinces more and more organizations to adopt the technology, the line separating the NSA's ability to snoop on phone calls will only get blurrier. VOIP technology has also increased security concerns because VOIP and similar technologies have made it more difficult for the government to determine where a target is physically located when communications are being intercepted, and that creates a whole set of new legal challenges.
In the US, the Federal Communications Commission now requires all interconnected VOIP service providers to comply with requirements comparable to those for traditional telecommunications service providers. VOIP operators in the US are required to support local number portability; make service accessible to people with disabilities; pay regulatory fees, universal service contributions, and other mandated payments; and enable law enforcement authorities to conduct surveillance pursuant to the Communications Assistance for Law Enforcement Act (CALEA). "Interconnected" VOIP operators also must provide Enhanced 911 service, disclose any limitations on their E-911 functionality to their consumers, and obtain affirmative acknowledgements of these disclosures from all consumers. VOIP operators also receive the benefit of certain US telecommunications regulations, including an entitlement to interconnection and exchange of traffic with incumbent local exchange carriers via wholesale carriers. Providers of "nomadic" VOIP service — those who are unable to determine the location of their users — are exempt from state telecommunications regulation.
Throughout the developing world, countries where regulation is weak or captured by the dominant operator, restrictions on the use of VOIP are imposed, including in Panama where VOIP is taxed, Guyana where VOIP is prohibited and India where its retail commercial sales is allowed but only for long distance service.In Ethiopia, where the government is monopolizing telecommunication service, it is a criminal offense to offer services using VOIP. The country has installed firewalls to prevent international calls being made using VOIP. These measures were taken after the popularity of VOIP reduced the income generated by the state owned telecommunication company.
In the European Union, the treatment of VOIP service providers is a decision for each Member State's national telecoms regulator, which must use competition law to define relevant national markets and then determine whether any service provider on those national markets has "significant market power" (and so should be subject to certain obligations). A general distinction is usually made between VOIP services that function over managed networks (via broadband connections) and VOIP services that function over unmanaged networks (essentially, the Internet).
VOIP services that function over managed networks are often considered to be a viable substitute for PSTN telephone services (despite the problems of power outages and lack of geographical information); as a result, major operators that provide these services (in practice, incumbent operators) may find themselves bound by obligations of price control or accounting separation.
VOIP services that function over unmanaged networks are often considered to be too poor in quality to be a viable substitute for PSTN services; as a result, they may be provided without any specific obligations, even if a service provider has "significant market power".
The relevant EU Directive is not clearly drafted concerning obligations which can exist independently of market power (e.g., the obligation to offer access to emergency calls), and it is impossible to say definitively whether VOIP service providers of either type are bound by them. A review of the EU Directive is under way and should be complete by 2007.
Detailed Design: Software design sits at the technical kernel of the software engineering process and is applied regardless of the development paradigm and area of application. Design is the first step in the development phase for any engineered product or system. The designer’s goal is to produce a model or representation of an entity that will later be built. Beginning, once system requirement have been specified and analyzed, system design is the first of the three technical activities -design, code and test that is required to build and verify software. The importance can be stated with a single word “Quality”. Design is the place where quality is fostered in software development. Design provides us with representations of software that can assess for quality. Design is the only way that we can accurately translate a customer’s view into a finished software product or system. Software design serves as a foundation for all the software engineering steps that follow. Without a strong design we risk building an unstable system – one that will be difficult to test, one whose quality cannot be assessed until the last stage.
During design, progressive refinement of data structure, program structure, and procedural details are developed reviewed and documented. System design can be viewed from either technical or project management perspective. From the technical point of view, design is comprised of four activities – architectural design, data structure design, interface design and procedural design.
Object Oriented Design:Object-oriented design is part of OO methodology and it forces programmers to think in terms of objects, rather than procedures, when they plan their code. An object contains encapsulated data and procedures grouped together to represent an entity. The 'object interface', how the object can be interacted, is also defined. An object-oriented program is described by the interaction of these objects. Object-oriented design is the discipline of defining the objects and their interactions to solve a problem that was identified and documented during object-oriented analysis.
To setup a VoIP communication we need:
1. First the ADC to convert analog voice to digital signals (bits)
Now the bits have to be compressed in a good format for transmission: there is a number of protocols
we'll see after.
Here we have to insert our voice packets in data packets using a real−time protocol (typically RTP
over UDP over IP)
4. We need a signaling protocol to call users: ITU−T H323 does that.
At RX we have to disassemble packets, extract datas, then convert them to analog voice signals and
send them to sound card (or phone)
All that must be done in a real time fashion cause we cannot waiting for too long for a vocal answer
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