What Is Video Conference Computer Science Essay

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Video conference is nothing but visual collaboration. It has a set of interactive telecommunication technology which allows two or more locations to interact via two video and audio transmissions concurrently. It has also been called visual collaboration and is a type of groupware.

Video conferencing is come from video phone call. It is designed for conference. It is an intermediate form of video telephony.

Video conferencing uses at different sites for together meeting, they use the telecommunication of audio and video. Video conferencing is useful to display information, and share documents. It mostly used at private office for video and audio conference meetings.

It consist of two closed circuit system integrates the video and audio connected via coaxial cable or radio.

NASA used two radio frequency (UHF or VHF) links, one in each direction. Reporting from distant location, where TV channels use TV conferencing. And it was very expensive. It couldn't be used for telemedicine, distance education and business meeting.

In 1980 it was possible using digital telephony transmission networks, like as ISDN, guarantying a minimum bit rate of 128kb/sec for compressed video and audio transmission. These equipments like software and network are very expensive and standard based technology.

IP based video conferencing invented in 1990. That time efficient video compression technology, personal computer based video conferencing was developed.

In 1992 CU-See Me was developed; it is an internet video conferencing client. It makes point to point video calls without a server or makes multi-point calls through server first called a "reflector" and later called conference server. It was developed by Tim Dorcey et al.

In 2000, video telephony was popularized via free internet like skype and ichat, if we want to video conference, where every location with internet connections. That time was very low quality.

In 2005, very high definition video conferencing systems was produced by life size communications, it can able to 30 frames per second at 1280x720 resolutions.

What technologies are used?

Core technologies are used in a videoconferencing system is digital compression of audio and video streams in real time. Compression is performed by the hardware and software is called Codec. The resulting digital stream of 1s and 0s is subdivided into label pockets, which later transmitted through a digital network of ISDN or IP. The transmission line allows for the use of Plain Old Telephone System, where use of audio modems in some low speed applications like as video telephony.

Video conference components are:

Video input: web cam, video camera

Video output: television, projector and computer monitor.

Audio input: micro phones, pre amp audio outlet.

Audio output: loudspeakers

Data transmission: analog or digital telephone network, LAN or internet.

Dedicated system and desktop systems are two kinds of video conferencing systems:

Dedicated system:

Required components are packaged into single piece of equipment, these cameras can be controlled at a distance to pan left and right, tilt up and down, and zoom, where console with a high quality remote controlled video camera. This is called as PTZ cameras. Where console contains the control computer, all electrical interfaces and hardware and software based codec. Console connected to all directional microphones, as well as loudspeakers with loudspeaker and video projector.

There are several types of dedicated videoconferencing devices:

Non portable and expensive devices are used at auditoriums and large rooms are called large video conference.

Portable devices are used at small meeting rooms is called small group video conference.

Personally we use the video conference, we combine the microphone, and loudspeakers into console are called individual video conference.

Desktop systems:

Video conference require devices like microphones, loudspeakers and different type and range cameras are connect to the normal desktop systems and transferred into the video conference device. Normal desktop systems are with H.323 standard and contain codec and transmission interface is called as e-meeting.

Layers of conferencing system:

Conferencing system are divided into different type of layers, they are:

User interface.

Conference control.

Signal plane.

Media plane.

User interface:

It is responsible for voice and graphical in video conferencing time. It has a number of different uses; it could be used for scheduling, making the call and setup. Administrator controls the other three layers through this layer.

Conference control:

This layer is long with first layer and performs resource allocation, routing and management, creates meetings.

Signal plane:

Signals are not limited to H.323 and SIP(session initiation protocol) protocols. And contain the stacks that signal different end points to create call or conference. This plane control incoming or outgoing connections as well as session parameters.

Media plane:

This layer manage protocols like UDP (user data gram packets), RTCP (real time transport control protocol), and controls the audio and video mixing and streaming. The RTCP and UDP carry information like payload type i.e, codec, frame rate, video size other side it control the protocol quality for detecting errors during streaming.

Multipoint video conferencing:

Multipoint video control is called the multipoint control unit (MCU). This is along with three or more remote points. These are interconnected with each other and calls from different sources. This MCU is for IP and ISDN based video conferencing. MCU is a combination of hardware and software and it can handle calls, transporting data rates and protocols, and features like continues presence, where multiple parties at onscreen at once and it is a hardware device.

MCU consists of two logical components they are:

Single multipoint controller (MC).

Multipoint processor (MP).

Multipoint is activates on the signaling plane, where it controls the conferencing and manages creation of conference and endpoint signaling. Every endpoint in the network negotiates parameters in the components. It operates on the media plane and receives media from each end point. It generates output streams from each endpoint. Information is redirects to other endpoint in the conference.

Some systems have a multipoint conference with no MCU, each station multipoint call exchanges video and audio directly with other stations with no central manager, when H.323 techniques are used in this is called as decentralized multipoint. This advantages is generates higher quality audio and video users can make ad-hoc multipoint calls without any worry about control of an MCU.

Modes of video conferencing:

They have several modes that are used:

VAS (voice activate switch).

Continuous presence.

In VAS mode, these end point switches can be seen by the other end points and if there are four peoples are in a conference, they talking, when the only one will be seen in the conference. Other participants will be seen the location with the loudest voice.

At the same point multiple participants displayed by the presence mode. In this mode, MP puts together the stream from different endpoints and together into a single video image. It sends the same type of images to all participants, where these images are called the layouts and it depends on the number of participants in the conference.

Echo cancellation:

AEC (acoustic echo cancellation) is the fundamental feature of the professional video conferencing system. It can be reflected by the source, where source interface with new wave created by source. It is an algorithm, it is able to detect, when sounds of the video conferencing codec. Some time is delay, where it is came from audio output of the same system. This can lead to several problems:

Other side person hearing their own voice coming at them

Representation voice channel useless as it become hard to understand.

Echo cancelation is a processor-intensive task that usually works over a narrow range of sound delays.

Video conference standards:

Video conferencing standards are ITU (international telecommunication union) is called as the standard for PSTN (public switched telephone networks). It is accessible with high speed internet connection and video conferencing over combined services digital networks.

Compression standard is H.264 SVC (scalable video coding) enables video conferencing systems to achieve highly error resilient IP video transmission over the public internet without quality of service improved lines. It has enabled wide scale deployment of high definition desktop video conferencing and made possible new architectures. Where it reduces a latency between transmitting source and receiver.

Otherwise an IP video conference attractive factor is that it is easy set-up a web conference with live video conference calls for use in data collaboration. This integrated technology enables user to have a richer environment for live meeting, collaboration and presentations.

In 2010 may19th, UCIF (unified communication interoperability forum) launched a nonprofit alliance between communication venders. The organization vision is to maximize the interoperability of UC based on existing standards. UCIF founding members include Microsoft, HP, polycom, Logitech communications and juniper networks.

ITU V.80 video conferencing is generally compatibilized with H.324 standard point to point video telephony over regular phone lines.

Video conference protocols:

H.323:

Telecommunication standardization sector (ITU-T) is recommended the H.323. In any pocket network, this protocol offers audio and video communication. The H.323 standard addresses control the calls signaling, multimedia transport, and bandwidth control for point to point and multi point conferences.

Voice and video conferencing equipment manufacturers are implemented widely and used with various internet real time application like as GnuGK and net meeting. And it is widely deployed worldwide by service providers and enterprises for both voice and video services over IP networks.

H.32x is a series of protocol and it is a part of the ITU-T, where address multimedia communications over ISDN, PSTN, SS7 and 3G mobile networks.

In the context of H.323, an IP based PBX may be a call controller element, where it provides services to telephone or videophone. It provides some services like call tracker, park, pick-up and hold.

H.323's strength lies in multimedia communication functionality designed specifically for IP networks and provides basic telephony functionality and interoperability.

In 1996, the first version was published by the ITU with enabling videoconferencing capabilities over a LAN (local area network), but it was quickly adopted by the industry as a means of transmitting voice communication over a variety of IP networks.

H.323 is commonly referred to as H.323v6. one strength of H.323 was the relatively availability of a set of standards and was the first VOIP standard to adopt the IETE (internet engineering task force) standard real-time transport protocol to transport audio and video over IP networks.

Protocols:

H.323 is a system of specification that explains the uses of several ITU-T and IETF protocols. This protocol compromise the cores of almost H.323 system are:

H.225.0 RAS(registration, administration and status), where it is between H.323 and gatekeeper to provide admission control service and resolution.

H.225.0 call signaling, where it is used between two H.323 entities in order to establish communication.

H.245 control protocol for multimedia communication, where it describes the messages and procedures used for capacity of exchange, logical channels are closing and opening for audio and video and data, control and indications.

RTP (real-time transport protocol), where it is used for sending or receiving multimedia information between two entities.

Security within H.323 is explains the H.235 series, security includes signaling and media.

Dual stream use in video conferencing is explains the H.235. Usually one for live video and other for images.

Different supplementary services explained by the H.450.

Optional extensions may be implemented by an end point is explained by the H.460. H.460.17, H.460.18, H.460.19 for NAT(network address translation).

Codec:

H.323 uses both ITU and defined Codec and it defined the outside ITU. These are implemented by H.323, it includes:

Audio Codecs: G.711, G.729, G.723.1, G.726, speex

Text Codecs: T.140

Video Codecs: H.261, H.263, H.264

Video communication must be provided by the H.323 terminals and it has encoding and decoding speech according to H.261 QCIF. Al terminals must be capacity of transmitting and receiving A-law and µ-law.

Architecture of H.323:

Several network elements are work together in order to deliver multimedia communication and it is defined by the H.323 system. That network element is terminals, MCU's (multipoint control units), gateways, gatekeepers and border elements and is referred to as endpoints. At least two elements required to enable communication between two people.

Network elements of H.323:

http://www.urdunetworks.com/index.php?option=com_awiki&view=mediawiki&article=H.323%3Fqsrc%3D3044&Itemid=55

H.323 terminals are most fundamental elements in any H.323 system, they may be exist in the form of simple IP phone or high definition video conferencing system.

This terminal is referred to as protocol stack, where it implements the functionality. Protocol stack would include an implementation of the basic protocol defined in ITU-T. the figure shows the complete stack that provides support for voice, video and various data communication.

Network signaling of H.323:

Binary protocol is defined by the H.323, where it allows for efficient message processing in network elements. This syntax is defined by the ASN.1 and it uses the pocket encoding rules form of message encoding for efficient message encoding on the wire.

Call signaling of H.225.0:

http://www.urdunetworks.com/index.php?option=com_awiki&view=mediawiki&article=H.323%3Fqsrc%3D3044&Itemid=55

the address of the end point is resolved, H.225.0 call signaling used by the end point and in order to establish communication with the remote entity. And these messages are:

Setup and setup acknowledge

Call proceeding

Connect

Altering

Information

Release complete

Facility

Progress

Status and status inquiry

Notify

In the simple form, an H.323 may be called established. The end point (EP) on the left initiated communication with the gate way on the right and the gate way connected the call with the called party. Most calls using fast connect procedures defined with H.323 and can be established with as few as 2 or 3 messages.

Use cases:

Voice over IP address and H.323:

VOIP (voice over internet protocol) explain the transmission of voice using the internet or other pocket switched networks. One of the standards is H.323 for using VOIP. VOIP requires an internet connection and subscribe to a VOIP service provider and client. The service provider offers the connection to other VOIP services. Most of It charges a monthly fee, and then additional charge, when the calls are made. VOIP is not necessary between two enterprises, when using VOIP.

Video conference services and H.323:

Video conference is a set of telecommunication technology and allowing more than two locations for interact via two way video and audio transmissions similarly. The basic type of video conferencing system is VTC system. In this, all components are combined into a single piece while desktop VTC system are adds-on to normal pc's, and then transforming them into a VTC device. Similarly video conferencing over three or more remote points is possible by means of MCU (multipoint control protocol). There are MCU bridges for IP and ISDN based video conferencing. in broad band, we using H.323 based IP video conferencing . it accessible to high speed internet connection, like as DSL. It is used in different situations like as distance education, telemedicine and business.

International conference:

Industry to using the international video conferences by using the H.323 that is larger than the typical video conference. One of the most widely conference is called mega conference.

Video conference protocols:

H.323:

Telecommunication standardization sector (ITU-T) is recommended the H.323. In any pocket network, this protocol offers audio and video communication. The H.323 standard addresses control the calls signaling, multimedia transport, and bandwidth control for point to point and multi point conferences.

Voice and video conferencing equipment manufacturers are implemented widely and used with various internet real time application like as GnuGK and net meeting. And it is widely deployed worldwide by service providers and enterprises for both voice and video services over IP networks.

H.32x is a series of protocol and it is a part of the ITU-T, where address multimedia communications over ISDN, PSTN, SS7 and 3G mobile networks.

In the context of H.323, an IP based PBX may be a call controller element, where it provides services to telephone or videophone. It provides some services like call tracker, park, pick-up and hold.

H.323's strength lies in multimedia communication functionality designed specifically for IP networks and provides basic telephony functionality and interoperability.

In 1996, the first version was published by the ITU with enabling videoconferencing capabilities over a LAN (local area network), but it was quickly adopted by the industry as a means of transmitting voice communication over a variety of IP networks.

H.323 is commonly referred to as H.323v6. one strength of H.323 was the relatively availability of a set of standards and was the first VOIP standard to adopt the IETE (internet engineering task force) standard real-time transport protocol to transport audio and video over IP networks.

Protocols:

H.323 is a system of specification that explains the uses of several ITU-T and IETF protocols. This protocol compromise the cores of almost H.323 system are:

H.225.0 RAS(registration, administration and status), where it is between H.323 and gatekeeper to provide admission control service and resolution.

H.225.0 call signaling, where it is used between two H.323 entities in order to establish communication.

H.245 control protocol for multimedia communication, where it describes the messages and procedures used for capacity of exchange, logical channels are closing and opening for audio and video and data, control and indications.

RTP (real-time transport protocol), where it is used for sending or receiving multimedia information between two entities.

Security within H.323 is explains the H.235 series, security includes signaling and media.

Dual stream use in video conferencing is explains the H.235. Usually one for live video and other for images.

Different supplementary services explained by the H.450.

Optional extensions may be implemented by an end point is explained by the H.460. H.460.17, H.460.18, H.460.19 for NAT(network address translation).

Codec:

H.323 uses both ITU and defined Codec and it defined the outside ITU. These are implemented by H.323, it includes:

Audio Codecs: G.711, G.729, G.723.1, G.726, speex

Text Codecs: T.140

Video Codecs: H.261, H.263, H.264

Video communication must be provided by the H.323 terminals and it has encoding and decoding speech according to H.261 QCIF. Al terminals must be capacity of transmitting and receiving A-law and µ-law.

Architecture of H.323:

Several network elements are work together in order to deliver multimedia communication and it is defined by the H.323 system. That network element is terminals, MCU's (multipoint control units), gateways, gatekeepers and border elements and is referred to as endpoints. At least two elements required to enable communication between two people.

Network elements of H.323:

http://www.urdunetworks.com/index.php?option=com_awiki&view=mediawiki&article=H.323%3Fqsrc%3D3044&Itemid=55

H.323 terminals are most fundamental elements in any H.323 system, they may be exist in the form of simple IP phone or high definition video conferencing system.

This terminal is referred to as protocol stack, where it implements the functionality. Protocol stack would include an implementation of the basic protocol defined in ITU-T. the figure shows the complete stack that provides support for voice, video and various data communication.

Network signaling of H.323:

Binary protocol is defined by the H.323, where it allows for efficient message processing in network elements. This syntax is defined by the ASN.1 and it uses the pocket encoding rules form of message encoding for efficient message encoding on the wire.

Call signaling of H.225.0:

http://www.urdunetworks.com/index.php?option=com_awiki&view=mediawiki&article=H.323%3Fqsrc%3D3044&Itemid=55

the address of the end point is resolved, H.225.0 call signaling used by the end point and in order to establish communication with the remote entity. And these messages are:

Setup and setup acknowledge

Call proceeding

Connect

Altering

Information

Release complete

Facility

Progress

Status and status inquiry

Notify

In the simple form, an H.323 may be called established. The end point (EP) on the left initiated communication with the gate way on the right and the gate way connected the call with the called party. Most calls using fast connect procedures defined with H.323 and can be established with as few as 2 or 3 messages.

Use cases:

Voice over IP address and H.323:

VOIP (voice over internet protocol) explain the transmission of voice using the internet or other pocket switched networks. One of the standards is H.323 for using VOIP. VOIP requires an internet connection and subscribe to a VOIP service provider and client. The service provider offers the connection to other VOIP services. Most of It charges a monthly fee, and then additional charge, when the calls are made. VOIP is not necessary between two enterprises, when using VOIP.

Video conference services and H.323:

Video conference is a set of telecommunication technology and allowing more than two locations for interact via two way video and audio transmissions similarly. The basic type of video conferencing system is VTC system. In this, all components are combined into a single piece while desktop VTC system are adds-on to normal pc's, and then transforming them into a VTC device. Similarly video conferencing over three or more remote points is possible by means of MCU (multipoint control protocol). There are MCU bridges for IP and ISDN based video conferencing. in broad band, we using H.323 based IP video conferencing . it accessible to high speed internet connection, like as DSL. It is used in different situations like as distance education, telemedicine and business.

International conference:

Industry to using the international video conferences by using the H.323 that is larger than the typical video conference. One of the most widely conference is called megaconference.

Sip:

The development of video conferencing signaling protocols related to VOIP. Early VOIP systems and many cases on existing protocols based on internet telecommunication. And then derivates the H.323, MGCP(media gateway control protocol), and it for session establishment and management. H.323 was developed extensive, but it leads complexity to internet. IETF(internet engineering task force) to develop a new protocol for session management and establishment. It has some constraints, they are:

Optimized for IP network,

Capable to control existing protocols,

Flexibility to support a range of media type, including voice and video.

The result of the IETF, SIP was developed and designed to needs of the mbone, in this integrate the voice and video over IP application and multicast network for audio and video over the internet. In this, SIP is a de-facto standard for system interconnectivity.

Many video conferencing systems still rely on H.323 for signaling and session management. As UC gains video conferencing vendors are diverging over their capability to integrate their system into SIP based architecture. Vendors capable to support SIP across different products without requiring gateway or Trans coding between SIP and non SIP systems.

SIP controls sessions between systems and enables customer to integrate a variety of end points with SIP based back end servers. As video conferencing systems starts to migrate to SIP.

What is SIP:

SIP is defined in IETF request for comments (RFC) 3261; it provides a common signaling language for managing, initiating and terminating any kind of rich media session, be it voice, video, or instant messaging.

SIP actually comprises two protocols SIP for initiating and terminating a session between endpoints, type of sessions, session parameters like codecs or encryption are defines the SDP (session description protocol). SDP allows application developer to control the pre-existing protocol, H.323 used in many implementation of SIP. Even though SIP is designed as an end-end protocol, variety of servers manages different locations for SIP based systems.

To implement security policies between endpoints, sip implementation, and provide translation between different fundamental media protocols, where SIP makes use of a session border controller.

End points of SIP find each other by means of a SIP uniform resource indicator (URI), where translates a SIP name to an IP address more in the same way a uniform resource locator (URL) works in the web.

IETF designed SIP from the start as an IP centric protocol, means that it follow the IP architectural model based on intelligence at the endpoints rather than in the core. SIP does not require components like as multi-point controls units, it support a variety of flexible deployment scenario; from point to point with a proxy server for management and control.

SIP also controls the HTTP (hyper text transport protocol), SIP messages are text based as opposed to H.323's binary formats. And it easier to implement, monitor and extend than H.323.

SIP and unified communication:

SIP has appear as the basis for unified communications for interconnecting various real time communications applications including voice, video, messaging and conferencing to providing a standard method. One SIP/SIMPLE based application can share to other SIP/SIMPLE application, otherwise messaging application based on extendable message presence protocol (XMPP).

To broaden the capacity of voice and video control away from traditional phones or endpoints to using SIP by developers increasing. SIP based controls can extend into other applications as well. SIP based clients to provide presence for developers can embed. Embed the same sort of functionality into their system by enterprise application developers, customer resource management applications. Customer resource management application can see a record of past interactions, when looking at a record by customer service agent. Who have spoken with the customer determine the presence status of those and initiate a text, voice or video communication session to one or more co-workers, all within the CRM.

Video conferencing services are based on SIP as well, meaning of SIP has looked as not only the protocol of choice for internal systems, internal systems are interconnected to external application.

Video is the component of a unified communications means that enterprise architects must co-ordinate video conferencing planting with those within their organization responsible for UC initiate.

Combining SIP and H.323:

Most of the organizations have large operations of H.323 video conferencing systems with no cost justification for replacing H.323 systems is not a visible strategy.

Otherwise, the optimal approach is one that interconnects H.323 and SIP systems into failure architecture, with no feature disparity and no awareness from the end-user perspective of whether endpoints are using SIP or H.323 protocols provides the lowest total cost of ownership across interoperability by the gateways. Organize an architecture based on SIP and H.323 interworking reduces integration complexity by allowing IT architecture to support existing protocols for as long as necessary as transition to SIP, while also capable to integrate the both H.323 and SIP endpoints into their UC architectures.

VOIP:

In this, number of protocols working to offer for VOIP communication services. Every device in the world uses a standard called real time protocol (RTP) for transmitting audio and video packets between computers through communication is defined by the IETF in RFC 3550. The payload format for a number of CODECs are defined in 3551, specifications of payload format are defined in documents and published by the ITU and in other IETF RFCs. Address issues like packet order and provides mechanisms to address delay and jitter.

Consider the one of the area for people communicating over the internet is the potential person to eavesdrop on communication.

Architecture

SIP (Session Initiation Protocol) Overview

SIP is an application-layer control protocol that can establish, modify, and terminate multimedia sessions. SIP can also invite participants to already existing sessions. SIP transparently supports name mapping and redirection services, which supports personal mobility - users can maintain a single externally visible identifier regardless of their network location (see RFC 3261 [1]). SIP can also be used to implement non-real-time services like Instant Messaging and Presence [2][3].

SIP supports five facets of establishing and terminating multimedia communications:

User location

- determination of the end system to be used for communication

User availability

- determination of the willingness of the called party to engage in communications

User capabilities

- determination of the media and media parameters to be used

Session setup

- establishment of session parameters at both called and calling party

Session management

- including transfer and termination of sessions, modifying session parameters, and invoking services

B2BUA High Level Architecture

The B2BUA consists of the following three main logical components:

Answering SIP User Agent

Call Control Logic

Originating SIP User Agent

The following diagram illustrates the B2BUA architecture and the primary components.

The components interact with each other using abstract events. Each User Agent (UA) represents a state machine, which receives SIP messages from the endpoint and converts them into events based on the type of message and the agent's own current state. The Call Control Logic acts as a go-between, passing events between the UAs. Depending on its own current state and the states of the UAs, the Call Control Logic could drop some events, convert even type in transition, or inject additional events. It can also remove one of the UAs and replace it with another one on different stages of the call, which allows implementing such features as failover call routing, controlled call transfer and so on.

Since the number of parameters passed by each event is well defined the B2BUA can isolate call legs from each other, allowing only controlled amount of information to pass through.

This architecture allows implementing different functionality by replacing the Call Control Logic, which consists of a small fraction of the B2BUA code. Two such implementations are described in the next section.

Typicall Call Process

A call is initiated when the Answering SIP UA receives an incoming INVITE message from the Originating SIP Endpoint.

After receiving this message, the Answering SIP UA generates a Try event (2) and passes it to the Call Control Logic, as illustrated in the following diagram.

The Call Control Logic receives the Try event, performs authentication and authorization, creates the Originating SIP UA, modifies the Try event to accommodate for any parameter translation logic, and passes it along with the routing information to the Originating SIP UA (3).

The Originating SIP UA receives the Try event and generates INVITE message (4) as shown in the following diagram.

After the Answering SIP endpoint receives the INVITE message, it starts ringing and sends back an 18x SIP provisional response (5).

The Originating SIP UA receives the message, generates a Ringing event, and passes it to the Call Control Logic (6).

The Call Control Logic receives the Ringing event and passes it to the Answering SIP UA (7), and in response, the Answering SIP UA sends an 18x SIP provisional response to the Originating SIP endpoint (8).

When the user at the Answering SIP endpoint picks up the phone, the endpoint generates a 200 OK SIP response and sends it back to the Originating SIP UA (9).

The UA generates a Connect event and passes it to the Call Control Logic (10), following which the Call Control Logic hands the event over to the Answering SIP UA (11).

The UA sends a 200 OK message to the Originating SIP endpoint (12). At this point, the session is established and endpoints start exchanging RTP media (13).

When either party hangs up, the respective SIP endpoint generates a SIP BYE message and sends the message to the associated SIP UA (14).

The UA generates a Disconnect event, which propagates to the other side of the B2BUA via the Call Control Logic (15), (16) and results in a BYE message, which is sent to the other endpoint (17).

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