Clarisys usb handset driver

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Your support registration has been received and is in the approval process. Your Company Administrator is reviewing your request in order to approve your login. The IPP provides the freedom to use any traditional telephone device to access the power and advanced features of a PC based IP telephone. This device is designed to be used with either cordless or corded phones and can drive multiple phones at once.

This device provides an RJ11 port to connect to standard telephone sets. Loop current, 90V ring voltage, hook supervision, full button integration and Caller ID display are provided. Quality communication is the result of proprietary echo cancellation, noise suppression and voice level compensation algorithms.

We operate either on a consultancy basis, or on a shared risk basis where we retain the intellectual property. We are equally happy developing new systems or providing modifications to our existing ones. Our expertise covers optics, sensors, electronics, firmware, software and the associated mechanical design, which allows us to provide full turn-key solutions for our clients. We also work on OEM designs for business partners to provide solutions optimised for compatibility with their specific surveillance system.

Finally, it has defined standards for supplementary services, the network retains the call state for the duration of the call providing greater call control for the user , and application services are available through the gatekeeper and best-of-breed application platforms. Maintaining call state in the network actually increases the cost to scale.

Many soft phones are proprietary, so they are not widely deployed. Both Megaco Media Gateway Controller and MGCP Media Gateway Control Protocol are protocols for control of elements in a physically decomposed multimedia gateway, which enables separation of call control from media conversion.

This means that the system is composed of a call agent, at least one media gateway whose function is to perform the conversion of media signals between circuits and packets, and at least one signaling gateway when connected to the PSTN. They both embrace a philosophy in which the network is smart and the endpoint is dumb. Services are provided by intelligent network elements.

As usually is the case, it is the result of different parties with different interests and agendas. Megaco is less widely implemented than MGCP. This is in contrast to H. With peer-to-peer protocols, all the intelligence is distributed to the network edge, embedded in the terminating devices or endpoints, requiring only a simple core network and great scalability but reducing the network's control over the user. Megaco addresses the relationship between a media gateway, which converts circuit-switched voice to packet-based traffic, and a media gateway controller also referred to as a call agent or softswitch , which dictates the service logic of that traffic.

Megaco instructs a media gateway to connect streams coming from outside a packet or cell data network onto a packet or cell stream such as RTP. Megaco is similar to MGCP from an architectural standpoint and in terms of the controller-to-gateway relationship, but Megaco supports a broader range of networks, such as ATM. As shown in Figure 9.

Media gateways contain endpoints on which the call agent can create, modify, and delete connections in order to establish and control media sessions with other multimedia endpoints. A media gateway is typically a network element that provides conversion between the audio signals carried on telephone circuits and data packets carried over the Internet or over other packet networks.

The call agent can instruct the endpoints to detect certain events and generate signals. The endpoints automatically communicate changes in service state to the call agent. Furthermore, the call agent can audit endpoints as well as the connections on endpoints.

MGCP assumes a call control architecture where the call control intelligence is outside the gateways and handled by call agents. It assumes that call agents will synchronize with each other to send coherent commands and responses to the gateways under their control.

MGCP does not define a mechanism for synchronizing call agents. MGCP assumes a connection model where the basic constructs are endpoints and connections. Creation of physical endpoints requires hardware installation, while creation of virtual endpoints can be done by software. Connections can be either point to point or multipoint. A point-to-point connection is an association between two endpoints with the purpose of transmitting data between those endpoints.

When that association is established for both endpoints, data transfer between them can take place. A multipoint connection is created by connecting the endpoint to a multipoint session.

Connections can be established over several types of bearer networks. In the MGCP model, the gateways focus on the audio signal translation function, and the call agent handles the call-signaling and call-processing functions. SIP, which is standardized under RFC , is a peer-to-peer protocol in which end devices, known as user agents, initiate sessions. SIP is designed in conformance with the Internet model. It is an end-to-end signaling protocol, which means that all the logic is stored in end devices, except the routing of SIP messages.

State is also stored in end devices only. There is no single point of failure with SIP, and networks designed this way scale well. The tradeoff for the distributiveness in scalability is the higher message overhead that results from the messages being sent end to end.

The aim of SIP is to provide the same functionality as the traditional PSTN, but with an end-to-end design that makes SIP networks much more powerful and open to the implementation of new services. SIP is an application-layer control protocol that can establish, modify, and terminate multimedia sessions.

Examples of a session include Internet telephone calls, distribution of multimedia, multimedia conferences, and distribution of computer games. SIP can also invite participants to already existing sessions, such as a multicast conference. Media can be added to and removed from an existing session. SIP transparently supports name-mapping and redirection services, which make personal mobility possible. Users can maintain a single externally visible identifier, regardless of their network locations.

SIP support five facets of establishing and terminating multimedia communications: user location the determination of the end system to be used for communication , user availability the determination of the willingness of the called party to engage in communications , user capabilities the determination of the media and media parameters to be used , session setup the establishment of session parameters at both the called and calling parties , and session management the transfer and termination of sessions, modification of session parameters, and invocation of services.

Although SIP should be used in conjunction with other protocols to provide complete services to users, the basic functionality and operation of SIP do not depend on any other protocols. The most common SIP operation is the invitation. Instead of directly reaching the intended callee, a SIP request may be redirected or may trigger a chain of new SIP requests by proxies.

Users can register their locations with SIP servers; SIP addresses can be embedded in Web pages and, therefore, can be integrated as part of powerful applications such as click-to-talk. The purpose of SIP is just to make the communication possible see Figure 9. The communication itself must be achieved by another means and possibly another protocol. RTP is used to carry the real-time multimedia data, including audio, video, and text.

RTP makes it possible to encode and split the data into packets and transport the packets over the Internet. SDP is used to describe and encode the capabilities of session participants. This description is then used to negotiate the characteristics of the session so that all the devices can participate.

For example, the description is used in negotiation of the codecs used to encode media so all the participants will be able to decode it and in negotiation of the transport protocol to be used. Basic SIP elements include user agents, proxies, registrars, and redirect servers see Figure 9. User agents usually but not necessarily reside on a user's computer in the form of an application, but user agents can also be cellular phones, PSTN gateways, PDAs, automated integrated voice response systems, and so on.

Each user agent actually contains both. The UAC is the part of the user agent that sends requests and receives responses; the UAS is the part of the user agent that receives requests and sends responses. A proxy server is an optional SIP component.

User agents can send messages to a proxy server. The most important task of a proxy server is to route session invitations closer to the callee. The session invitation usually traverses a set of proxies until it finds one that knows the actual location of the callee. Such a proxy then forwards the session invitation directly to the callee, and the callee then accepts or declines the session invitation. A registrar server is another optional SIP component.

It receives registrations from users, extracts information about their current locations such as IP address, port, and user name , and then stores the information in local databases. A registrar is very often a logical entity only and is usually collocated with proxy servers. A redirect server is yet another optional SIP component.

A redirect server receives requests and looks up the intended recipient of a request in the location database created by a registrar. It then creates a list of current locations of the user and sends this list to the request originator. The redirect server does not route SIP messages; rather, it returns a redirect to the user agent for direct routing. Presence is defined as the ability, willingness, desire, and capability of a user to communicate across media end devices and even time and space.

Presence systems collect and distribute presence information to interested parties such as other users or applications. Policythat is, who is allowed to see what and whenis central to presence. The value of presence is based on the richness of the data it has access to. Presence is the ability to see in real-time where someone is, how that person prefers to be reached, and even what the person is doing.

Today, only one in seven business calls is completed successfully. Instead, we get busy signals or are routed to voicemail. Users spend a lot of time trying many different numbers for the same person and calling many numbers in succession, trying to reach someone. Presence, therefore, has great value in improving the productivity of an organization because it can see ahead of time whether a call will succeed.

It can tell the caller which is the best way to reach a user, and it can allow a caller to quickly see who among a set of candidates is available for a call.

Accuracy is paramount to presence. The greater the accuracy, the greater the value of presence. The worse the accuracy, the lower the value. Productivity enhancement depends on accurate presence data, and accuracy is achieved by combining multiple sources of presence data. There are many sources for presence data within an enterprise, such as detecting whether somebody is on a call or off a call, based on information from enterprise phones; determining whether a person is in a meeting or out of a meeting, based on information from the enterprise calendar systems; determining whether a person is in a conference or out of a conference, based on enterprise conferencing systems; or determining whether a person has logged in or logged out of the enterprise IM system.

They can also support presence-based call routing as well as voicemail ringback.