Voice Over IP

• About VoIP (Voice over IP)

• Speech Quality

• VoIP Client - SJphone

• Call Termination (Calls to PSTN)

• SJgatekeeper

About VoIP (Voice over IP)

Quick Concept:

How traditional PSTN (Public Switch Telephone Network) works:
You pick up your phone and dial a phone number, the call goes through your local telephone company to your long distance provider.

How long distance works with VoIP:
You pick up your phone and dial a long distance phone number, the call goes through your local telephone company to a VoIP provider, this is a local call. The call then goes over the Internet to the receiver's local calling area were a local call is placed (by the VoIP provider) to complete the connection.

Overview:
The possibility of voice communications traveling over the Internet, rather than the PSTN, first became a reality in February 1995 when Vocaltec, Inc. introduced its Internet Phone software. Designed to run on a 486/33-MHz (or higher) (PC) equipped with a sound card, speakers, microphone, and modem, the software compressed the voice signal and translated it into IP packets for transmission over the Internet. This PC-to-PC Internet telephony worked, however, only if both parties were using Internet Phone software.

In the relatively short period of time since then, Internet telephony has advanced rapidly. Many software developers now offer PC telephony software but, more importantly, gateway servers are emerging to act as an interface between the Internet and the PSTN. Equipped with voice-processing cards, these gateway servers enable users to communicate via standard telephones over great distances without going over the telephone network.

A call goes over the local PSTN to the nearest gateway server, which digitizes the analog voice signal, compresses it into IP packets, and moves it onto the Internet for transport to a gateway server at the receiving end. This server converts the digital IP signal back to analog and completes the call locally. With its support for computer-to-telephone calls, telephone-to-computer calls and telephone-to-telephone calls, VoIP represents a significant step toward the integration of voice and data networks.

VoIP could be applied to almost any voice communications requirement, ranging from a simple inter-office intercom to complex multi-point teleconferencing/shared screen environments.

Widespread deployment of a new technology seldom occurs without a clear and sustainable justification, and this is also the case with VoIP. Demonstrable benefits to end-users are also needed if VoIP products (and services) are to be a long-term success. Generally, the benefits of technology can be divided into the following four categories:

• Cost Reduction. The sharing of equipment and operations costs across both data and voice users can also improve network efficiency since excess bandwidth on one network can be used by the other, thereby creating economies of scale for voice (especially given the rapid growth in data traffic).

• Simplification. An integrated infrastructure that supports all forms of communication allows more standardization and reduces the total equipment complement. This combined infrastructure can support dynamic bandwidth optimization and a fault tolerant design. The differences between the traffic patterns of voice and data offer further opportunities for significant efficiency improvements.

• Consolidation. Since people are the most significant cost elements in a network, any opportunity to combine operations, to eliminate points of failure, and to consolidate accounting systems would be beneficial. In the enterprise, SNMP (Simple Network Management Protocol)-based management can be provided for both voice and data services using VoIP. Universal use of the IP protocols for all applications holds out the promise of both reduced complexity and more flexibility. Related facilities such as directory services and security services may be more easily shared.

• Advanced Applications. Even though basic telephony and facsimile are the initial applications for VoIP, the longer term benefits are expected to be derived from multimedia and multi-service applications. For example, Internet commerce solutions can combine WWW access to information with a voice call button that allows immediate access to a call center agent from the PC. Needless to say, voice is an integral part of conferencing systems that may also include shared screens, whiteboarding, etc. Combining voice and data features into new applications will provide the greatest returns over the longer term.

Although the use of voice over packet networks is relatively limited at present, there is considerable user interest and trials are beginning. End user demand is expected to grow rapidly over the next five years. Frost & Sullivan and other research firms have estimated that the compound annual growth rate for IP-enabled telephone equipment will be 132% over the period from 1997 to 2002 (from $47.3M in 1997 to $3.16B by 2002). It is expected that VoIP will be deployed by 70% of the Fortune 1000 companies by the year 2002. Industry analysts have also estimated that the annual revenues for the IP fax gateway market will increase from less than $20M in 1996 to over $100M by the year 2002. It is clear that a market has already been established and there exists a window of opportunity for developers to bring their products to market, and for consumers to realize significant savings. 

Speech Quality

Speech quality, especially in Voice over Internet Protocol (VoIP) systems, is influenced by many parameters like e.g.

• delay, mainly determining the conversational quality and the echo perception for both subjects

• the quality of implemented echo cancellers

• speech quality in the listening situation influenced by different implemented codecs or algorithms tuned for packet loss concealment (PLC)

• voice switching problems due to the occurrence of packet loss in the network or voice activity detection (VAD)

• the transmission of background noise, which is a very critical parameter for the naturalness of the conversation

VoIP Client - SJphone

Click here to download SJphone VoIP software.

SJphone™ is Windows CE and Windows 98/NT/2000/XP software H.323/SIP Terminal, which provides two-way voice communications with any other H.323/SIP Terminal or Gateway via Internet or Wireless/Wired IP network. SJphone™ users can place a call over the Internet to/from PSTN via Gateway. It allows the PC, PDA and Web Pad users to communicate to both: traditional and IP telephones. The list of SJphone™ features includes:

• Dual Standard: H.323 and SIP

• Multi-session Support

• Teleconferencing

• WEB Interface (Client can be activated from WEB)

• Advanced Calling:
  Registration with Gatekeeper or with SIP Proxy
  Various compression algorithms support: G.711, G.723, G.729

• runs on PDA with Win CE or PC with Win ME/2K/XP

Call Termination (Calls to PSTN)

SJphone obviously allows making PC-to-PC calls over the Internet or any IP network.
In order to make calls from your device to a telephone you need to have a H.323 or SIP gateway. Some corporations own ones or just outsource access to gateway network. There is also an economic solution for SOHO (Small Office Home Office) users. It usually includes 1 or 2 port gateway, SJphone and (optional) SJgatekeeper™. We carry several lines and types of gateways – please refer to our site. If you need more information about this product please email us.

SJgatekeeper™

An intelligent way to support the call management of VoIP network

Click here to request an evaluation version of SJgatekeeper^TM

SJgatekeeper™ gives enterprise and service providers a large capacity H.323 compliant gatekeeper for VoIP networks. SJgatekeeper™ provides the call routing and administration for all the H.323 VoIP endpoints in your network, including other SJgatekeepers, gateways, IP phones and PC or Win CE clients.

SJgatekeeper™ provides the unique ability to dynamically self configure the VoIP routing database. It will also support static routing to provide connectivity to endpoints that are not part of your VoIP network. The advanced call routing enables SJgatekeeper™ to provide load balancing among multiple gateways and gatekeepers. SJgatekeeper™ also provides for authentication/authorization of all calls on VoIP network.

Unique call processing intelligence
With its unique call processing intelligence which includes H.245, H.225 and media streams support SJgatekeeper™ makes deployment of any size VoIP network easy, stable and risk-free. Besides, compliance with RADIUS allows to implement wide variety of billing systems in our VoIP network

High level of intelligence provides easier network configuration.
SJgatekeeper™ dynamically self configures your VoIP network by collecting routing information during endpoint registration. The routing database is automatically constructed and shared among all the gatekeepers in the VoIP network. New endpoints are automatically added and configured into the network.

More intelligence means greater security
SJgatekeeper™ provides authorization and authentication for all calls on your VoIP network. This feature provides protection against unauthorized calls and insures proper endpoint identification.

Remote management
SJgatekeeper™ can be ran as a daemon in UNIX, as a service in Windows NT/2000/XP and as an application in Windows 98/SE/ME. The remote console installed on local PC provides for full manageability of SJgatekeeper™ which makes support of remote gatekeepers on your VoIP network easy and reliable.