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Voice over Internet Protocol – Definition and Overview

Voice over Internet Protocol, or VoIP for short, is slowly becoming a buzz word in the home as well as in the workplace. Many people are asking about this new “free” telephony system over the Internet. VoIP is a telephony technology and a new set of IP-based applications that allows users to make calls using their broadband Internet connection. This application can be used when making calls to any phone anywhere in the globe, regardless whether the person at the other end has VoIP or not.

Calls can be made by using either your PC with broadband Internet connection or you standard phone that is connected to the Internet through a special VoIP adapter. There are now new generations of “VoIP-enabled” telephones that can be directly connected to the Internet router or cable modem. VoIP has become a very popular communication alternative for both business and personal use over the last few years.

We are now seeing that it is already beginning to replace traditional phone lines that operate on the telephony system adopted by mainstream service providers. The main appeal of this IP-based telephony system is its significant cost-saving feature. Overseas and long distance calls come out significantly cheaper. The providers of VoIP phone services are not weighed down by cost and other intrinsic limitations to maintain exiting phone networks. This easily translates to a type of phone service that is significantly cheaper.

VoIP retains the features that are currently provided by traditional phone systems. These include call forwarding, caller ID, call waiting, call blocking and voicemail. It also comes with value-added features that are not available in traditional phone systems such as the ability to set up a virtual contact number – a phone number which can be accessed from any area code that is currently available. This means that you can receive calls from other people even if they are outside of your calling area without paying the rate charged for long distance calls.

Common Channeling Signaling System No. 7 (SS7) Overview

Common Channel Signaling System No. 7, or simply SS7, is a global telecommunication standard whose procedures and protocols are defined by the Telecommunication Standardization Sector (ITU-T) by the International Telecommunication Union (ITU). This ITU-I standard defines the protocol and procedures through which elements of the communication network in the PSTN exchange information over a digital-based signaling communication platform are used to effect land-based and wireless configuration, routing and control. This ITU SS7 definition serves as the basis for national variants like ANSI and Telcordia Technologies standards, which are currently used in North America, and ETSI standard, which is currently used in Europe.

SS7 provides for a distinct packet switching network for the relay of call control messages through telecommunications networks. There are two main distinctions between the classical packet switched networks and the specialized packet networks. These are higher level of performance and reliability in handling large volume of traffic. The SS7 protocol delivers a high level of performance and reliability while retaining its robustness even under changing conditions and systems failures.

The SS7 protocol and network are used for:

SS7 messages are shared among network elements over 56 or 64 kbps bi-directional communication channels known as signaling links. The exchange of messages happens along out-of-band on dedicated channels instead of in-band signaling on voice channels.

This signaling setup results to:

A numeric point code is used to identify SS7 network’s signaling point. These codes are present in the signaling messages that are shared between the signaling points for the purpose of message source and destination identification. Each of the signaling points relies on a routing table for the selection of the appropriate message signaling path.

There are three kinds of SS7 network’s signaling points. These are:

Signaling Gateways

There are three essential components of an IP-SS7 converged communication network. These are the service switching point (SSP), signal transfer point (STP) and signaling end point (SEP). The SSP, which is equipped with SS7 application, is considered as an “end-node” or a telephone switch, that relays the signaling message to other STPs or SSPs to set up, handle and relay voice circuits that are needed for the completion of the call. The signal transfer point is considered as a signal packet router or switch that receives signaling messages and routes them to their ultimate destinations. SEP is the generic designation to an SS7-defined end node that has similar functionality as the service switching point in an IP-based telephony network.

The SSP is a conventional voice switch that offers call control, trunk signaling and switching along a large single entry. These defined functions under an IP-based phone communication configuration are provided by separate entities. These distributed entities include a media gateway or softswitch controller, a signaling gateway and a media gateway. The virtual switch becomes an SEP with the combination of these functions under the IP setup. Both SEPs and SSps are linked to the STPs using the “A” links and are considered as single point codes under the SS7 communication network.

The STP and SEP assume unique functions within the network. An STP-configured signaling gateway provides concentrating and routing capabilities. This type of signaling gateway may also provide SS7 traffic concentration and global title translation. In general terms, when a signaling gateway functions as an STP in a converged communication network, it leads to the connection to other STPs within the public-switched telephone network using “B” or “D” links. This setup also provides flexibility of routing to various public-switched-telephone network locations such as Line Information Database, 800 or local number portability database. STP functionality is needed when customers require point codes and routing to IP-based system application.