The purpose and function of a VoIP Gateway is to provide an interface between the traditional telephone networks using digital TDM (Time Division Multiplexing) technology and that of IP Networks designed to carry IP packets containing digital speech. The VoIP Gateway has to translate the digital media format used on the local network and the PSTN (Public Switched Telephone Network) in both directions. In addition, the Gateway will also have to translate between the different signalling protocols used on the local network and PSTN.

 Today’s Voice over IP networks digitise analogue speech for telephone calls my means of a Codec which stands for Coder Decoder or Compressor De-compressor. There are a number of codecs used within VoIP systems, with the most common of these being ITU-T codecs G.711 and G.729, and the GSM codecs used within the mobile telephone networks. Two versions of the g.711 codec are used within the traditional digital telephone networks which are based largely on 64Kbps digital telephone channels. The digital telephone channels are multiplexed together using a method referred to as Time Division Multiplexing to form either T1 or E1 lines. A T1 line is designed to carry 24 digital telephone calls, whereas an E1 line is designed to carry 30 digital telephone calls. These multiplexed trunk lines are used to interconnect exchanges and also connect a traditional PBX (Private Branch Exchange) to a Telephone Operator’s exchange.

A VoIP Gateway will need to support a number of codecs, so as to be able to translate between the digital codec formats used by the PSTN and the VoIP enabled Local Area Network.

 In addition to translating digital voice formats, the gateway must also be able to translate between different signalling formats. The Public Switched Telephone Network uses two main forms of signalling which are referred to as CAS (Channel Associated Signalling) and CCS (Common Channel Signalling). The former is used on the analogue telephone line between the telephone user and the exchange by passing telephone dialling codes often by means of audio tones, and this method is known as DTMF (Dual Tone Multi Frequency). Pulse dialling is also used in some systems. Common Channel Signalling is used between exchanges and gateways to pass telephone signalling information within the wider telephone network.

 SIP (Session Information Protocol) has evolved as the dominant signalling protocol within VoIP Systems, and has largely replaced the cumbersome ITU signalling protocol H.323. SIP is a simple client / server protocol that is used to setup, maintain and teardown VoIP telephone calls. MGCP (Media Gateway Control Protocol) is used by some Voice over IP systems to control the actions of a VoIP Gateway and Cisco uses SCCP (Skinny Call Control Protocol) on some older systems as a means of communicating between the end users and the Cisco Callmanager call control agent.

 Signalling System 7 is the CCS system used within all the Public Telephone Networks, including land-based and mobile systems. In fact, it is sometimes referred to as the “Glue” that links all Public networks together and allows us to make calls between Mobile phones, land-based phones and VoIP phones.

 Other main functions and features of VoIP Gateways include:

 Fax as well as voice compression and decompression

• Packetising digital voice information

• Control signalling

• Call Routing

• Interface to other external devices such as SoftSwitches and H.323 Gatekeepers

Gateways supplied by different vendors are often packed with proprietary features as well as the standard features expected of a VoIP Gateway. Selecting the right one for your organisation or business requires careful thought and some research.

 A good small business VoIP Gateway is Cisco’s SPA3102 which provides a single FXS (Foreign Exchange Service) RJ-11 connector, enabling the user to connect a single standard telephone or FAX machine on the local network side, and a single FXO (Foreign Exchange Office) port to enable connection to the local Telco. Two additional Fast Ethernet ports provide connection to the home or office Local Area Network. An additional Fast Ethernet port provides a connection to a broadband modem or router.

 For additional small business capability, Cisco’s VGA204 Analogue Voice Gateway provides 4 x FXS RJ-11 ports to allow connectivity for 4 telephones or fax machines and 2 Fast Ethernet network ports. It supports all the major network signalling standards such as SIP, SCCP, MGCP and H.323v4.

 VoIP Gateways are continuing to evolve as enhancements to existing protocols continue and new VoIP protocols are being developed. All the major Gateway vendors are trying to get a larger market share and so proprietary features are something to look out for, particularly in the small business market. Check out your options before jumping in and buying the first Gateway and be sure of your ultimate requirements.

 This article on VoIP Gateways was written by David Christie, MD at NSTUK Ltd,  Website http://www.ipexpress.co.uk .