VoIP (voice over IP - that is, voice delivered using the Internet Protocol) is a term used in IP telephony for a set of facilities for managing the delivery of voice information using the Internet Protocol (IP). In general, this means sending voice information in digital form in discrete packets rather than in the traditional circuit-committed protocols of the public switched telephone network (PSTN). A major advantage of VoIP and Internet telephony is that it avoids the tolls charged by ordinary telephone service.
VoIP, now used somewhat generally, derives from the VoIP Forum, an effort by major equipment providers, including Cisco, VocalTec, 3Com, and Netspeak to promote the use of ITU-T H.323, the standard for sending voice (audio) and video using IP
on the public Internet and within an intranet. The Forum also promotes the user of directory service standards so that users can locate other users and the use of touch-tone signals for automatic call distribution and voice mail.
In addition to IP, VoIP uses the real-time protocol (RTP) to help ensure that packets get delivered in a timely way. Using public networks, it is currently difficult to guarantee Quality of Service (QoS). Better service is possible with private networks managed by an enterprise or by an Internet telephony service provider (ITSP).
A technique used by at least one equipment manufacturer, Adir Technologies (formerly Netspeak), to help ensure faster packet delivery is to use ping to contact all possible network gateway computers that have access to the public network and choose the fastest path before establishing a Transmission Control Protocol (TCP) sockets connection with the other end.
Using VoIP, an enterprise positions a "VoIP device" at a gateway. The gateway receives packetized voice transmissions from users within the company and then routes them to other parts of its intranet (local area or wide area network) or, using a T-carrier system or E-carrier interface, sends them over the public switched telephone network.
VoIP phones are telephones which connect to VoIP networks instead of to the PSTN.
- VoIP phones with Ethernet connections
- VoIP phones with Wi-Fi / 802.11 connections
- VoIP phones with dialup modem connections
- Software VoIP phones
VoIP phones with Ethernet connections
A VoIP phone with an Ethernet connection is the easiest type of VoIP telephone to use. Instead of a standard telephone RJ-11 connector to plug into the PSTN, these phones have RJ-45 connectors to plug into Ethernet networks.
The Ethernet connection is used to connect these VoIP phones to the VoIP server or VoIP gateway.
VoIP phones with Wi-Fi / 802.11 connections
Wi-Fi (802.11) VoIP phones provide the same service as Ethernet VoIP phones, but they do it wirelessly.
A Wi-Fi enabled VoIP phone connects to a VoIP server or VoIP gateway through your existing Wi-Fi network.
VoIP phones with dialup modem connections
VoIP phones with dialup modem connections are very similar to VoIP phones with Ethernet connections.
Instead of connecting to an Ethernet network, these VoIP phones dialup over the PSTN to VoIP service providers. Using a VoIP phone with a dialup modem connection requires a regular analog POTS telephone line, but enables long-distance and international calls to be made over VoIP networks, usually at a significant savings.
Software VoIP phones
Software VoIP phones turn your PC into a VoIP telephone.
Software VoIP telephones are less expensive than the choices listed above, if you already own a personal computer.
Hardware for Software VoIP phones
Software VoIP phones use the PC's sound card, speakers or earphones, and microphone. This hardware works to emulate a telephone, even though this is not what the PC was designed for.
For better ease-of-use, many companies manufacture USB VoIP phones. These devices give your PC a normal-looking telephone handset or headset.
Free VOIP gateways
SIPRG (SIP Residential Gateway): The SIP Residential Gateway (SIPRG) is an open source application based on the Session Initiation Protocol (SIP). The SIPRG is an IP Telephony Gateway that allows a SIP User Agent to make and receive calls between the Public Switched Telephone Network (PSTN) and a SIP-based network such as VOCAL.
The SIPRG was developed with the VOVIDA SIP stack version 1.3.0, and uses a QuickNet LineJACK card for connecting an Analog telephone line. Currently, it supports only a single LineJACK card and is therefore a single-line gateway.
isdnh323: isdn2h323 is a Linux based H.323 - ISDN gateway. At the moment the gateway supports the following features.
- ISDN and H.323 users can initiate a connection
- The number of simultaneous incoming and outgoing calls is limited by the number of available ISDN channels only.
- H.323 users can specify the ISDN number of the other party.
- The gateway's administrator can assign an ISDN MSN to a H.323 user. This makes it possible for an ISDN user to call a H.323 user directly. The gateway will choose the H.323 user id depending on the called ISDN MSN.
- The gateway discovers an available H.323 gatekeeper and registers with the gatekeeper. It's possible to specify one or more phone prefixes the gateway is responsible for.
- ISDN's touch-tones (DTMF) are translated to H.323's user input messages and vice versa.
- Automatic gain control (AGC)
- Automatic echo compensation (AEC)
- To avoid security problems the gateway offers an option to restrict the IPs allowed to use the gateway for an outgoing ISDN call.
- The status of the lines and the configuration of the gateway are written to a HTML file.
- Errors and other information are logged using Linux's syslog() feature.
- Three H.323 codecs are supported: ALaw, muLaw, and GSM.
- Least Cost Router
PSTNGw: PSTNGw is a very simple PSTN to H.323 gateway program using the OpenH323 library. It allows H.323 clients to make outgoing calls, and incoming calls to be routed to a specific H.323 client.
PSTNGw makes use of PWLib and the OpenH323 stack from Equivalence Ltd Pty.