Voice over Internet Protocol (VoIP) is a protocol optimized for the transmission of voice through the Internet or other packet switched networks. VoIP is often used abstractly to refer to the actual transmission of voice (rather than the protocol implementing it). VoIP is also known as IP Telephony, Internet telephony, Broadband telephony, Broadband Phone and Voice over Broadband. “VoIP” is pronounced voyp.
Companies providing VoIP service are commonly referred to as providers, and protocols which are used to carry voice signals over the IP network are commonly referred to as Voice over IP or VoIP protocols. They may be viewed as
commercial realizations of the experimental Network Voice Protocol (1973) invented for the ARPANET providers. Some cost savings are due to utilizing a single network to carry voice and data, especially where users have existing underutilised network capacity that can carry VoIP at no additional cost. VoIP to VoIP phone calls are sometimes free, while VoIP to public switched telephone networks, PSTN, may have a cost that is borne by the VoIP user.
Voice over IP protocols carry telephony signals as digital audio, typically reduced in data rate using speech data compression techniques, encapsulated in a data packet stream over IP.
There are two types of PSTN to VoIP services: Direct Inward Dialing (DID) and access numbers. DID will connect the caller directly to the VoIP user while access numbers require the caller to input the extension number of the VoIP user.
Voice over Internet Protocol has been a subject of interest almost since the first computer network. By 1973, voice was being transmitted over the early Internet. Technology for transmitting voice conversations over the internet has been available to end users since at least the 1990’s. For instance, in 1996, a shrink-wrapped software product called Vocaltec Internet Phone Release 4 provided VoIP, along with extra features such as voice mail and caller id. However, it did not offer a gateway to the analog POTS, so it was only possible to speak to other Vocaltec Internet Phone users. In 1997, Level 3 began development of its first softswitch (a term they invented in 1998); softswitches were designed to replace a traditional hardware switchboards by serving as the gateway between two telephone networks.
VoIP can facilitate tasks and provide services that may be more difficult to implement or expensive using the more traditional PSTN. Examples include:
- The ability to transmit more than one telephone call down the same broadband-connected telephone line. This can make VoIP a simple way to add an extra telephone line to a home or office.
- 3-way calling, call forwarding, automatic redial, and caller ID; features that traditional telecommunication companies (telcos) normally charge extra for.
- Secure calls using standardized protocols (such as Secure Real-time Transport Protocol.) Most of the difficulties of creating a secure phone over traditional phone lines, like digitizing and digital transmission are already in place with VoIP. It is only necessary to encrypt and authenticate the existing data stream.
- Location independence. Only an internet connection is needed to get a connection to a VoIP provider. For instance, call center agents using VoIP phones can work from anywhere with a sufficiently fast and stable Internet connection.
- Integration with other services available over the Internet, including video conversation, message or data file exchange in parallel with the conversation, audio conferencing, managing address books, and passing information about whether others (e.g. friends or colleagues) are available online to interested parties.
Mobile phones & Hand held Devices
Telcos and consumers have invested billions of dollars in mobile phone equipment. In developed countries, mobile phones have achieved nearly complete market penetration, and many people are giving up landlines and using mobiles exclusively. Given this situation, it is not entirely clear whether there would be a significant higher demand for VoIP among consumers until either public or community wireless networks have similar geographical coverage to cellular networks (thereby enabling mobile VoIP phones, so called WiFi phones or VoWLAN) or VoIP is implemented over 3G networks. However, “dual mode” telephone sets, which allow for the seamless handover between a cellular network and a WiFi network, are expected to help VoIP become more popular.
Phones like the NEC N900iL, and later many of the Nokia Eseries and several WiFi enabled mobile phones have SIP clients hardcoded into the firmware. Such clients operate independently of the mobile phone network unless a network operator decides to remove the client in the firmware of a heavily branded handset. Some operators such as Vodafone actively try to block VoIP traffic from their network and therefore most VoIP calls from such devices are done over WiFi.
Several WiFi only IP hardphones exist, most of them supporting either Skype or the SIP protocol. These phones are intended as a replacement for PSTN based cordless phones but can be used anywhere where WiFi internet access is available.
Another addition to hand held devices are ruggedized bar code type devices that are used in warehouses and retail environments. These type of devices rely on “inside the 4 walls” type of VoIP services that do not connect to the outside world and are solely to be used from employee to employee communications.
The two major competing standards for VoIP are the ITU standard H.323 and the IETF standard SIP. Initially H.323 was the most popular protocol, though in the “local loop” it has since been surpassed by SIP. This was primarily due to the latter’s better traversal of NAT and firewalls, although recent changes introduced for H.323 have removed this advantage.
However, in backbone voice networks where everything is under the control of the network operator or telco, H.323 is the protocol of choice. Many of the largest carriers use H.323 in their core backbones, and the vast majority of callers have little or no idea that their POTS calls are being carried over VoIP.
Where VoIP travels through multiple providers’ softswitches the concepts of Full Media Proxy and Signalling Proxy are important. In H.323, the data is made up of 3 streams of data: 1) H.225.0 Call Signaling; 2) H.245; 3) Media. So if you are in London, your provider is in Australia, and you wish to call America, then in full proxy mode all three streams will go half way around the world and the delay (up to 500-600 ms) and packet loss will be high. However in signaling proxy mode where only the signaling flows through the provider the delay will be reduced to a more user friendly 120-150 ms.
One of the key issues with all traditional VoIP protocols is the wasted bandwidth used for packet headers. Typically, to send a G.723.1 5.6 kbit/s compressed audio path requires 18 kbit/s of bandwidth based on standard sampling rates. The difference between the 5.6 kbit/s and 18 kbit/s is packet headers. There are a number of bandwidth optimization techniques used, such as silence suppression and header compression. This can typically save 35% on bandwidth usage.
VoIP trunking techniques such as TDMoIP can reduce bandwidth overhead even further by multiplexing multiple conversations that are heading to the same destination and wrapping them up inside the same packets. Because the packet header overhead is shared between many simultaneous streams, TDMoIP can offer near toll quality audio with a per-stream packet header overhead of only about 1 kbit/s.