Voice over Internet Protocol, also called IP telephony, is a method and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol networks, such as the Internet.
As with any other SaaS (Software as a Service) solution, ‘Hosted’ or ‘Cloud’ VoIP solutions involve a service provider or telecommunications carrier hosting the prerequisite telephone system as a software solution within their own infrastructure.
Typically this will be one or more datacenters, with geographic relevance to the end user(s) of the system. This infrastructure is external to the user of the system, and is deployed and maintained by the service provider.
Endpoints, such as VoIP telephones or softphone applications (apps running on a computer or mobile device), will therefore connect to the VoIP service remotely. These connections typically take place over public internet links, such as local fixed WAN breakout or mobile carrier service (such as 4G).
Because of the bandwidth efficiency and low costs that VoIP technology can provide, businesses are migrating from traditional copper-wire telephone systems to VoIP systems to reduce their monthly phone costs. In 2008, 80% of all new Private branch exchange (PBX) lines installed internationally were VoIP. For example, in the United States, the Social Security Administration is converting its field offices of 63,000 workers from traditional phone installations to a VoIP infrastructure carried over its existing data network.
VoIP allows both voice and data communications to be run over a single network, which can significantly reduce infrastructure costs. The prices of extensions on VoIP are lower than for PBX and key systems. VoIP switches may run on commodity hardware, such as personal computers. Rather than closed architectures, these devices rely on standard interfaces. VoIP devices have simple, intuitive user interfaces, so users can often make simple system configuration changes. Dual-mode phones enable users to continue their conversations as they move between an outside cellular service and an internal Wi-Fi network, so that it is no longer necessary to carry both a desktop phone and a cell phone. Maintenance becomes simpler as there are fewer devices to oversee.
VoIP solutions aimed at businesses have evolved into unified communications services that treat all communications—phone calls, faxes, voice mail, e-mail, web conferences, and more—as discrete units that can all be delivered via any means and to any handset, including cellphones. Two kinds of service providers are operating in this space: one set is focused on VoIP for medium to large enterprises, while another is targeting the small-to-medium business (SMB) market.
Skype, which originally marketed itself as a service among friends, has begun to cater to businesses, providing free-of-charge connections between any users on the Skype network and connecting to and from ordinary PSTN telephones for a charge
In general, the provision of VoIP telephony systems to organizational or individual users can be divided into two primary delivery methods: private or on-premises solutions, or externally hosted solutions delivered by third-party providers. On-premises delivery methods are more akin to the classic PBX deployment model for connecting an office to local PSTN networks.
While many use cases still remain for private or on-premises VoIP systems, the wider market has been gradually shifting toward ‘Cloud’ or Hosted’ VoIP solutions. Hosted systems are also generally better suited to smaller or personal use VoIP deployments, where a private system may not be viable for these scenarios.
Several protocols are used in the data link layer and physical layer for quality-of-service mechanisms that help VoIP applications work well even in the presence of network congestion. Some examples include:
- IEEE 802.11e is an approved amendment to the IEEE 802.11 standard that defines a set of quality-of-service enhancements for wireless LAN applications through modifications to the Media Access Control (MAC) layer. The standard is considered of critical importance for delay-sensitive applications, such as voice over wireless IP.
- IEEE 802.1p defines 8 different classes of service (including one dedicated to voice) for traffic on layer-2 wired Ethernet.
- The ITU-T G.hn standard, which provides a way to create a high-speed (up to 1 gigabit per second) Local area network (LAN) using existing home wiring (power lines, phone lines and coaxial cables). G.hn provides QoS by means of Contention-Free Transmission Opportunities (CFTXOPs) which are allocated to flows (such as a VoIP call) which require QoS and which have negotiated a contract with the network controllers.
The quality of voice transmission is characterized by several metrics that may be monitored by network elements and by the user agent hardware or software. Such metrics include network packet loss, packet jitter, packet latency (delay), post-dial delay, and echo. The metrics are determined by VoIP performance testing and monitoring.