Voice over IP

Voice over IP (VoIP), also called Internet telephony or IP telephony, is the transmission of voice telephony services over IP, the Internet Protocol.

Its advantages over traditional telephony include:

• lower costs per call, especially for long-distance calls
• lower infrastructure costs: once IP infrastructure is installed, no or little additional telephony infrastructure is needed.

Note that voice over IP traffic does not necessarily have to travel over the public Internet: it may also be deployed on private IP networks.

The benefit of using this technology is the need for only one class of circuit connection and better use of the available bandwidth. IP telephony is commonly used to route traffic that may be originated from and terminated at conventional PSTN telephones.

VoIP is now widely deployed by carriers, especially for international telephone calls. Most commonly, users are completely unaware that their telephone call is being routed over IP infrastructure for most of its distance, instead of entirely over the circuit switched PSTN.

VoIP is also used by large companies to eliminate call charges between their offices, by using their data network to carry inter-office calls. They may also use VoIP to reduce the costs of calls outside the company, by carrying them to the nearest point on their network before handing them off to the PSTN.

Because IP does not by default provide any mechanism to ensure that data packets are delivered in sequential order, or provide any Quality of Service guarantees, implementations of VoIP face problems dealing with latency and possible data integrity problems.

One of the central challenges for VoIP implementers is restructuring streams of received IP packets, which can come in any order and have packets missing, to ensure that the ensuing audio stream maintains a proper time consistency. Another important challenge is keeping packet latency down to acceptable levels, so that users do not experience significant lag time between when they speak and the signal is decoded on the other end of the connection.

Solutions to these problems:

• Certain hardware solutions can distinguish VoIP packets and provide priority.
• Alternately packets can be buffered but this can lead to an overall delay similar to that encountered on satellite circuits.
• The network operator can also ensure that there is enough bandwidth end-to-end to guarantee low-latency low-loss traffic: this is easy to do in private networks, but much harder to do in the public Internet.

In the overwhelming majority of implementations, the RTP protocol is used to transmit VoIP traffic ("media").

For signaling, there are several alternative protocols:

• H.323
• Skinny Client Control Protocol, proprietary protocol from Cisco
• SIP, the IETF Session Initiation Protocol
• MeGaCo[?] (a.k.a. H.248) and MGCP, both Media Gateway[?] control protocols

See also: IntServ, DiffServ , computer conferencing