Voice over IPVoice 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.
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.
There are companies which offer a gateway to the PSTN from any VoIP phone. You can simply dial a conventional telephone number and the telephone call will be routed over your internet connection to the company that operates the gateway, and they will bill you, not the local phone company. Enum makes it possible to dial traditional E.164 phone numbers, but be connected entirely over the internet if the other party uses Enum, so you don't incur any expenses other than the internet connection fees.
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.
For signaling, there are several alternative protocols:
- SIP, the IETF Session Initiation Protocol, a newcomer gaining popularity
- H.323, an older protocol still used by many legacy applications
- Skinny Client Control Protocol, proprietary protocol from Cisco
- MeGaCo (a.k.a. H.248) and MGCP, both Media Gateway control protocols