Voice over IP

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 protocols used to carry the signal over the IP network are commonly referred to as Voice over IP, or VoIP protocols. VoIP protocols include the heavyweight H.323, which also provides videoconferencing and data capability, MGCP (Media Gateway Control Protocol), SGCP (Simple Gateway Control Protocol), and SIP (Session Initiation Protocol), a protocol to initiate VoIP connections.

Corporate and telco use of VoIP

Although few office environments and even fewer homes use a pure VoIP infrastructure, telecommunications providers routinely use IP telephony, often over a dedicated IP network, to connect between their switching stations, where they convert the dedicated voice signal to IP packets and back. The result is a data-abstracted digital network which the provider can easily upgrade and use for multiple purposes.

Corporate customer support centers which provide support over telephone often use IP telephony exclusively to take advantage of the data abstraction that comes with it.

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 do not incur any expenses other than the internet connection fees.

VoIP implementation challenges

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 queuing for this class of service.
• 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.

VoIP protocols

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

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
• IAX, the Inter-Asterisk eXchange protocol used by the Asterisk open source PBX server and associated client software.

Mass-market telephony over broadband Internet access

A new development has been the introduction of mass-market VoIP services over broadband Internet access services, in which subscribers make and receive calls as they would over the PSTN. This requires an analog telephone adapter (ATA) to connect a telephone to the broadband internet connection. Companies in the US, such as Vonage, VoicePulse, and Packet8, use IP to offer unlimited calling to the US, and sometimes to Canada or to selected countries in Europe or Asia, for a flat monthly fee. One advantage of this is the ability to make and receive calls as you would at home, anywhere in the world, at no extra cost. As calls go via IP, this does not incur charges as call diversion does via the PSTN, and the called party does not have to pay for the call.

For example, somebody may call you on a number with a US area code, but you could be in London, and if you were to call another number with that area code, it would be treated as a local call, regardless of where you are in the world. However, the broadband phone is likely to complement, rather than replace a PSTN line, as it still needs a power supply, while calling the US emergency services number 911, may not automatically be routed to the nearest local emergency dispatch center, or be of any use for subscribers outside the US.

Another challenge for these services is the proper handling of outgoing calls from Fax machines, TiVO boxes, satellite television receivers, alarm systems, conventional modems or FAXmodems, and other similar devices that depend on access to a voice-grade telephone line for some or all of their functionality. At present, these types of calls sometimes go through without a hitch, but in other cases they won't go through at all. And in some cases, this equipment can be made to work over a VoIP connection if the sending speed can be changed to a lower bits per second rate. If VoIP and cellular substitution becomes very popular, some ancillary equipment makers may be forced to redesign equipment, because it can no longer be assumed that a conventional voice-grade telephone line will be available in nearly every home in the United States and Canada.

There is also a free service called Free World Dialup (FWD), that permits users to make free telephone calls to other FWD users, and that has only limited connections to and from the public switched telephone network.

See also

• SIPphone
• Skype
• IntServ
• DiffServ
• computer conferencing

External links

• BroadbandReports.Com VoIP forum
• Voxilla.com
• How to Distribute VoIP Throughout a Home

Referenced By