VoIP

Description:

Voice over Internet Protocol, also called VoIP (pronounced "vee-oh-eye-pee" [1] or "voyp"), IP Telephony, Internet telephony, and Broadband Phone is the routing of voice conversations over the Internet or through any other IP-based network.

Protocols 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.

Voice over IP traffic can be deployed on any IP network, including those lacking a connection to the rest of the Internet, for instance on a local area network.

Enablers:

Cost In general, phone service via VoIP is free or costs less than equivalent service from traditional sources but similar to alternative PSTN (Public Switched Telephone Network) service providers. Some cost savings are due to using a single network to carry voice and data, especially where users have existing under-utilized network capacity they can use for VoIP at no additional cost. VoIP to VoIP phone calls on any provider are typically free, whilst VoIP to PSTN calls generally costs the VoIP user.

There are two types of PSTN to VoIP services: DID (Direct Inward Dialing) and access numbers. DID will connect the caller directly to the VoIP user while access numbers requires the caller to input the extension number of the VoIP user. Access numbers are usually charged as a local call to the caller and free to the VoIP user while DID usually has a monthly fee. There are also DID that are free to the VoIP user but is chargeable to the caller.

Functionality VoIP can facilitate tasks that may be more difficult to achieve using traditional phone networks:

Incoming phone calls can be automatically routed to your VoIP phone, regardless of where you are connected to the network. Take your VoIP phone with you on a trip, and wherever you connect to the Internet, you can receive incoming calls. Free phone numbers for use with VoIP are available in the USA, UK and other countries from organizations such as VoIP User. Call center agents using VoIP phones can work from anywhere with a sufficiently fast Internet connection. Many VoIP packages include PSTN features that most telcos normally charge extra for, or may be unavailable from your local telco, such as 3-way calling, call forwarding, automatic redial, etc.

Mobility VoIP allows users to travel anywhere in the world and still make and receive phone calls:

Subscribers of phone-line replacement services can make and receive local phone calls regardless of their location. For example, if a user has a New York City phone number and is traveling in Europe and someone calls the phone number, it will ring in Europe. Conversely, if a call is made from Europe to New York City, it will be treated as a local call. Of course, there must be a connection to the Internet e.g. WiFi to make all of this possible. Users of Instant Messenger based VoIP services can also travel anywhere in the world and make and receive phone calls. VoIP phones can integrate 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.

Inhibitors:

Drawbacks VoIP technology still has a few shortcomings that have led some to believe that it is not ready for widespread deployment. However, many industry analysts predicted that 2005 was the "Year of Inflection," where more IP PBX ports shipped than legacy digital PBX ports.

Implementation challenges Because IP does not provide a mechanism to ensure that data packets are delivered in sequential order, or provide Quality of Service guarantees, VoIP implementations face problems dealing with latency and jitter. This is especially true when satellite circuits are involved. The receiving node must restructure IP packets that may be out of order, delayed or missing, while ensuring that the audio stream maintains a proper time consistency. This functionality is usually accomplished by means of a jitter buffer.

Another challenge is routing VoIP traffic through firewalls and address translators. Private Session Border Controllers are used along with firewalls to enable VoIP calls to and from a protected enterprise network. Skype uses a proprietary protocol to route calls through other Skype peers on the network, allowing it to traverse symmetric NATs and firewalls. Other methods to traverse firewalls involve using protocols such as STUN or ICE.

VOIP challenges:

Delay. Packet loss. Jitter. Echo. Fixed delays cannot be controlled but some delays can be minimized by marking voice packets as being delay-sensitive (see, for example, Diffserv).

The principal cause of packet loss is congestion, which can be controlled by congestion management and avoidance. Carrier VoIP networks avoid congestion by means of traffic engineering.

Variation in delay is called Jitter. The effects of jitter can be mitigated by storing voice packets in a buffer (called a play-out buffer) upon arrival, before playing them out. This avoids a condition known as buffer underrun, in which the playout process runs out of voice data to play because the next voice packet has not yet arrived.

Common causes of echo include impedance mismatches in analog circuitry, and acoustic coupling of the transmit and receive signal at the receiving end.

Experts:

Google

Timing:

When the demand for new IP-addresses is high enough to force the industry to invest in IPv6 infrastructure. But before that happens IPv6 will be stimulated by many IPv6 networks that communicate over the web via IPv4 (tunneling/translation).

Web Resources:

[1] Voice over IP

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