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.

Timing:

VoIP History January 03, 2005

Tom Keating and I've been in telecom/datacom since 1994 when I joined Technology Marketing Corporation (TMC), a publisher of VoIP news, call center & CRM news, as well as other telecom information.

I often get asked "When did VoIP begin?" or "What is the history of VoIP?". Well, VoIP goes further back in time than most would expect - 9 years in fact. In fact, I did the first product review of a VoIP product back in 1996 within CTI Magazine (which later became Communication Solutions Magazine). CTI (Computer Telephony Integration) Magazine was at the forefront of news, articles, and reviews of anything to do with computers and telecom (or telephony if you prefer). In 1996 (aka 'ancient history'), CTI was best known for "screen pops", IVRs with database integration, but it also came to symbolize the convergence of the telecom and the datacom worlds. Most of this "CTI" technology was deployed first and foremost in the #1 industry that uses the telephone - namely the call center industry. Since CTI brings efficiencies to the call center, call centers were one of the first industries to deploy CTI. CTI Magazine was all about "convergence". It was perhaps a bit ahead of its time, since it predicted and preached "convergence", but alas, this magazine is now defunct, but not before spinning off the most successful VoIP magazine within the VoIP industry - Internet Telephony Magazine, which was launched in late 1997 with the first issue appearing in early 1998. So CTI Magazine certainly helped lay the foundation for convergence, which is certainly all the rage in any magazine or newspaper that you read now, including the New York Times, Newsweek, or the Wall Street Journal. Internet Telephony Magazine is now the torchbearer of VoIP and I could not be prouder than to write for this magazine. Getting back to my history lesson, in 1996, I wrote the first product review of one of the true pioneers of VoIP - Vocaltec. I reviewed Vocaltec's Internet Phone product, which was perhaps the first "true" VoIP software application. It helped lay the groundwork to make VoIP mainstream. In fact, to my knowledge, Vocaltec's Internet Phone was the first VoIP product on the shelves of Compusa and other retail outlets. In re-reading my review, it reminded me of the old days of VoIP full-duplex issues/soundcard full-duplex driver issues. If you didn't have the latest sound card driver, you'd get a half-duplex CB/walkie-talkie type experience. In fact, the Internet hadn't really taken off at that point in history, so I had to use Compuserve of all things to download the latest sound card driver to get full-duplex VoIP sound. Ahhh, now those were the days... I miss BBSs (Bulletin Board Systems) as well.

If I may sidetrack for a moment, similar to MTV's "I Love the 80's", "I Love the 70's" shows, someone should have a retro-history technology equivalent for us tech geeks. Maybe if G4TechTV (formerly TechTV) is reading my blog, they can launch such a show. Anyway, here's a screenshot of my actual Vocaltec Internet Phone review. For nostalgia sake, click on the image below to read the review.

Web Resources:

[1] Voice over IP

Retrieved from "http://en.wikipedia.org/wiki/VoIP"