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Voice over Internet Protocol (VoIP) is a recently developed, cost-saving alternative to traditional telephone service that enables voice data to be transported over IP networks, like the Internet, instead of... more/see it nowthe public switched telephone network (PSTN) or a cellular network.
VoIP, which operates strictly over IP networks, can connect to other VoIP nodes or traditional phone lines. The IP network used may be the Internet or a private network.
In either instance, the actual data-transport portion of this network can still be made up of the full gamut of network services: high-speed leased lines, Frame Relay, ATM, DSL, copper, fiber, wireless, satellite, and microwave signals. VoIP simply digitizes voice data and adds it to other information traveling along the same network.
With this flexible technology, a phone call can be placed between two PCs, between a PC and a standard telephone, between a PC and an IP phone, between
an IP phone and a standard telephone, or between two IP phones. It will take a long time for the PSTN to support this technology seamlessly, but this seems to be the direction in which phone systems are headed.
Benefits of VoIP
Because VoIP is inexpensive, has a worldwide reach, and operates on a few simple principles, it’s exploded in popularity in recent years—especially among both small and large businesses that incur significant long-distance telephone expenses.
Without question, the primary benefit
of a VoIP system is decreasing or eliminating long-distance telephone charges. Organizations with a high volume of long-distance voice traffic stand to save quite a lot of money by implementing a VoIP system. However, this factor alone may not warrant a full commitment to VoIP for some companies.
Setup fees for VoIP are usually quite low so your organization can generally start saving money after only a month or two of service. And with the wide variety of VoIP products and services on the market, it’s easier than ever to set up a VoIP phone system over your network.
VoIP can be set up in a way that enables you to use phone numbers in exactly the same way as you did before VoIP. Most of the services you get with traditional phone service—Voice Mail, Call Waiting, and Call Routing, for instance—are also available with VoIP.
VoIP doesn’t interfere with other network services either, so you can surf the Web while making a VoIP call.
VoIP doesn’t tie you to one phone or to a single location. Anywhere you find high-speed reliable Internet access, you can use VoIP. Your phone number stays the same wherever you are—office, home, hotel,
or even traveling overseas.
Although the ITU standards for VoIP have evolved significantly in the last few years, VoIP is still suffering from a lack of generally accepted interoperability standards.
H.323, a standard for real-time audio, video, and data communications across IP-based networks (including the Internet), is almost universally accepted as the primary standard for VoIP call setup and signaling. It’s actually a collection of standards that works together for sending multimedia and data over networks that don’t provide guaranteed Quality of Service (QoS).
The H.323 standard includes:
- Real-Time Transport Protocol (RTP) specifies end-to-end network transport functions for applications transmitting real-time data such as video. RTP provides services like payload type identification, sequence numbering, time stamping, and delivery monitoring to real-time applications. Plus, it works with RTCP.
- Real-time Transport Control Protocol (RTCP) works with RTP to provide a feedback mechanism, providing QoS status and control information to the streaming server.
- Registration, Admission, Status (RAS) is a gateway protocol that manages functions such as signaling, registration, admissions, bandwidth changes, status, and disengage procedures.
- Q.931 manages call setup and termination.
- H.245 negotiates channel usage and capabilities.
- H.235 provides security and authentication.
As VoIP product manufacturers began conducting interoperability tests for more complex operations, they recognized that they needed a simpler and more adaptable standard for call handling and signaling protocol.
To this end, the IETF developed the Session Initiation Protocol (SIP). SIP is built with less computer code than H.323 is, so it’s less cumbersome. Because SIP is similar in nature to HTML—it uses ASCII text for configuration—users can adapt it more easily for specific VoIP systems. In contrast, modifying H.323 for VoIP applications requires a knowledgeable computer programmer.
Both H.323 and SIP are considered “thick clients,” where intelligence is maintained in the end devices such as IP telephones. In this respect, H.323 has a head start, although most VoIP systems today support both H.323 and SIP.
Despite the fact that VoIP standards are still developing, providers are already flooding the market with products and services while forming partnerships and matching expertise to strengthen their position in this new market. The biggest of these players and alliances—the ones who have the size and experience to grasp technical issues and quickly build infrastructures over which to offer VoIP services—are able to keep up with (and often influence) the continual changes in this market and keep rolling out new services.
A VoIP system depends on devices that connect your traditional phone or phone system to an IP network. Components that you’ll see in a VoIP system include:
- End-user devices
- Gateways or gatekeepers
- IP Networks
End-user devices are usually VoIP telephones or PCs running VoIP software. End-user devices have their own IP address and make a direct connection to the IP network.
A gateway is a device that converts circuit-switched analog voice calls from a traditional PBX into VoIP packets and transmits them over an IP network either
to another gateway or directly to an end-user device.
A gateway can have additional features such as voice compression, echo cancellation, and packet prioritization.
Because VoIP-enabled end-user devices can communicate directly with each other over an IP network, a gateway is not a required component of a VoIP system as long as the VoIP devices are connected directly to the IP network.
An IPBX is a PBX with a built-in gateway. IPBX systems are equipped for hundreds of telephone ports, with WAN support for trunk connections to the PSTN, and with high-speed IP WAN links. In addition to VoIP features, these systems usually include other features typical of traditional PBX systems such as music on hold, auto-attendant, and call management. Often, they include Ethernet ports to support VoIP telephones.
VoIP can be set up with or without a connection to standard PSTN phone service. You can, of course, place calls over the Internet directly from your PC or IP phone to another VoIP-enabled device. But what makes VoIP so versatile is that, through the use of a gateway service, it can also be used to call the numbers of phones connected to standard land-line or cellular phone services. They can also receive calls from standard telephones.
Not all fun and free calls
There are still things to consider when you’re deciding whether or not to invest in VoIP.
Much of the government regulation of VoIP is still being worked out. The U.S. government hasn’t decided whether VoIP is going to be regulated as phone service
or whether to tax it. VoIP isn’t available worldwide because some governments fear the loss of tax revenue or control.
Although older VoIP equipment may still have some compatibility issues, current VoIP products from different vendors generally work together.
For all the popular talk about VoIP being free, it isn’t truly free. Any VoIP system has costs associated with its implementation—equipment, high-speed Internet access, and gateway service. So, although it’s inexpensive, it’s a long way from being free. For organizations with a high volume of long-distance calls, especially to international locations, VoIP almost always pays for itself quickly. However, private users or organizations with a low volume of long-distance calls primarily within the U.S., may find that a standard service is actually more economical in the short- to mid-term.
VoIP depends on having a fast, reliable network to operate. A fast network connection with guaranteed bandwidth is not a problem in a corporate intranet where you have complete control over
the network. However, if you’re using the Internet for VoIP, you’re using a public network that may be subject to slowdowns that cause drop-outs and distortion. You may find that your high-speed Internet connection is faster than the actual Internet and that the quality of your connection is generally unacceptable or is unacceptable at times when Internet usage is high.
There are four common network issues that can cause problems with a VoIP system:
- Latency is a delay in data transmission. With VoIP, this usually results in people speaking over one another because neither can tell when the other is finished talking.
- Loss. Losing a small percentage of voice transmission doesn’t affect VoIP, but too much (more than 1%) compromises the quality of the call.
- Jitter—is common to congested networks with bursty traffic. Jitter can be managed to some degree with software buffers.
- Sequence errors—or changes in the order of packets when they’re recompiled at the receiving station, degrades sound quality.
If you subscribe to a VoIP gateway service that enables you to use your VoIP phone like a regular phone, be aware that you may not be able to call 911 for emergencies. If 911 service is important to you because you don’t have an alternative way to call 911, shop for a VoIP provider who does provide this service.
Consider, too, that VoIP needs both working Internet access and power to work. If you lose your Internet service, your phone goes, too. And, unlike regular phone service that can keep basic telephones working when the power goes out, VoIP needs power—if you lose power, you lose your phone.
Before VoIP technology becomes truly universal, the current worldwide PSTN will have to migrate to a packet-based IP equivalent. Industry inertia alone dictates this will not occur instantly. The current worldwide PSTN system has grown to what it is over a period of 125 years. Given the sheer complexity of the existing PSTN, the migration to an IP packet network will probably occur during several decades.
As migration from the PSTN to IP-based networks proceeds, businesses and home users will gradually discover reasons of their own to implement VoIP. It won’t happen right away, but we predict that VoIP will become a big part of telecommunications
in the not-so-distant future.
Although it’s not quite as convenient as conventional phone service, VoIP can offer serious savings—particularly if you now regularly pay for multiple overseas phone calls. Keep in mind though, VoIP isn’t a one-size-fits-all solution. But with a little planning, VoIP could spell savings for you! collapse
This hands-on Tips for Techs video takes you inside the Black Box Training Center to show you how to terminate and test cabling for a copper installation. The video begins... more/see it nowwith a step-by-step demonstration of terminating a patch cable up to the CAT6 specification, with special focus on how to cut and strip the cabling in a way that ensures good signaling integrity. Next, it covers how to validate and verify a backbone link between a patch panel and workstation. The video tutorial then demonstrates what kind of tester you’ll need if you must certify and document a copper link, then ends with a discussion on what you’ll need to simply verify the Ethernet performance of a cable run. collapse
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