Black Box VOE231A manual Benefits of VoIP, Savings, Convenience, Standards

Technically Speaking

VoIP. Portability

Voice over Internet Protocol (VoIP) is a cost-saving alternative to traditional telephone service that enables voice data to be transported over IP networks, like the Internet, instead of the 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 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 recently—especially among both small and large businesses that incur significant long-distance telephone expenses.

Savings

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.

Convenience

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.

Standards

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.

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