Title - Voice over IP (VoIP) Applications

Compelling VoIP Applications

The VoIP technology only becomes useful when compelling applications meet the needs of customers. Three such applications are corporations that replace PBX systems, cable operators offering telephony services using their plant, and video conferencing. These existing applications are the driving factors in allowing manufactures to make equipment, service providers to offer services, and customers to increase their productivity. Having an established market and a profitable business model allows VoIP to begin addressing the next generation of applications which then will allow the market to continue to grow.

Corporate

LAN-based telephony applications offer attractive business models to consumers today. The most important applications are the replacement of the traditional PBX, new client side applications using the PC, and the reduction in maintenance expenses for wiring changes.

LAN based PBX systems provide superior return on investment to traditional PBX systems. Although initial equipment costs are comparable, LAN PBXs typically cost much less than PBXs to install because they use the existing data infrastructure (Category 5 cabling) rather than separate voice wiring. Administration is also less burdensome because LAN and server administrators can manage the system without the need for dedicated telephony technicians.

The most compelling reason businesses consider IP telephony-type applications is for the integration of applications with voice. Over the years, a significant amount of work has gone into computer telephony integration (CTI) in traditional PBXs. These systems began to offer application-programming interfaces such as Telephony API (TAPI), Telephony Services API (TSAPI), and Java Telephony API (JTAPI). This work has resulted in advanced call center functions, including screen pops for agents and active call routing between call centers.

Client (end station) products offer LAN telephony services through the use of a software client on the user's PC, while others offer telephone instruments that plug into the LAN. When there is a need to relocate the equipment, it needs only to be unplugged from one data port and plugged in at the new destination. This process avoids the wiring changes typically done for convention phones and thus VoIP clients reduce the cost of ownership. Over time, these savings add up to the point that a LAN-based telephony system can offer considerable savings over traditional PBXs.

PacketCable

PacketCable™ is the telephony architecture for using VoIP over the Cable TV system. With buried Cable TV plant passing hundreds of millions of homes worldwide, it is logical to assume that cable operators desire to offer new services that make use of their installed system.

The PacketCable standards (1.0, 1.1 and 1.2) were developed by CableLabs®, the research group of the cable operators and makes use of the same equipment used for the cable modem services. The cable modem architecture, known as DOCSIS (Data over Cable Service Interface Specification), is able to meet the requirements of transporting VoIP packets. DOCSIS version 1.1 provides Quality of Service (QoS), security features, and the prioritization of packet traffic that is necessary for voice communication.

The PacketCable specification also incorporates the Network-based Call Signaling (NCS) protocol for signaling voice calls over cable networks. NCS leverages the existing Media Gateway Control Protocol (MGCP) and the protocol is sometimes referred to as MGCP NCS. NCS uses network-based call agents to negotiate cable-based IP telephony calls.

Traditional telephones draw all the power they need from the phone lines. Part of the reason that the public phone system has evolved to such a reliable state, is that it is essentially immune from the effects of power outages. Electrical utilities in most areas do not offer this degree of unfailing reliability and this reality is an important issue for the cable television plant.
Normally, head-end and customer-premises cable equipment rely solely on the local electric company for their power and this puts users at risk of losing phone service should a power outage occur. There is not universal agreement among the cable operators about offering "lifeline" telephony service.

Some Time Warner systems focus their telephony service towards the "second" phone line in the home. They suggest that the customer keep the existing primary line and then add the Time Warner telephony service for a business line, fax line, or children's phone. AT&T has taken a different approach by enhancing many of their networks with alternate power sources thereby allowing lifeline telephony services to their customers.

The PacketCable specifications are available at www.packetcable.com/specifications.html.

Video Conferencing

Video conferencing includes the use of packetized voice and is an important application for the home and business. There are several video conferencing standards and one of the significant standards is known as H.32x where x can be 0, 1, 2, 3, or 4 representing video conferencing on different types of communications links. The H.32x model includes establishing connections between the source and destination devices and this method connection establishment has been adapted for use in VoIP gateways.

 

Table 1. The ITU-T H.32x Video Conferencing Standards

 

H.320

H.321

H.322

H.323

H.324

Approval Date

1990

1995

1995

1996

1996

Network

Narrowband switched digital ISDN

Broadband ISDN ATM LAN

QoS packet switched networks

Non-QoS networks

(Ethernet)

The analog phone system

Audio

G.711

G.722

G.728

G.711

G.722

G.728

G.711

G.722

G.728

G.711

G.722

G.728

G.723

G.729

G.723

Video

H.261

H.261/H.263

H.261/H.263

H.261/H.263

H.261/H.263

Control

H.230/H.242

H.242

H.230/H.242

H.245

H.245

Multiplexing

H.221

H.221

H.221

H.225

H.213

Comm. Interface

I.400

AAL 
I.363 
AJM I.361 
PHY I.400

I.400& 
TCP/IP

TCP/IP

V.34 Modem


A summary of the audio coding recommendations can be found in the section Voice Coding Algorithms.

The H.323 recommendation supports the largest number of audio coding standards. In fact, the audio codecs of the other recommendations are a subset of the audio codecs outlined in the H.323 recommendation. In addition, the H.323 is the recommendation that is made for a non-guaranteed bandwidth, packet switched networks such as the Internet.

H.323 Video Conferencing

The H.323 recommendation covers the technical requirements for audio and video communications services in LANs that do not provide a guaranteed Quality of Service (QoS). The scope of H.323 does not include the LAN itself or the transport layer that may be used to connect various LANs. Only elements needed for interaction with the Switched Circuit Network (SCN) are within the scope of H.323.

H.323 defines four major components for a network-based communications system: Terminals, Gateways, Gatekeepers, and Multipoint Control Units (MCUs). The terminal must support voice transmission with data and video transfer being an option.

Figure 1. The H.323 Protocol Stack

 

H.323 uses the following standards as part of the video conferencing protocol stack (Figure 1):

  • Q.931 QoS - Quality of Service. When the connection between devices is being established, the end stations and each data link in the route negotiate to determine what bandwidth is available, how much delay the application can tolerate, and how much jitter there will be in the packet arrival. Once the links agree to the QoS message, they must guarantee those parameters for the duration of the video conferencing session.

  • H.245 - Control Channel Protocol. Provides capability negotiation between the two end-points such as voice compression algorithm to use, conferencing requests, etc. The H.245 channels transport must be reliable (e.g. TCP, SPX)

  • H.225 RAS - Registration, Admission, and Status (RAS) Protocol. Used to convey the registration, admissions, bandwidth change and status messages between IP Telephone devices and servers called Gatekeepers that provide address translation and access control to devices.

  • RTCP - Real-time Transport Control Protocol (RTCP). Provides statistics information for monitoring the quality of service of the voice call.

Control messages (Q.931 signaling, H.245 capability exchange and the RAS protocol) are carried over the reliable TCP layer. This ensures that important messages get retransmitted if necessary so they can make it to the other side. Media traffic is transported over the unreliable UDP layer and includes two protocols: RTP (Real-Time Protocol) that carries the actual media and RTCP (Real-Time Control Protocol) that includes periodic status and control messages. Audio and Video information is carried over UDP because it need not be retransmitted because if a sound packet is lost and then transmitted, it would most probably arrive too late to be used for the real-time conferencing.

More Information

Additional VoIP seminars:

An Introduction to VoIP - An overview of the VoIP technology, architecture, and the interconnection issues.

Voice Coding Algorithms - A description of the various methods for digitizing speech.

VoIP Problems - Deployment of VoIP has been slower than expected because of problems with underlying networks, standardization issues, and network control devices.

 


In Summary:

  • Corporate applications include replacing the PBX, integrating phone applications with the corporate data bases, and reducing the cost of moving phones between areas.

  • PacketCable uses the DOCSIS 1.1 cable modem infrastructure to provide telephony services by cable operators..

  • Video Conferencing is an important application and its control mechanisms are being incorporated into VoIP gateways.

 

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