The ITU also defines the compression and decompression standards. Standard G.711 defines the 64-kbit/s signal, more commonly known as pulse code modulation (PCM). An 8-kbit/s voice signal known as G.729a probably is the most commonly used compression standard.
The G.723 compression standard produces a lower-quality 5.3-kbit/s signal, while G.723.1 produces a 6.4-kbit/s voice stream. Such high compression is possible because so much of a voice conversation is silence, including pauses between words and the conversation exchange. A DSP usually performs the compression as well as echo cancellation, a common problem in analog or digital telephony.
At the other end of the conversation, the compressed voice is decompressed and converted back into its PCM format. This format is then converted back to voice with a digitaltoanalog converter (DAC). The ADC and DAC plus the serialize/deserialize function usually are called the codec (coder-decoder). Then the compressed voice is packetized and transmitted over a local-area network, and subsequently the Internet. In many corporate systems, the packetized data is transmitted via the widely available asynchronous transfer mode (ATM).
The key problem in VoIP is the latency, or the time it takes for voice packets to traverse through all of the various LAN, metro-area-network (MAN), and wide-area-network (WAN) connections. Up to a 150-ms delay is permitted, as longer delays produce an annoying voice lag. For example, international reporters on television often take several seconds to respond to questions, which must pass through the 22,300-mile uplink and back. In two-way phone conversations, such delays are unacceptable.
Several different types of IP phone arrangements exist. Each phone in an enterprise system is an IP phone. This means it contains the codec, compression/decompression, SIP, and other circuits that generate Real Time Protocol (RTP) and User Datagram Protocol (UDP) packets, which eventually get translated into TCP/IP. These can go on the Internet. Or, the VoIP packets can be transmitted via ATM.
In most modern systems, VoIP packets are translated into Ethernet packets for connection to a LAN. The usual connection from a desk phone to the LAN is the ever-popular CAT5 UTP cable with RJ-45 connectors. The phone simply appears as a node on the enterprise LAN, and servers deal with it accordingly. In most cases, an employee has a PC as well as a phone. Thus, most current VoIP chips feature a built-in, twoport Ethernet switch so that both the phone and PC only need one LAN connection.
The typical home IP phone differs. While homeowners can buy an IP phone, in most cases any existing phone can be plugged into the RJ-11 modular jack on a piece of equipment called an analog terminal adapter (ATA) supplied by the IP phone provider. The internal home wiring can be disconnected from the phone company's incoming line, and all of it can connect to the RJ-11 input on the ATA. As a result, any phone in the house can make a call as usual.
The ATA contains the circuitry that terminates the standard phone and then takes the resulting analog voice and processes it (Fig. 1). It contains a subscriber line interface (SLIC) that the standard telephone expects to see. Afterward, the usual IP processing occurs, with the resulting Ethernet packets sent out over the broadband connection—either a DSL line or cable TV connection. With VoIP becoming more common, cable-TV companies and DSL providers are simply building the VoIP circuitry directly into set-top boxes and modems, readying them for VoIP service.
WIRELESS VoIP
VoIP via wireless is possible, too. The most common arrangement is 802.11/Wi-Fi. NEC, Motorola, and Nokia already make dual-mode phones that include a standard cell phone plus Wi-Fi VoIP. Subscribers can make a call using the traditional cell-phone network or by Wi-Fi through an available hotspot or company wireless access point.
Such phones should be attractive for larger companies, because employees can be issued cell phones that also double as cordless internal telephones. Consumers can use such an arrangement in which the Wi-Fi phone talks to the wireless router connected to the broadband line, turning it into a cordless home phone.
This technology, Unlicensed Mobile Access (UMA), is also known as the Generic Access Network (GAN). It has been formally standardized by the 3rd Generation Partnership Project (3GPP), the standard organization associated with 3G UMTS WCDMA cell-phone technology. The UMA system uses the existing ITU standard, which includes H.323 for call setup.
A similar system, seamless converged communications across networks (SCCAN), was developed by Motorola, Avaya, and Proxim. It also uses GSM and Wi-Fi but incorporates the more popular SIP standard of the IETF. In addition, the IEEE is working on a standard (802.21) that will provide a common method of handoffs between 802 networks and non-802 networks.
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