CODEC Operations                                                           

Voice communication is analog, while data networking is digital. The process of converting analog waveforms to digital information is done with a coder-decoder (CODEC, which is also known as a voice coder-decoder [VOCODER]). There are many ways an analog voice signal can be transformed, all of which are governed by various standards. The process of conversion is complex and beyond the scope of this paper. Suffice to say that most of the conversions are base on pulse coded modulation (PCM) or variations. Each encoding scheme has its own history and merit, along with its particular bandwidth needs.

In addition to performing the analog to digital conversion, CODECs compress the data stream, and provide echo cancellation. Compression of the represented waveform can afford you bandwidth savings. The bandwidth savings for the voice services can come in several forms and work at different levels.

For example, analog compression can be part of the encoding scheme (algorithm) and does not need further digital compression from the higher working layers of the media gateway application. Another way to save bandwidth is the use of silence suppression, which is the process of not sending voice packets between the gaps in human conversations. Using compression and/or silence suppression can result in sizable bandwidth savings. However, there are some applications that could be adversely affected by compression.

One example is the impact on modem users. Compression schemes can interfere with the functioning of modems by confusing the constellation encoding used. The result could be modems that never synchronize or modems that exhibit very poor throughput. Some gateways might implement some intelligence that can detect modem usage and disable compression. Another potential issue deals with low-bit-rate speech compression schemes, such as G.729 and G.723.1. These encoding schemes try to reproduce the subjective sound of the signal rather than the shape of the waveform.

A greater amount of packet loss or severe jitter is more noticeable than that of a non-compressed waveform. However, some standards might employ interleaving and other techniques that can minimize the effects of packet loss.The output from the CODECs is a data stream that is put into IP packets and transported across the network to an endpoint. These endpoints must use the standards, as well as a common set of CODEC parameters.

The result of using different standards or parameters on both ends is unintelligible communication. Table 1 lists some of the more important encoding standards covered by the International Telecommunications Union (ITU). As you can see, there is a price paid for reduced bandwidth consumption by increased conversion delay.

 

Table 1: ITU Encoding Standards

ITU Standard	Description	Bandwidth (Kbps)	Conversion Delay (ms)
G.711	PCM	64	< 1.00
G.721	ADPCM	32,16,24,40	< 1.00
G.728	LD-CELP	16	~ 2.50
G.729	CS-ACELP	8	~ 15.00
G.723.1	MultiRate CELP	6.3,5.3	~ 30.00

VoIP Components                                               

The major components of a VoIP network are very similar in functionality to that of a circuit-switched network. VoIP networks must perform all of the same tasks that the PSTN does, in addition to performing a gateway function to the existing public network. Although using different technology and approach, some of the same component concepts that make up the PSTN also create VoIP networks. There are three major pieces to a VoIP network.

• Media gateways
• Media gateway / signaling controllers
• IP network

Media Gateways

Media gateways are responsible for call origination, call detection, analog-to-digital conversion of voice, and creation of voice packets (CODEC functions). In addition, media gateways have optional features, such as voice (analog and/or digital) compression, echo cancellation, silence suppression, and statistics gathering.

The media gateway forms the interface that the voice content uses so that it can be transported over the IP network. Media gateways are the sources of bearer traffic. Typically, each conversation (call) is a single IP session transported by a Real-time Transport Protocol (RTP) that runs over UDP. Media gateways exist in several forms. For example, media gateways could be a dedicated telecommunication equipment chassis, or even a generic PC running VoIP software. Their features and services can include some or all of the following.

• Trunking gateways that interface between the telephone network and a VoIP network. Such gateways typically manage a large number of digital circuits.
• Residential gateways that provide a traditional analog interface to a VoIP network. Examples of residential gateways include cable modem/cable set-top boxes, xDSL devices,and broadband wireless devices.
• Access media gateways that provide a traditional analog or digital PBX interface to a VoIP network. Examples include small-scale (enterprise) VoIP gateways.
• Business media gateways that provide a traditional digital PBX interface or an integrated soft PBX interface to a VoIP network.
• Network access servers that can attach a modem to a telephone circuit and provide data access to the Internet.
• Discreet IP telephones units.

Media Gateway Controllers                                            

Media gateway controllers house the signaling and control services that coordinate the media gateway functions. Media gateway controllers could be considered similar to that of H.323 gatekeepers. The media gateway controller has the responsibility for some or all of the call signaling coordination, phone number translations, host lookup, resource management, and signaling gateway services to the PSTN (SS7 gateway). The amount of functionality is based on the particular VoIP enabling products used.

In a scalable VoIP network, you can breakup the role of a controller into signaling gateway controller and media gateway controller. For calls that originate and terminate within the domain of the VoIP network, only a media gateway controller might be needed to complete calls. However, a VoIP network is frequently connected to the public network. You could use a signaling gateway controller to directly connect to the SS7 network, while also interfacing to the VoIP network elements. This signaling controller would be dedicated to the message translation and signaling needed to bridge the PSTN to the VoIP network.

The services of these devices are defined by the protocols and software they are running. There are several protocols and implementations that any number of vendors could deploy. Knowing the details of how the devices use their suite of protocols is important to designing the IP backbone that is to service the VoIP elements.

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