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VoIP Functions: Signaling, CODEC, Call connect

VoIP components must be able perform the same features as the PSTN network, which are
  • Signaling

  • Database services

  • Call connect and disconnect (bearer control)

  • CODEC operations

Signaling

Signaling in a VoIP network is just as critical as it is in the legacy phone system. The signaling in a VoIP network activates and coordinates the various components to complete a call. Although the underling nature of the signaling is the same, there are some technical and architectural differences in a VoIP network.

Signaling in a VoIP network is accomplished by the exchange of IP datagram messages between the components. The format of these messages is covered by any number of standard protocols. Regardless of which protocol and product suites that are used, these message streams are critical to the function of a voice-enabled network and might need special treatment to guarantee their delivery.

Database Services

Database services are a way to locate an endpoint and translate the addressing that two (usually heterogeneous) networks use. For example, the PSTN uses phone numbers to identify endpoints, while a VoIP network could use an IP address (address abstraction could be accomplished with DNS) and port numbers to identify an endpoint. A call control database contains these mappings and translations. Another important feature is the generation of transaction reports for billing purposes. You can employ additional logic to provide network security, such as to deny a specific endpoint from making overseas calls on the PSTN side. This functionality, coupled with call state control, coordinates the activities of the elements in a VoIP network.

Call Connect and Disconnect (Bearer Control)

The connection of a call is made by two endpoints opening communications sessions between each other. In the PSTN, the public (or private) switch connects logical DS-0 channels through the network to complete the calls. In a VoIP implementation, this connection is a multimedia stream (audio, video, or both) transported in real time. This connection is the bearer channel and represents the voice or video content being delivered. When communication is complete, the IP sessions are released and optionally network resources are freed.

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.