Let's use the data in Table 3.2 to calculate what speed and load are required for each node of a network for providing services. The calculation is performed with the following formula

- Loading:

-Speed transfer

:

Where:

Reliability of use to - ї services

(or С in the table):quantity of calls per hour

Speed of data transmission

Определения технологии, моделирования и оптимизация

The VOIP`s Protocols
The most frequently referenced VoIP protocols are the call-signaling protocols. VoIP networks use these protocols to locate the device at the other end of the communication and then negotiate the exchange between the sending and receiving devices. Here are the two most often used call-signaling protocols:
-Session Initiation Protocol (SIP), defined by the Internet Engineering Task Force (IETF)
-H.323, defined by the International Telecommunications Union (ITU)

These two protocols basically do the same thing, and most VoIP devices use one or the other. Under the hood, the two protocols work differently to accomplish the establishment of a VoIP connection; SIP is ASCII-based, and H.323 is binary-based. Although H.323 was by far the more popular at first – and many feel it’s superior in its ability to work with the public switched telephone network (PSTN) and to transmit video — SIP has become increasingly popular due to support from the devices of many VoIP vendors. Many users also find SIP to be easier to deploy.
SIP is an application layer protocol that provides a means for identification of the calling and called numbers, authentication of the caller and recipient, and forwarding of calls. In identifying the caller and recipient, SIP addresses are similar to the PSTN with phone numbers, but SIP addresses look a little like e-mail addresses; the format is sip:userID@gateway.com.
Users register their addresses with SIP servers called registrars, and the caller sends a SIP request to the server. Users can send SIP messages over either TCP or User Datagram Protocol (UDP).
H.323 is a suite made up of a number of many different protocols that perform specific tasks together. Some members of the suite include:
-H.225.0, which establishes the connection
-H.332, used for large conferences
-H.235, which provides security and authentication
-H.245, which negotiates channel usage
- RAS, which handles registration, admission, and status messages

The VOIP codec

Let's consider the basic codecs used in devices of an IP-telephony. The codecs standardised ITU-T:
Codec G.711: Recommendation G.711 describes the codec using transformation of an analogue signal with accuracy of 8 bits, clock frequency 8 кГц and the elementary compression of amplitude of a signal. Speed of the data flow on a converter exit makes 64 Kbit/c (8 bits x 8 кГц). For decrease in noise of quantization and improvement of transformation of signals with small amplitude at coding nonlinear quantization on level according to the special pseudo-logarithmic law is used. Typical estimation MOS makes 4.2. Usually any device VoIP supports this type of coding.

Codec G.723.1: the given codecs are obliged By the occurrence to mobile communication systems. The given algorithm of transformation allows to lower speed of the coded information to 5,3 - 6,3 Kbit/with without appreciable deterioration of speech. The codec has two speeds and two variants of coding: 6,3 kbit/c with algorithm MP-MLQ (Multi-Pulse - Multi Level Quantization - plural pulse, multilevel квантизация) and 5,3 kbit/c with algorithm CELP (Code-Excited Linear Prediction - coding with a linear prediction).

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