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PING TRACEROUTE PATHPING

PING TRACEROUTE PATHPING

Control devices are intended by the multipoint H.323 conferences for the organization of multimedia information exchange with an involvement of three and more subscribers. These devices can be separate venues

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or can be implemented in the form of the functional modules entering other H.323 devices. Distinguish two types of modules of the organization of conferences:

□ Multipoint Controllers is controllers of conferences;

□ Multipoint Processors is processors of conferences.

Controllers of conferences provide accomplishment of conferences of preliminary procedures for coordination of protocols and transmission formats of data, necessary for the organization. Controllers of conferences can enter as the functional modules such devices of recommendation H.323 as the terminal, the gateway and the janitor.

Processors of conferences are intended for direct manipulation by flows of the multimedia information. Processors of conferences can enter as the functional modules such devices of recommendation H.323 as the gateway and the janitor.

Terminal H.323 devices

As it was already marked above, terminal H.323 devices execute direct and inverse transformation of media information flows in the sequence of the data units intended for drive through a packet switched network in our case — through the Internet. Ha of fig. 7.3 is introduced the functional diagram of the organization of information interaction of the terminal H.323 device with sources of external information.

Fig. 7.3. The functional diagram of interaction of the H.323 terminal with sources of external information.

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Components and functions of terminal H.323 devices

B composition of the terminal H.323 device can enter the following function blocks:

□ Video Codec is a video codec (codec);

□ Audio Codec — an audio codec;

□ Data Channel is transmission channel;

□ System Control Unit is the system control unit.

Video codecs of H.323 terminals

Video codecs fall into to optional components of the terminal H.323 device. They are intended for accomplishment of an interconversion of a video stream in the sequence of data units of the network layer. For the video terminal H.323 support of the QCIF mode (Quarter Common Intermediate Format is the reduced multiple-purpose format of submission of the image) H.261 rekomevdation is mandatory. B addition to it the video terminals H.323 can support any modes determined by the recommendations of H.261 and H.263. Also use of other formats of data representation of a video stream approved by means of the H.245 procedure is allowed. B quality of an additional possibility of the video terminal H.323 can be organized the mode of simultaneous drive and acceptance of several video streams.

Critical parameters of the entering video traffic: the transmission rate of video data and the image format acceptable for the terminal decoder and also use of additional algorithms shall be approved by means of the H.245 procedure. At the same time the code device of the remote video terminal can use any of the modes supported by the decoder for representation of video data. The video decoder can initiate the procedure of establishment of the concrete mode of forming of a video traffic, however adequate response to such request is not mandatory for the video coder of the remote terminal H.323.

The video terminals H.323 shall provide a possibility of acceptance and drive of asymmetric video streams with different values of data transmission rate and the image format. B that case if the video terminal provides a possibility of acceptance of several video streams, as an additional opportunity can be used the mode of commixing of video images.

Audio codecs of H.323 terminals.

Audio codecs (voice) are also optional components of the terminal H.323 device. They are intended for accomplishment

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interconversion of an audio stream in the sequence of data units of the network layer. For the voice terminal the possibility of coding and decoding of an audio stream according to requirements of recommendation G.711 is mandatory. At the same time encoding and decoders of the terminal shall provide non-uniform encoding and decoding of a voice signal according to diagrams of A-Iaw and c-law.

^ Note ^

Diagrams of non-uniform encoding A-Iaw and c-law are used respectively in the European and American telephone systems for compensating of impact of a quantization noise on the transmitted signal.

Also use of other algorithms of data representation of an audio stream approved by means of the H.245 procedure is allowed.

H.323 terminal. shall provide use of the asymmetric modes when the receiver and the transmitter use different transform circuits of an audio stream. The format of an audio stream shall conform to the requirements stated in a rekomevdation of H.225.0. The packets containing coded data of an audio stream shall be passed between a transport layer protocol with the period determined by type of the used encoding diagram (audio frame interval). H.323 terminals can provide a concomitant use of several voice flows. B this case on the terminal can be used the additional mode providing commixing of the received flows.

For support of time mutual synchronization of audio-and video streams H.323 terminals shall provide a possibility of forming and drive of the message of “h2250MaximiimSkewIndication” defining the maximum value of a temporal mismatch of audio-and video streams of one direction.

Voice G.711 codec

B of the recommendation ITU-T G.711 are given the principles of implementation and the requirement to the main characteristics of the encoding and decoding devices using an algorithm of classical PCM for conversion of an analog signal. The range of change of the digitized signal makes 8 bits, conversion is made with clock rate 8 kHz. Thus, data stream rate on an output of the code device makes 64 Kbps (8 bits x 8 kHz). For drop of influence of a quantization noise on signals with small range when coding non-linear conversion of an input analog signal according to diagrams of A-Iaw or c-law is used.

Thanks to singleness of implementation the G.711 codec is widespread in the systems of traditional telephony circuit switched. However codecs, it is constructed –

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ny according to requirements of a rekomevdation of G.711, are rather seldom applied in the modern systems of the integrated data transfer because of the high requirements imposed to the handling capacity of the used port. Use of the encoding diagrams meeting the requirements of a rekomevdation of G.711 can be read justified only when it is required to provide the maximum quality of coding of the voice information in case of a small number of double talks.

Voice G.723.1 codec

B of the recommendation ITU-T G.723.1 are given the principles of implementation and the requirement to the main characteristics of the encoding and decoding devices using technology of multipulse multilevel quantization for conversion of voice information (Miilty Pulse-Multy Level Quantization — MP-MLQ). The code devices G.723.1 execute conversion of an analog signal to a data stream with a speed of 64 Kbps on a standard algorithm of PCM, and then by means of the multiple digital bandpass filter select the frequency phonemes in the created signal and form the value of the resultant output code determined by a current status of phonemes in an input analog signal. Use of a similar algorithm of coding allows to reduce transmission rate of encoded vocalization of information to 5.3 — 6.3 Kbps without notable quality degradation of the speech.

The encoding diagrams ITU-T G.723.1 provide a possibility of use of two options of coding. The first option assumes use of the encoding algorithm MP-MLQ and provides forming of a digital stream with intensity no more than 6.3 Kbps. The second option assumes use of the encoding algorithm of the linear extrapolation with code initiation (Code Excited Linear Prediction — CELP) and provides forming of a digital stream with intensity no more than 5.3 Kbps. The algorithm MP-MLQ is held for use on networks with packet forwarding of a voice and provides the best quality of coding in comparison with CELP option. However this algorithm is rather less fitted to use on networks with the commixed type of a traffic (voices/data).

Voice G.726 codec

B of the recommendation ITU-T G.726 are given the principles of implementation and the requirement to the main characteristics of the encoding and decoding devices using technology of the adaptive differential pulse code modulation (ADPCM) for conversion of voice information. Use of this technology provides forming of a digital stream with intensity 32 Kbps, 24 Kbps and 16 Kbps. The encoding diagrams constructed according to requirements of the recommendation of ITU-T G.726 can be applied jointly with diagrams G.711 to increase in effectiveness of use of network resources.

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Voice G.728 codec

B of the recommendation ITU-T G.728 are given the principles of implementation and the requirement to the main characteristics of the encoding and decoding devices using for conversion of voice information technology of the linear extrapolation with code initiation and short delays (Low Delay Code Excited Linear Prediction — LD-CELP).

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