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As discussed in 27, some inservice measurements are more representative than others of the actual error performance of the traffic signal. Furthermore, some are applicable to the path measurement, provided the parameters are not reset at an intermediate network node. Others are only useful at the line or section level. The most commonly used error detection codes (EDCs) are frame word errors, parity errors, or cyclic redundancy checksum errors. Out-of-service measurements involve removing live traffic from the link and replacing it with a known test signal, usually a pseudorandom binary sequence (PRBS). These tests are disruptive if applied to working networks, but are ideal for installation and commissioning tests because they give precise performance measurement. Every bit is checked for error. Although the PRBS appears random to the digital system, the error detector (Figure 5.2) knows exactly what it should receive and so detects every error. The error detector calculates the probability of error as the bit error ratio (BER). BER is defined as the number of errors counted in the measurement period, divided by the total number of bits received in the measurement period. Thus the bit errors or error events can be detected by out-of-service or in-service techniques. These are sometimes referred to as the performance primitives. To be useful for assessing quality of service (QoS), however, they must be analyzed statistically as a function of time according to the various error performance standards specified in Table 5.2. This analysis yields percentages for the availability of a digital communication link, and the portion of time that it exceeds certain performance criteria that are acceptable to the customer. One of the most important standards is the ITU-T Recommendation M.2100/2110. 5.6 Protocol Analysis in the Telecommunications Network Up to this point we have discussed the capability of the telecom network to transmit digital bits or analog signals over a path without errors or quality degradation. Testing BER, for example, assumes that the traffic carried by the network is completely random data, or at least that the payload within a frame structure is random. This apparently random traffic signal will, in fact, always have a structure. It might be a PCM voice signal, a data signal, a signaling message for controlling network switching, or possibly an ISDN signal or an ATM cell data stream for broadband services. 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lost or delayed and the assembly of packets in the correct order at the destination end. TCP is so widespread that the term TCP/IP is almost as common as IP and is sometimes used synonymously with IP. Although the mechanisms used by TCP are appropriate and successful for data transfer (such as file transfers and e-mail), they are not appropriate for the delivery of voice traffic. Most data traffic is asynchronous and extremely error sensitive. For example, it hardly matters if an e-mail message takes 10 seconds or 30 seconds to reach its destination, but it is critical that every bit is received correctly. On the other hand, voice traffic is synchronous in nature and a little more tolerant of errors. In speech, when someone speaks, the listener should hear it practically immediately, although it is not as critical that every millisecond of speech is heard. Given that IP provides no guarantees regarding the efficient transport of data packets, one wonders why IP would even be considered as a means for transporting voice, particularly with the stringent delay requirements that voice imposes. One also wonders how VoIP can be made to match the quality, reliability, and scalability of traditional networks. After all, if VoIP is to be a successful competitor to traditional telephony technology, then it must meet all the requirements met by traditional telephony, it must offer new and attractive capabilities beyond traditional telephony, and it must do so at a lower cost. (7.25)
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