Page 29 - From GMS to LTE
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Global System for Mobile Communications (GSM)  15

               Speech is converted by a  Bandwidth-limited  Pulse-amplitude-  Digitized
               microphone to an analog     signal         modulated signal   speech signal
                     signal                                every 125s       at 64 kbit/s


                               Band-pass           Sampler           Quantitizer
                                 filter
                            300 Hz – 3.4 kHz   Sample frequency: 8000 Hz  13-segment curve
                                                                   256 values, 8 bit
               Figure 1.11  Digitization of an analog voice signal.


               areas. For small input‐signal amplitudes, a much higher range of digital values is used
               than for high‐amplitude values. The resulting digital data stream is called a pulse code‐
               modulated (PCM) signal. Which volume is represented by which digital 8‐bit value is
               described in the A‐law standard for European networks and in the μ‐law standard in
               North America.
                The use of different standards unfortunately complicates voice calls between networks
               using different standards. Therefore, it is necessary, for example, to convert a voice
                 signal for a connection between France and the United States.
                As the MSC controls all connections, it is also responsible for billing. This is done
               by creating a billing record for each call, which is later transferred to a billing server.
               The billing record contains information like the number of the caller and the calling
               party, cell ID of the cell from which the call originated, time of call origination, dura-
               tion of the call, and so on. Calls for prepaid subscribers are treated differently as the
               charging is already done while the call is running. The prepaid billing service is usually
               implemented on an IN system and not on the MSC, as further described in Section 1.11.

               MSC‐Server and Media Gateway
               In most of today’s mobile voice networks, circuit‐switched components have been
               replaced with IP‐based devices. The MSC has been split into an MSC‐Server (MSC‐S)
               and an MGW. This is shown in Figure 1.10 and has been specified in 3GPP TS 23.205
               [4]. The MSC‐Ss are responsible for CC and MM (signaling), and the MGWs handle the
               transmission of virtual voice circuits (user data).
                To establish a voice connection, MSC‐Ss and MGWs communicate over the Mc
               interface. This interface does not exist in the classical model, as the MSC contained
               both components. 3GPP TS 29.232 [5] describes this interface on which the H.248/
               MEGACO (Media Gateway Control) protocol is used [6]. The protocol is used, for
               example, to establish voice channels to two parties and then to logically connect the
               two channels in the MGW. The protocol is also used to instruct the MGWs to play
               announcements to inform users of events, for example, where the called party is cur-
               rently not available or is busy, and to establish conference calls between more than
               two subscribers. To add redundancy and for load‐balancing reasons, several MSC‐Ss
               and MGWs can be interconnected in a mesh. If an MSC‐S fails, an MGW can thus
               still continue to operate, and is then controlled by another server. Thus, a single
               MSC‐S is no longer solely responsible for a single geographical area as was the case in
               the traditional model.
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