Mobile Communications/Similarities Between GSM and UMTS - Revision history

Guenter at 14:28, 7 February 2023

2023-02-07T14:28:16Z

Guenter at 14:05, 7 February 2023

2023-02-07T14:05:32Z

Guenter at 13:56, 7 February 2023

2023-02-07T13:56:19Z

Guenter at 16:50, 6 February 2023

2023-02-06T16:50:18Z

Guenter at 16:31, 6 February 2023

2023-02-06T16:31:44Z

Hwang at 13:51, 26 January 2023

2023-01-26T13:51:27Z

Hwang at 13:50, 26 January 2023

2023-01-26T13:50:03Z

Hwang at 13:47, 26 January 2023

2023-01-26T13:47:08Z

Hwang at 13:44, 26 January 2023

2023-01-26T13:44:46Z

Hwang at 13:39, 26 January 2023

2023-01-26T13:39:33Z

@@ Line 128: / Line 128: @@
-[[File:EN_Mob_T_3_2_S4_v3.png|center|frame|Components of the voice and data communication with GSM|class=fit]]
+Let us first look at the &nbsp;$\text{GSM voice transmission}$, i.e. the upper branch of the graphic:
-Let us first look at the &nbsp;$\text{GSM voice transmission}$, i.e. the upper branch of the graphic:
+*The data rate of a PCM speech signal limited to&nbsp; $4\ \rm kHz$&nbsp; is obtained by sampling with&nbsp; $8\ \rm kHz$&nbsp; and quantizing with&nbsp; $13\ \rm bit$&nbsp; to&nbsp; $104 \ \rm kbit/s$.&nbsp; With GSM, speech coding is used to extract exactly&nbsp; $260\ \rm bit$&nbsp; for each&nbsp; $T_{\rm R}=20\ \rm ms$&nbsp; frame.&nbsp; A bit stream with&nbsp; $13  \ \rm kbit/s$&nbsp; is generated.
 *The task of the&nbsp; $\text{Voice Activity Detection}$&nbsp; (dotted line) is to decide whether the current voice frame actually contains a speech signal or a voice pause during which the power of the transmit amplifier can be reduced.<br>

@@ Line 54: / Line 54: @@
 [[File:EN_Mob_T_3_2_S3.png|right|frame|Scenarios for the Near-Far effect|class=fit]]
-*If both mobile stations transmit with the same power, the reception '''KORREKTUR: received?''' power of the red user &nbsp;$\rm A$&nbsp; at the base station is significantly lower than that of the blue user &nbsp;$\rm B$&nbsp; due to the path loss (left scenario).&nbsp; In large macro cells the difference can be up to&nbsp; $\text{100 dB}$.&nbsp; This way the red signal is largely hidden by the blue one.<br>
+*If both mobile stations transmit with the same power, the received power of the red user &nbsp;$\rm A$&nbsp; at the base station is significantly lower than that of the blue user &nbsp;$\rm B$&nbsp; due to the path loss (left scenario).&nbsp; In large macro cells the difference can be up to&nbsp; $\text{100 dB}$.&nbsp; This way the red signal is largely hidden by the blue one.<br>
-*You can largely avoid the Near&ndash;Far effect if the more distant user &nbsp;$\rm A$&nbsp; transmits with higher power than user &nbsp;$\rm B$, as indicated in the right scenario.&nbsp; At the base station the reception '''KORREKTUR: received?''' power of both mobile stations is then (almost) equal.<br><br>
+*You can largely avoid the Near&ndash;Far effect if the more distant user &nbsp;$\rm A$&nbsp; transmits with higher power than user &nbsp;$\rm B$, as indicated in the right scenario.&nbsp; At the base station the received power of both mobile stations is then (almost) equal.<br><br>
-<i>Note:</i> &nbsp; In an idealized system&nbsp; $($one&ndash;way channel, ideal A/D converters, completely linear amplifiers$)$&nbsp; the transmitted data of the users are orthogonal to each other and one could detect the users individually even with very different receiving '''KORREKTUR: received?''' powers.&nbsp;
+<i>Note:</i> &nbsp; In an idealized system&nbsp; $($one&ndash;way channel, ideal A/D converters, completely linear amplifiers$)$&nbsp; the transmitted data of the users are orthogonal to each other and one could detect the users individually even with very different received powers.&nbsp;
 This statement is valid for GSM due to the multiple access methods FDMA and TDMA, but also for UMTS (CDMA) and for the 4G system LTE (TDMA/OFDMA).<br>
 In reality, however, orthogonality is not always given due to the following reasons:
-*different receive '''KORREKTUR: received?''' paths &nbsp; &#8658; &nbsp; multipath channel,<br>
+*different received paths &nbsp; &#8658; &nbsp; multipath channel,<br>
 *not ideal properties of spreading and scrambling codes with CDMA,<br>
@@ Line 91: / Line 91: @@
 == Various handover strategies ==
 <br>
-A second problem besides the Near&ndash;Far effect occurs when a mobile user switches from one cell to another.&nbsp; In order to make the transition between different cells appear as uninterrupted as possible to the user, a so&ndash;called&nbsp; [[Examples_of_Communication_Systems/UMTS Network Architecture#Handover_in_UMTS| $\text{Handover}$]]&nbsp; is used for circuit&ndash;switched UMTS services and for GSM.&nbsp; A distinction is made between two types:
+A second problem besides the Near&ndash;Far effect occurs when a mobile user switches from one cell to another.&nbsp; In order to make the transition between different cells appear as uninterrupted as possible to the user, a so&ndash;called&nbsp; [[Examples_of_Communication_Systems/UMTS Network Architecture#Handover_in_UMTS| &raquo;$\text{Handover}$]]&laquo;&nbsp; is used for circuit&ndash;switched UMTS services and for GSM.&nbsp; A distinction is made between two types:
 *&raquo;'''Hard Handover'''&laquo;: &nbsp; Here, at a certain point in time the connection is suddenly redirected from the current base station to another base station.<br>
@@ Line 98: / Line 98: @@
 {{GraueBox|TEXT=
 $\text{Example 1:}$&nbsp; The graphic shows a downlink scenario where a mobile station can receive its signal from two different base stations&nbsp; $(\rm BS1$&nbsp; and&nbsp; $\rm BS2)$&nbsp; at certain locations.
-[[File:EN_Mob_T_3_2_S3b_v2.png|right|frame|Handover scenarios for the downlink '''KORREKTUR: scenarios'''|class=fit]]
+[[File:EN_Mob_T_3_2_S3b_v2.png|right|frame|Handover scenarios for the downlink|class=fit]]
-*For&nbsp; "Hard Handover"&nbsp; the mobile station  at point&nbsp; $\rm A$&nbsp; only evaluates the signal from&nbsp; $\rm BS1$&nbsp; and at point&nbsp; $\rm C$&nbsp; only the signal from&nbsp; $\rm BS2$.&nbsp; Switchover is performed immediately when the user is at point&nbsp; $\rm B$&nbsp;.
+*For&nbsp; "Hard Handover"&nbsp; the mobile station  at point&nbsp; $\rm A$&nbsp; only evaluates the signal from&nbsp; $\rm BS1$&nbsp; and at point&nbsp; $\rm C$&nbsp; only the signal from&nbsp; $\rm BS2$.&nbsp; Switchover is performed immediately when the user is at point&nbsp; $\rm B$.
 *If you use&nbsp; "Soft Handover"&nbsp; and&nbsp; "Soft Combining", the mobile station benefits from both signals.
-#At any location&nbsp; $(\rm A, \ B, \ C)$&nbsp; the reception '''KORREKTUR: received?''' power increases.&nbsp;
+#At any location&nbsp; $(\rm A, \ B, \ C)$&nbsp; the received power increases.&nbsp;
 #This results in a diversity gain depending on the channel SNR,
 #and additionally in a coherence gain of&nbsp; $\text{3 dB}$.
@@ Line 135: / Line 135: @@
 *The task of the&nbsp; $\text{Voice Activity Detection}$&nbsp; (dotted line) is to decide whether the current voice frame actually contains a speech signal or a voice pause during which the power of the transmit amplifier can be reduced.<br>
-*$\text{Channel coding}$&nbsp; $\rm (CRC$ and convolutional code$)$&nbsp; is used to add redundancy to enable error correction at the receiver.&nbsp; This increases the data rate to&nbsp; $22.8 \ \rm kbit/s$, whereby the more important bits of the speech coder '''KORREKTUR: encoder?''' are especially protected.<br>
+*$\text{Channel coding}$&nbsp; $\rm (CRC$ and convolutional code$)$&nbsp; is used to add redundancy to enable error correction at the receiver.&nbsp; This increases the data rate to&nbsp; $22.8 \ \rm kbit/s$, whereby the more important bits of the speech encoder  are especially protected.<br>
-*The&nbsp; $\text{Interleaver}$&nbsp; scrambles the bit sequence of the channel coder '''KORREKTUR: encoder?''' and thus reduces the influence of burst errors.&nbsp; For this purpose, the&nbsp; $456$&nbsp; input bits are divided and interleaved into four time frames of&nbsp; $114\ \rm bit$&nbsp; each.&nbsp; Successive bits are always transmitted in eight different bursts.<br>
+*The&nbsp; $\text{Interleaver}$&nbsp; scrambles the bit sequence of the channel encoder ' and thus reduces the influence of burst errors.&nbsp; For this purpose, the&nbsp; $456$&nbsp; input bits are divided and interleaved into four time frames of&nbsp; $114\ \rm bit$&nbsp; each.&nbsp; Successive bits are always transmitted in eight different bursts.<br>
@@ Line 180: / Line 180: @@
 To keep the sampling theorem with a certain tolerance, the audio signal is limited by filtering to the frequency range from&nbsp; $300\ \rm Hz$&nbsp; to&nbsp; $3.4\ \rm kHz$.&nbsp; The necessary compression by the factor&nbsp; $8$&nbsp; is the task of the&nbsp; &raquo;'''Speech coding'''&laquo;&nbsp; (this is a special form of the&nbsp; [[Information_Theory/General_Description|$\text{Source coding}$]]), for which several standards were defined in the 1990s:
-*The&nbsp; [[Examples_of_Communication_Systems/Speech_Coding#GSM_Full_Rate_Vocoder| $\text{GSM Full Rate Vocoder}$]]&nbsp;  is based on the three compression methods&nbsp; [[Examples_of_Communication_Systems/Speech_Coding#Linear_Predictive_Coding|$\text{LPC}$]] (Linear Predictive Coding),&nbsp; [[Examples_of_Communication_Systems/Speech_Coding#Long_Term_Prediction| $\text{LTP}$]]&nbsp; (Long Term Prediction) and&nbsp; [[Examples_of_Communication_Systems/Speech_Coding#Regular_Pulse_Excitation_.E2.80.93_RPE_Coding |$\text{RPE}$]] (Regular Pulse Excitation).&nbsp; &ndash; &nbsp; From each&nbsp; $20\ \rm ms$&nbsp; speech frame,&nbsp; this coder extracts $74$&nbsp; parameters with a total range of&nbsp; $260\ \rm bit$, which results in the data rate&nbsp; $13\ \rm kbit/s$&nbsp;.&nbsp; At the receiver, the speech signal must be synthesized again from these&nbsp; $260\ \rm bit$&nbsp;.
+*The&nbsp; [[Examples_of_Communication_Systems/Speech_Coding#GSM_Full_Rate_Vocoder| &raquo;$\text{GSM Full Rate Vocoder}$]]&laquo;&nbsp;  is based on the three compression methods&nbsp; [[Examples_of_Communication_Systems/Speech_Coding#Linear_Predictive_Coding|$\text{LPC}$]] (Linear Predictive Coding),&nbsp; [[Examples_of_Communication_Systems/Speech_Coding#Long_Term_Prediction| $\text{LTP}$]]&nbsp; (Long Term Prediction) and&nbsp; [[Examples_of_Communication_Systems/Speech_Coding#Regular_Pulse_Excitation_.E2.80.93_RPE_Coding |$\text{RPE}$]] (Regular Pulse Excitation).&nbsp; &ndash; &nbsp; From each&nbsp; $20\ \rm ms$&nbsp; speech frame,&nbsp; this coder extracts $74$&nbsp; parameters with a total range of&nbsp; $260\ \rm bit$, which results in the data rate&nbsp; $13\ \rm kbit/s$&nbsp;.&nbsp; At the receiver, the speech signal must be synthesized again from these&nbsp; $260\ \rm bit$&nbsp;.
-*The&nbsp; [[Examples_of_Communication_Systems/Speech_Coding#Half_Rate_Vocoder_and_Enhanced_Full_Rate_Codec| $\text{GSM Half Rate Vocoder}$]]&nbsp;  was specified in 1994 and offers the possibility to transmit an audio signal bandlimited to&nbsp; $4\ \rm kHz$&nbsp; with nearly the same quality in half a traffic channel.&nbsp; Today, this speech codec plays only a minor role.<br>
+*The&nbsp; [[Examples_of_Communication_Systems/Speech_Coding#Half_Rate_Vocoder_and_Enhanced_Full_Rate_Codec| &raquo;$\text{GSM Half Rate Vocoder}$]]&laquo;&nbsp;  was specified in 1994 and offers the possibility to transmit an audio signal bandlimited to&nbsp; $4\ \rm kHz$&nbsp; with nearly the same quality in half a traffic channel.&nbsp; Today, this speech codec plays only a minor role.<br>
-*The&nbsp; [[Examples_of_Communication_Systems/Speech_Coding#Half_Rate_Vocoder_and_Enhanced_Full_Rate_Codec| $\text{Enhanced Full Rate Codec}$]]&nbsp; (short "EFR Codec")&nbsp; was developed in 1995 for the US&ndash;American DCS 1900&ndash;system.&nbsp; It works according to the coding method&nbsp; "Algebraic Code Excited Linear Prediction"&nbsp; $\rm (ACELP)$&nbsp; and offers a significantly higher speech quality compared to the conventional full rate codec due to the ACELP principle and because of the improved error detection and obfuscation.
+*The&nbsp; [[Examples_of_Communication_Systems/Speech_Coding#Half_Rate_Vocoder_and_Enhanced_Full_Rate_Codec| &raquo;$\text{Enhanced Full Rate Codec}$]]&laquo;&nbsp; (short "EFR Codec")&nbsp; was developed in 1995 for the US&ndash;American DCS 1900&ndash;system.&nbsp; It works according to the coding method&nbsp; "Algebraic Code Excited Linear Prediction"&nbsp; $\rm (ACELP)$&nbsp; and offers a significantly higher speech quality compared to the conventional full rate codec due to the ACELP principle and because of the improved error detection and obfuscation.

@@ Line 17: / Line 17: @@
 With the&nbsp; "GSM&ndash;D" net&nbsp; $(f_{\rm T} = 900 \ \ \rm MHz)$&nbsp; the cell radius is specified with maximum &nbsp;$\text{35 km}$&nbsp;.&nbsp; In the&nbsp; "GSM&ndash;E"&nbsp; net, the maximum radius is only half as large because of the larger carrier frequency &nbsp;$(1800 \ \ \rm MHz)$.&nbsp;
-[[File:EN_Bei_T_4_2_S8.png|80px|left|frame|Cell structure in UMTS '''KORREKTUR'''|class=fit]]
+[[File:EN_Bei_T_4_2_S8.png|80px|left|frame|Cell structure in UMTS|class=fit]]
 <br><br><br><br><br>
 In the UMTS network&nbsp; $(f_{\rm T} \approx 2 \ \rm GHz)$&nbsp; there are different types of radio cells:

@@ Line 98: / Line 98: @@
 {{GraueBox|TEXT=
 $\text{Example 1:}$&nbsp; The graphic shows a downlink scenario where a mobile station can receive its signal from two different base stations&nbsp; $(\rm BS1$&nbsp; and&nbsp; $\rm BS2)$&nbsp; at certain locations.
-[[File:EN_Mob_T_3_2_S3b.png|right|frame|Handover scenarios for the downlink '''KORREKTUR: scenarios'''|class=fit]]
+[[File:EN_Mob_T_3_2_S3b_v2.png|right|frame|Handover scenarios for the downlink '''KORREKTUR: scenarios'''|class=fit]]
 *For&nbsp; "Hard Handover"&nbsp; the mobile station  at point&nbsp; $\rm A$&nbsp; only evaluates the signal from&nbsp; $\rm BS1$&nbsp; and at point&nbsp; $\rm C$&nbsp; only the signal from&nbsp; $\rm BS2$.&nbsp; Switchover is performed immediately when the user is at point&nbsp; $\rm B$&nbsp;.
@@ Line 128: / Line 128: @@
-[[File:EN_Mob_T_3_2_S4.png|center|frame|Components of the voice and data communication with GSM|class=fit]]
+[[File:EN_Mob_T_3_2_S4_v3.png|center|frame|Components of the voice and data communication with GSM|class=fit]]
 Let us first look at the &nbsp;$\text{GSM voice transmission}$, i.e. the upper branch of the graphic:

@@ Line 17: / Line 17: @@
 With the&nbsp; "GSM&ndash;D" net&nbsp; $(f_{\rm T} = 900 \ \ \rm MHz)$&nbsp; the cell radius is specified with maximum &nbsp;$\text{35 km}$&nbsp;.&nbsp; In the&nbsp; "GSM&ndash;E"&nbsp; net, the maximum radius is only half as large because of the larger carrier frequency &nbsp;$(1800 \ \ \rm MHz)$.&nbsp;
-[[File:P_ID1520__Bei_T_4_2_S8a_60.png|80px|left|frame|Cell structure in UMTS '''KORREKTUR'''|class=fit]]
+[[File:EN_Bei_T_4_2_S8.png|80px|left|frame|Cell structure in UMTS '''KORREKTUR'''|class=fit]]
 <br><br><br><br><br>
 In the UMTS network&nbsp; $(f_{\rm T} \approx 2 \ \rm GHz)$&nbsp; there are different types of radio cells:

@@ Line 154: / Line 154: @@
 An essential&nbsp; $\text{difference between GSM and UMTS}$&nbsp; are the different modulation and multiple access methods:
 *For GSM: &nbsp; "Gaussian Minimum Shift Keying"&nbsp; $\rm (GMSK)$&nbsp; together with&nbsp; $\rm FDMA$&nbsp;  and&nbsp; $\rm TDMA$,
-*for UMTS:&nbsp; "Quaternary Phase Shift Keying"&nbsp; $\rm (QPSK)$&nbsp; together withwith&nbsp; $\rm CDMA$&nbsp;  and&nbsp; $\rm TDMA$.}}
+*for UMTS:&nbsp; "Quaternary Phase Shift Keying"&nbsp; $\rm (QPSK)$&nbsp; together with&nbsp; $\rm CDMA$&nbsp;  and&nbsp; $\rm TDMA$.}}

← Older revision		Revision as of 13:44, 26 January 2023
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	*$\text{Channel coding}$  $\rm (CRC$ and convolutional code$)$  is used to add redundancy to enable error correction at the receiver.  This increases the data rate to  $22.8 \ \rm kbit/s$, whereby the more important bits of the speech coder '''KORREKTUR: encoder?''' are especially protected.<br>		*$\text{Channel coding}$  $\rm (CRC$ and convolutional code$)$  is used to add redundancy to enable error correction at the receiver.  This increases the data rate to  $22.8 \ \rm kbit/s$, whereby the more important bits of the speech coder '''KORREKTUR: encoder?''' are especially protected.<br>

−	*The  $\text{Interleaver}$  scrambles the bit sequence of the channel coder and thus reduces the influence of burst errors.  For this purpose, the  $456$  input bits are divided and interleaved into four time frames of  $114\ \rm bit$  each.  Successive bits are always transmitted in eight different bursts.<br>	+	*The  $\text{Interleaver}$  scrambles the bit sequence of the channel coder '''KORREKTUR: encoder?''' and thus reduces the influence of burst errors.  For this purpose, the  $456$  input bits are divided and interleaved into four time frames of  $114\ \rm bit$  each.  Successive bits are always transmitted in eight different bursts.<br>