Mobile Communications/Similarities Between GSM and UMTS - Revision history

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2026-03-16T12:28:03Z

Add German interlanguage link

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	{{Display}}		{{Display}}
			[[de:Mobile_Kommunikation/Gemeinsamkeiten_von_GSM_und_UMTS]]

Guenter at 14:28, 7 February 2023

2023-02-07T14:28:16Z

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	[[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  $\text{GSM voice transmission}$, i.e. the upper branch of the graphic:
			[[File:EN_Mob_T_3_2_S4_v3.png\|right\|frame\|Components of the voice and data communication with GSM\|class=fit]]

	~~Let us first look at the  $\text{GSM voice transmission}$, i.e. the upper branch of the graphic:~~		*The data rate of a PCM speech signal limited to  $4\ \rm kHz$  is obtained by sampling with  $8\ \rm kHz$  and quantizing with  $13\ \rm bit$  to  $104 \ \rm kbit/s$.  With GSM, speech coding is used to extract exactly  $260\ \rm bit$  for each  $T_{\rm R}=20\ \rm ms$  frame.  A bit stream with  $13 \ \rm kbit/s$  is generated.
	*The data rate of a PCM speech signal limited to  $4\ \rm kHz$  is obtained by sampling with  $8\ \rm kHz$  and quantizing with  $13\ \rm bit$  to  $104 \ \rm kbit/s$.  With GSM, speech coding is used to extract exactly  $260\ \rm bit$  for each  $20\ \rm ms$  frame.  A bit stream with  $13 \ \ \rm kbit/s$  is generated.

	*The task of the  $\text{Voice Activity Detection}$  (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>		*The task of the  $\text{Voice Activity Detection}$  (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>

Guenter at 14:05, 7 February 2023

2023-02-07T14:05:32Z

Guenter at 13:56, 7 February 2023

2023-02-07T13:56:19Z

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	With the  "GSM–D" net  $(f_{\rm T} = 900 \ \ \rm MHz)$  the cell radius is specified with maximum  $\text{35 km}$ .  In the  "GSM–E"  net, the maximum radius is only half as large because of the larger carrier frequency  $(1800 \ \ \rm MHz)$.		With the  "GSM–D" net  $(f_{\rm T} = 900 \ \ \rm MHz)$  the cell radius is specified with maximum  $\text{35 km}$ .  In the  "GSM–E"  net, the maximum radius is only half as large because of the larger carrier frequency  $(1800 \ \ \rm MHz)$.

	[[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>		<br><br><br><br><br>
	In the UMTS network  $(f_{\rm T} \approx 2 \ \rm GHz)$  there are different types of radio cells:		In the UMTS network  $(f_{\rm T} \approx 2 \ \rm GHz)$  there are different types of radio cells:

Guenter at 16:50, 6 February 2023

2023-02-06T16:50:18Z

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	{{GraueBox\|TEXT=		{{GraueBox\|TEXT=
	$\text{Example 1:}$  The graphic shows a downlink scenario where a mobile station can receive its signal from two different base stations  $(\rm BS1$  and  $\rm BS2)$  at certain locations.		$\text{Example 1:}$  The graphic shows a downlink scenario where a mobile station can receive its signal from two different base stations  $(\rm BS1$  and  $\rm BS2)$  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  "Hard Handover"  the mobile station at point  $\rm A$  only evaluates the signal from  $\rm BS1$  and at point  $\rm C$  only the signal from  $\rm BS2$.  Switchover is performed immediately when the user is at point  $\rm B$ .		*For  "Hard Handover"  the mobile station at point  $\rm A$  only evaluates the signal from  $\rm BS1$  and at point  $\rm C$  only the signal from  $\rm BS2$.  Switchover is performed immediately when the user is at point  $\rm B$ .

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	[[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  $\text{GSM voice transmission}$, i.e. the upper branch of the graphic:		Let us first look at the  $\text{GSM voice transmission}$, i.e. the upper branch of the graphic:

Guenter at 16:31, 6 February 2023

2023-02-06T16:31:44Z

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	With the  "GSM–D" net  $(f_{\rm T} = 900 \ \ \rm MHz)$  the cell radius is specified with maximum  $\text{35 km}$ .  In the  "GSM–E"  net, the maximum radius is only half as large because of the larger carrier frequency  $(1800 \ \ \rm MHz)$.		With the  "GSM–D" net  $(f_{\rm T} = 900 \ \ \rm MHz)$  the cell radius is specified with maximum  $\text{35 km}$ .  In the  "GSM–E"  net, the maximum radius is only half as large because of the larger carrier frequency  $(1800 \ \ \rm MHz)$.

	[[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>		<br><br><br><br><br>
	In the UMTS network  $(f_{\rm T} \approx 2 \ \rm GHz)$  there are different types of radio cells:		In the UMTS network  $(f_{\rm T} \approx 2 \ \rm GHz)$  there are different types of radio cells:

Hwang at 13:51, 26 January 2023

2023-01-26T13:51:27Z

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	*The task of the  $\text{Voice Activity Detection}$  (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>		*The task of the  $\text{Voice Activity Detection}$  (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>

	*$\text{Channel coding}$  $\rm (CRC$ 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 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>

Hwang at 13:50, 26 January 2023

2023-01-26T13:50:03Z

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

Hwang at 13:47, 26 January 2023

2023-01-26T13:47:08Z

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	*The task of the  $\text{Voice Activity Detection}$  (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>		*The task of the  $\text{Voice Activity Detection}$  (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>

	*$\text{Channel coding}$  $\rm (CRC$ 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 are especially protected.<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>		*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>

Hwang at 13:44, 26 January 2023

2023-01-26T13:44:46Z

← Older revision		Revision as of 15:44, 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>

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	[[File:EN_Mob_T_3_2_S3.png\|right\|frame\|Scenarios for the Near-Far effect\|class=fit]]		[[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  $\rm A$  at the base station is significantly lower than that of the blue user  $\rm B$  due to the path loss (left scenario).  In large macro cells the difference can be up to  $\text{100 dB}$.  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  $\rm A$  at the base station is significantly lower than that of the blue user  $\rm B$  due to the path loss (left scenario).  In large macro cells the difference can be up to  $\text{100 dB}$.  This way the red signal is largely hidden by the blue one.<br>

	*You can largely avoid the Near–Far effect if the more distant user  $\rm A$  transmits with higher power than user  $\rm B$, as indicated in the right scenario.  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–Far effect if the more distant user  $\rm A$  transmits with higher power than user  $\rm B$, as indicated in the right scenario.  At the base station the received power of both mobile stations is then (almost) equal.<br><br>

	<i>Note:</i>   In an idealized system  $($one–way channel, ideal A/D converters, completely linear amplifiers$)$  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.		<i>Note:</i>   In an idealized system  $($one–way channel, ideal A/D converters, completely linear amplifiers$)$  the transmitted data of the users are orthogonal to each other and one could detect the users individually even with very different received powers.

	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>		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:		In reality, however, orthogonality is not always given due to the following reasons:
	*different ~~receive '''KORREKTUR:~~ received~~?'''~~ paths   ⇒   multipath channel,<br>		*different received paths   ⇒   multipath channel,<br>

	*not ideal properties of spreading and scrambling codes with CDMA,<br>		*not ideal properties of spreading and scrambling codes with CDMA,<br>
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	== Various handover strategies ==		== Various handover strategies ==
	<br>		<br>
	A second problem besides the Near–Far effect occurs when a mobile user switches from one cell to another.  In order to make the transition between different cells appear as uninterrupted as possible to the user, a so–called  [[Examples_of_Communication_Systems/UMTS Network Architecture#Handover_in_UMTS\| $\text{Handover}$]]  is used for circuit–switched UMTS services and for GSM.  A distinction is made between two types:		A second problem besides the Near–Far effect occurs when a mobile user switches from one cell to another.  In order to make the transition between different cells appear as uninterrupted as possible to the user, a so–called  [[Examples_of_Communication_Systems/UMTS Network Architecture#Handover_in_UMTS\| »$\text{Handover}$]]«  is used for circuit–switched UMTS services and for GSM.  A distinction is made between two types:
	*»'''Hard Handover'''«:   Here, at a certain point in time the connection is suddenly redirected from the current base station to another base station.<br>		*»'''Hard Handover'''«:   Here, at a certain point in time the connection is suddenly redirected from the current base station to another base station.<br>

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	{{GraueBox\|TEXT=		{{GraueBox\|TEXT=
	$\text{Example 1:}$  The graphic shows a downlink scenario where a mobile station can receive its signal from two different base stations  $(\rm BS1$  and  $\rm BS2)$  at certain locations.		$\text{Example 1:}$  The graphic shows a downlink scenario where a mobile station can receive its signal from two different base stations  $(\rm BS1$  and  $\rm BS2)$  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  "Hard Handover"  the mobile station at point  $\rm A$  only evaluates the signal from  $\rm BS1$  and at point  $\rm C$  only the signal from  $\rm BS2$.  Switchover is performed immediately when the user is at point  $\rm B$~~ ~~.		*For  "Hard Handover"  the mobile station at point  $\rm A$  only evaluates the signal from  $\rm BS1$  and at point  $\rm C$  only the signal from  $\rm BS2$.  Switchover is performed immediately when the user is at point  $\rm B$.

	*If you use  "Soft Handover"  and  "Soft Combining", the mobile station benefits from both signals.		*If you use  "Soft Handover"  and  "Soft Combining", the mobile station benefits from both signals.
	#At any location  $(\rm A, \ B, \ C)$  the ~~reception '''KORREKTUR:~~ received~~?'''~~ power increases.		#At any location  $(\rm A, \ B, \ C)$  the received power increases.
	#This results in a diversity gain depending on the channel SNR,		#This results in a diversity gain depending on the channel SNR,
	#and additionally in a coherence gain of  $\text{3 dB}$.		#and additionally in a coherence gain of  $\text{3 dB}$.
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	*The task of the  $\text{Voice Activity Detection}$  (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>		*The task of the  $\text{Voice Activity Detection}$  (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>

	*$\text{Channel coding}$  $\rm (CRC$ 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 encoder are especially protected.<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>		*The  $\text{Interleaver}$  scrambles the bit sequence of the channel 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>


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	To keep the sampling theorem with a certain tolerance, the audio signal is limited by filtering to the frequency range from  $300\ \rm Hz$  to  $3.4\ \rm kHz$.  The necessary compression by the factor  $8$  is the task of the  »'''Speech coding'''«  (this is a special form of the  [[Information_Theory/General_Description\|$\text{Source coding}$]]), for which several standards were defined in the 1990s:		To keep the sampling theorem with a certain tolerance, the audio signal is limited by filtering to the frequency range from  $300\ \rm Hz$  to  $3.4\ \rm kHz$.  The necessary compression by the factor  $8$  is the task of the  »'''Speech coding'''«  (this is a special form of the  [[Information_Theory/General_Description\|$\text{Source coding}$]]), for which several standards were defined in the 1990s:
	*The  [[Examples_of_Communication_Systems/Speech_Coding#GSM_Full_Rate_Vocoder\| $\text{GSM Full Rate Vocoder}$]]  is based on the three compression methods  [[Examples_of_Communication_Systems/Speech_Coding#Linear_Predictive_Coding\|$\text{LPC}$]] (Linear Predictive Coding),  [[Examples_of_Communication_Systems/Speech_Coding#Long_Term_Prediction\| $\text{LTP}$]]  (Long Term Prediction) and  [[Examples_of_Communication_Systems/Speech_Coding#Regular_Pulse_Excitation_.E2.80.93_RPE_Coding \|$\text{RPE}$]] (Regular Pulse Excitation).  –   From each  $20\ \rm ms$  speech frame,  this coder extracts $74$  parameters with a total range of  $260\ \rm bit$, which results in the data rate  $13\ \rm kbit/s$ .  At the receiver, the speech signal must be synthesized again from these  $260\ \rm bit$ .		*The  [[Examples_of_Communication_Systems/Speech_Coding#GSM_Full_Rate_Vocoder\| »$\text{GSM Full Rate Vocoder}$]]«  is based on the three compression methods  [[Examples_of_Communication_Systems/Speech_Coding#Linear_Predictive_Coding\|$\text{LPC}$]] (Linear Predictive Coding),  [[Examples_of_Communication_Systems/Speech_Coding#Long_Term_Prediction\| $\text{LTP}$]]  (Long Term Prediction) and  [[Examples_of_Communication_Systems/Speech_Coding#Regular_Pulse_Excitation_.E2.80.93_RPE_Coding \|$\text{RPE}$]] (Regular Pulse Excitation).  –   From each  $20\ \rm ms$  speech frame,  this coder extracts $74$  parameters with a total range of  $260\ \rm bit$, which results in the data rate  $13\ \rm kbit/s$ .  At the receiver, the speech signal must be synthesized again from these  $260\ \rm bit$ .

	*The  [[Examples_of_Communication_Systems/Speech_Coding#Half_Rate_Vocoder_and_Enhanced_Full_Rate_Codec\| $\text{GSM Half Rate Vocoder}$]]  was specified in 1994 and offers the possibility to transmit an audio signal bandlimited to  $4\ \rm kHz$  with nearly the same quality in half a traffic channel.  Today, this speech codec plays only a minor role.<br>		*The  [[Examples_of_Communication_Systems/Speech_Coding#Half_Rate_Vocoder_and_Enhanced_Full_Rate_Codec\| »$\text{GSM Half Rate Vocoder}$]]«  was specified in 1994 and offers the possibility to transmit an audio signal bandlimited to  $4\ \rm kHz$  with nearly the same quality in half a traffic channel.  Today, this speech codec plays only a minor role.<br>

	*The  [[Examples_of_Communication_Systems/Speech_Coding#Half_Rate_Vocoder_and_Enhanced_Full_Rate_Codec\| $\text{Enhanced Full Rate Codec}$]]  (short "EFR Codec")  was developed in 1995 for the US–American DCS 1900–system.  It works according to the coding method  "Algebraic Code Excited Linear Prediction"  $\rm (ACELP)$  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  [[Examples_of_Communication_Systems/Speech_Coding#Half_Rate_Vocoder_and_Enhanced_Full_Rate_Codec\| »$\text{Enhanced Full Rate Codec}$]]«  (short "EFR Codec")  was developed in 1995 for the US–American DCS 1900–system.  It works according to the coding method  "Algebraic Code Excited Linear Prediction"  $\rm (ACELP)$  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.