RMA.ppt

New PowerPoint TemplateKnow how AMR works
Understanding of AMR parameters
HW/SW requirements for AMR
* © NOKIA Presentation_Name.PPT / DD-MM-YYYY / Initials
Nokia AMR interaction with other Nokia features
AMR support in Nokia system
Nokia AMR parameter
Nokia AMR KPI
Company Confidential
Hard/Soft Blocking
Hard blocking
The whole radio resource is in use - no more calls can be established
due to lack of free radio timeslots.
Soft blocking
The capacity of individual cells is limited by the level of the interference
rather than the number of TRXs available
Dominates with large reuse factors = Wideband deployment
Is dominating with tight reuse patterns = Narrowband deployments
Assuming no lack of radio resources or HW blocking
Dropped calls due to coverage gaps
Targeted quality level
Two alternative solutions
increased quality
Effective Frequency Load Defined
EFL is a measure of the average frequency utilization in the area Represents how loaded each frequency can be across the system
EFL is proportional to spectral efficiency
EFL is directly proportional to the carried traffic x % higher EFL = x % more carried traffic
Busy hour area level average Erlangs/cell
Total number of frequencies used to carry the traffic
Average number of timeslots/TRX
Effective Frequency Load Explained
EFL is a measure of the average frequency utilization in the area Represents how loaded each frequency can be across the system
Assume 1.2 Mhz (6 x 200 kHz carriers) of hopping frequencies in addition to the BCCH carrier
Assume in each cell 5 simultaneous voice users on the average
In this case the Effective frequency load is ~ 5 Erlangs / 48 timeslots = 10.4%
Time
Frequency
Adaptive Multi-Rate Codec (1/2)
Adaptive Multi-Rate (AMR) codec consists of a family of codecs (source and channel codecs with different trade-off bit-rates) operating in the GSM FR and HR channels modes
The AMR system exploits the channel performance and robustness added by the coding rates by adapting the speech and channel coding rates according to the quality of the radio channel
AMR adapts its error protection level (select its optimum channel mode and codec mode) to the local radio channel and traffic load conditions to deliver the best possible combination of speech quality and system capacity
Codec mode adaptation for AMR is based on received channel quality estimation in both MS and BTS, followed by a decision on the most appropriate speech and channel codec mode to apply at a given time
The basic AMR codec mode sets for MS and BTS are provided by BSC via layer 3 signaling
MS shall support all speech codec modes, although only a set of up to 4 speech codec modes is used during a call
Adaptive Multi Rate Codec (2/2)
GSM FR/EFR channel gross bit-rate is 22.8 kbit/s in GSM FR/EFR: 13 kbit/s speech coding and 9.8 kbit/channel coding (HR channel gross bit rate 11.4 kbit/s)
For AMR case, different codecs use different bit rate to encode speech (source coding). The rest of the gross bit-rate is used for channel protection
Speech Qual
Algorithms Related to AMR
In order to select the codec, MS and infrastructure vendors implement the Link Adaptation algorithm or Codec Mode Adaptation
Additionally, there is another algorithm to change the channel rate between FR and HR codecs, which is called Channel Mode Adaptation
Set of Codecs
Channel Mode Adaptation
Codec Mode Adapt.
Capacity and Coverage Gain
Link level results show very high improvement in the terms of TCH FER when robust AMR modes are used
As high as 6 dB improvement at 1% FER in C/I can be achieved Therefore, high capacity gain can be expected when robust AMR modes are utilized
In addition, increased robustness to channel errors can be utilized in the cell coverage, i.e. lower C/I can be allowed at the cell edge
However, in the mixed traffic case the cell coverage has to be planned according to EFR mobiles
With respect to signaling channels, the retransmissions schemes used by SACCH and FACCH channels maintain the probability of signalling success even for very degraded conditions
Capacity Increase with AMR
Due to robust AMR codec modes, very low TCH FER compared to EFR
In 850 MHz case all mobiles are AMR capable, but this comparison illustrates
the capacity gain AMR provides when it is introduced in a typical network
ONE-LAYER (RF-hopping 2/2, no BCCH included)
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~150% gain relative to EFR
Capacity gain based on the 2% outage of the bad TCH FER samples
Improved BCCH Plan
Since the average C/I found in a cell area can be measurably less than that used in a non-AMR network and still provide comparable quality to EFR, the existing clean BCCH layer can be tightened, potentially releasing frequencies to be used on the non-BCCH layer
This offers improved speech quality and extra capacity for TCH, especially in the narrow band deployment (frequency band less than 5 MHz)
However, if EFR roaming mobiles are to be taken care of, the BCCH will have to be planned accordingly
How to plan networks to ensure the quality for the old EFR mobiles?
One method is to use more aggressive power adjustment for AMR mobiles in order to decrease the average interference level in the network
Therefore, the overall interference decreases in the network (smaller average transmission power) and thus the quality of the existing EFR connections increase
Half-Rate Utilization in AMR Codec
Half-rate is an efficient way to increase capacity in the case of limited number of TRXs per cell
AMR HR codec obtains remarkable better speech quality than previous GSM EFR HR codec
AMR FR obtains better quality than AMR HR only when higher FR modes than 7.4 are used (due to higher number of speech coding bits)
AMR FR 7.4 kbit/s mode and AMR HR 7.4 kbit/s mode have the same speech quality when the C/I is high (error free case)
AMR HR channels can be then used in high C/I conditions without noticeably speech quality loss
In theory for ideal frequency hopping about 11-12 dB C/I is required for AMR HR to obtain the evaluated good speech quality limit (in real networks, depending on the BTS configuration and on FH mode used, it might be necessary 1-4 dB higher)
Based on this, all connections having at least 12 dB C/I could be handed over to HR channel remaining the good speech quality
AMR FR+HR vs. AMR FR Usage
The performance degradation in FER between the usage of AMR FR+HR and AMR FR is equivalent to the quality loss of 0.2 in the MOS between AMR FR and HR codec
The FER performance of AMR FR+HR and AMR FR are about the same
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MOS vs. CIR
A user in good radio conditions perceives the same quality as EFR.
However, a user in bad radio conditions still receives acceptable speech quality while with EFR it would not received satisfactory speech quality.
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Coverage enhancement (>4dB)
Benefits For Operator
Capacity / Coverage Gains
Summary
Speech quality enhancement: AMR maintains good speech quality in the situation where the connection faces low C/I or low signal level
Capacity and coverage gain: Link level simulation results illustrated improvement in terms of TCH FER (up to 5.5dB at 1% FER in C/I)
Signalling channel performance: due to retransmissions schemes used by these channels the probability of signalling success maintain very high even for very degraded conditions
Improved BCCH plan: tighter frequency reuse or better quality with same frequency reuse, potentially releasing frequencies to be used on the non-BCCH layer.
HR utilisation increases the hardware capacity of the cell since two half-rate connections can be allocated to fill only one timeslot.
When compare AMR HR to previous GSM HR codec, it is noticed that AMR HR obtains remarkable better speech quality
Quality loss of ~ 0.2 between AMR HR and FR
New AMR family of codecs tolerates 6 dB higher interference than current GSM EFR codec
Can be directly utilized for higher capacity with Frequency Hopping
Higher interference tolerance
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Full Rate in Clean Speech
Channel and Speech Codecs for AMR
In high-error conditions more bits are used for error correction to obtain error robust coding, while in good transmission conditions a lower amount of bits is needed for sufficient error protection and more bits can therefore be allocated for source coding
Channel mode
Channel codec
4.75 kbit/s
0.10 kbit/s
6.45 kbit/s
0.10 kbit/s
(*) Requires 16 kbit/s TRAU. Therefore it is not seen as a feasible codec mode and will not be supported by Nokia BSS10.
Highest quality
HR utilisation doubles the capacity of the cell
When compare AMR HR to previous GSM HR codec, it is noticed that AMR HR obtains remarkable better speech quality
Link Adaptation in AMR Codec (1/2)
Link Adaptation is the capability of AMR feature to vary the codec used according to the link conditions
Both network, for uplink, and MS, for downlink, measure the radio conditions in each link and take decisions on which codec should be applied to each way AMR codec mode adaptation is done independently in UL and DL
There are two link adaptation (LA) modes; the ETSI specified fast LA and the Nokia proprietary slow LA
slowAmrLaEnabled: if it is set to "N" (default) it is used ETSI fast LA; if it is set to "Y" it used Nokia slow LA
With slow LA, BTS allows in-band codec mode changes only on the SACCH frame interval of 480 ms
Two different types of link adaptation algorithms are defined: Codec Mode Adaptation and Channel Mode Adaptation
AMR codec mode adaptation algorithm adapts the bit-rate partitioning between the speech and channel coding for a given channel mode to track changes in the radio link and to account for specific input conditions (speech signal characteristics, acoustic environmental characteristics, etc.)
AMR channel mode adaptation algorithm allocates a half-rate or full-rate channel according to channel quality and the traffic load on the cell in order to obtain the best balance between quality and capacity
AMR
Codec Mode Adaptation
Codec Mode Adaptation or Link Adaptation (LA) is the algorithm that selects which codec has to be used each moment by the MS (in UL) or by the network (in DL direction).
The basic AMR codec mode sets for MS and BTS are provided by BSC via layer 3 signalling
Both the MS and the network implement their own C/I measurement algorithms
C/I measuremnt algorithms are vendor dependant / proprietary
Nokia has common UL/DL link adaptation thresholds
Objective of Codec Mode Adaptation
Select the codec that provides the best speech quality depending on radio conditions
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Parameters for Link Adaptation
When deploying AMR the following parameters are important for the Link Adaptation:
ACS (Active Codec Set) which defines the codecs that can be used in a BTS during a call.
Thresholds used: Defines the CIR value to change the codec from a less robust codec to the immediate more robust one in the ACS
Hysteresis: the values in dB to add to the thresholds in order to go from a robust codec to the immediate less robust one in the ACS.
For instance: ACS= [AFS12.2, AFS7.9, AFS4.75], Thresholds: 12dB, 8dB, Hysteresis: 1dB, 1dB
With these settings the change from codec AFS7.9 to AFS4.75 will happen when the CIR is below 8dB, while from AFS4.75 up to AFS7.9 it will be with 9dB.
Meas
CIR
norm
(LQE)
Codec mode changed according to channel conditions (UL/DL C/I)
Only up to four codecs can be used during a call
Goal—the highest MOS (Mean Opinion Score)
Mode indications inform the receiver about the currently applied codec mode
Mode Command informs MS about the codec mode to be applied on the uplink
Channel mode adaptation
Based on BTS load (BSC level) and channel condition (RxQual)
Codec modes are constrained to change only every second speech frame
CMR
CMR
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CMC
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CMR
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CMR
DL
UL
Impact of wrong LA
Due to wrong LA threshold selection, or wrong estimation of radio conditions, the codec used under certain conditions might not be the best performing one, reducing Speech Quality.
AFS12.2
AFS7.90
AFS475
Ideal CIR for codec changes
Actual CIR for codec changes for LA 2
non-ideal LA 2 (Lower MOS & low FER)
Actual CIR for codec changes for LA 1
Channel Mode Adaptation
Channel Mode Adaptation is an intra BTS HO algorithm that aims at select the correct channel rate (FR or HR).
The selection of the channel rate depends on 2 main factors: load and quality
load
Channel mode adaptation: Packing
Handover between AMR FR and AMR HR is intra-cell (intra BTS) handover
Spontaneous packing of FR AMR calls to HR AMR calls is triggered when the cell load is high enough, the number of free full rate resources reduces below the value of the parameter btsLoadDepTCHRate (HRL).
Packing continues until the cell load is low enough, the number of free full rate resources increases above the value of the parameter btsLoadDepTCHRate (HRU).
Free FR TCHs
btsLoadDepTCHRate(HRU)
btsLoadDepTCHRate(HRL)
Packing from AMR FR to AMR HR (1/2)
Spontaneous Packing of AMR FR to AMR HR call is triggered when
free full rate resources reduces below the value of the parameter btsLoadDepTCHRate(HRL) or btsSpLoadDepTCHRate (FRL)
HRL is a BSC level parameter
FRL is a BTS level parameter, once defined, it can overwrite HRL
AND FR calls which quality is better than amrHandoverFr(IHRF) for both UL and DL
Px: The Px of Threshold quality DL Px (QDP) is used.
Nx: The Nx of Threshold quality DL Nx (QDN) is used.
Note:
Packing quality threshold is for both UL/DL threshold
Packing quality does not have its own Px/Nx
Packing quality does not have its own averaging windows and weighting
Packing happens to permanent HR channels
Packing happens to DR channels which half has been occupied
Packing happens when there are even number of FR calls to DR channels.
Packing continues until the number of free full rate resources increases above the value of the parameter btsLoadDepTCHRate (HRU) or btsSpLoadDepTCHRate (FRU)
HRU is a BSC level parameter
FRU is a BTS level parameter, once defined, it can overwrite HRU
Packing is triggered by new TCH allocation
Queueing is not allowed for packing procedure
Unpacking from AMR HR to FR
Spontaneous unpacking of AMR HR calls to AMR FR calls is triggered when the quality of a AMR HR call degrades below the amrHandoverHr(IHRH) for either UL or DL
Px: The Px of Threshold quality DL Px (QDP) is used.
Nx: The Nx of Threshold quality DL Nx (QDN) is used.
Queuing is allowed for unpacking procedure
DADL/B
New adjacent cell parameter to handover AMR calls from non-AMR-capable cells to co-located AMR-capable cells during call set-up phase
Handover
Prioritization of AMR capable cells during internal and external handovers (AMR capable cells which load is low (BTS load threshold (BLT) parameter), are on the top of the handover target cell list)
New RxQual thresholds for AMR FR and AMR HR
New RxQual thresholds for HOs between AMR channel rates (relates to AMR FR call packing and AMR HR call unpacking)
IFH and IUO
New good and bad C/I thresholds for AMR FR and AMR HR
Background
3GPP 05.08 states that Radio Link Failure (RLF) in the MS is determined by the success rate of decoding messages on the downlink SACCH
The aim of determining RLF in the MS is to ensure that calls with unacceptable voice/data quality, which cannot be improved either by RF power control or handover, are either re-established or released in a defined manner
The Radio Link Timeout (RLT) parameter controls that a forced release (drop) will not normally occur until the call has degraded to a quality below that at which the majority of subscribers would have manually released it
The RLF procedure is implemented in the RRM at the BSC and is as follows:
After the assignment of a dedicated channel a counter is initialized to RLT
When a SACCH message is unsuccessfully decoded the counter is decreased by 1
When a SACCH message is successfully decoded the counter is increased by 2
If the counter reaches 0 a RLF is declared Call is released
AMR FR vs. EFR - Test 1
The RLT is based on SACCH erased frames, which are independent of speech frames
The tests were aimed to find RLT value producing the same speech degradation in AMR as EFR would suffer with default RLT value for this traffic (i.e. 20)
The indicator used was number of BQS-FER (with FER>50%/25%) between the time when the counter starts decreasing from its top value (64) to the point where the link would be released (i.e., when the counter is decreased by the RLT parameter)
RLT has very high impact on DCR
AMR-FR vs. EFR – Test 2
Aim is to evaluate when AMR-FR is used which RLT value will result in comparable performance (point at which call is released) to the recommended RLT for EFR
The RLT is based on SACCH erased frames, which are independent of speech frames. The principle of the tests is to find RLT value producing the same speech degradation (FER > 15 % MOS < 1.5 no audible speech during 30 sec before dropping) in AMR as EFR would suffer with default RLT value for this traffic (i.e. 20)
The driving route started at a good coverage location and ended at a bad coverage area
AMR vs EFR: FER Before Dropping
FER average every 5 seconds, during last 30 seconds before dropping for: RLT = 20, 28, 32, 36
Dropped Call Experience
Dropped call experience: how long terminal does not receive any audible speech (MOS<1.5) before it drops
During 30 seconds before dropping, FERAMR is lower than FEREFR
FEREFR >…

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