03.503GPPTransmission planning aspects of the speech service in the GSM Public Land Mobile Network (PLMN) systemTS
The symbol definitions for the calculations in this clause are:
Trftx: The time required for transmission of a TCH radio interface frame over the air interface due to the interleaving and de‑interleaving (system dependent).
Ttransc: The speech encoder processing time, from input of the last PCM sample to output of the final encoded bit (implementation dependent).
Tsample: The duration of the segment of PCM speech operated on by the speech transcoder.
Tencode: The time required for the channel encoder to perform channel encoding (implementation dependent).
Trxproc: The time required after reception over the air interface to perform equalization, channel decoding and SID frame detection (implementation dependent).
Tproc: The time required after reception of the first gain vector (GSPO) to process the speech encoded data for the half rate speech decoder and to produce the first PCM output sample (implementation dependent).
Ta/d: Delay in the analogue to digital converter in the downlink.
Tmargin: An allowance for system entities that are implementation dependent.
Td/a: Delay in the digital to analogue converter in the uplink.
Tpcm: The duration of a segment of PCM speech for the downlink processing delay.
The half rate MS speech delay in the uplink direction is the delay between an acoustic event at the MRP to the last bit of the corresponding speech frame at the antenna connector and shall not exceed:
MS uplink delay = Ta/d + Tmargin + Tsample + Ttransc + Tencode + Trftx
= 1,0 + 1,9 + 24,4 + 12,1 + 1,2 + 32,9 ms
= 73,5 ms
The half rate MS speech delay in the downlink direction is the delay between the first bit of a speech frame at the antenna connector and the last acoustic event at the ERP corresponding to that speech frame and shall not exceed:
MS downlink delay = Tpcm + Trftx + Trxproc + T proc + Tmargin + Td/a
= 24,4 + 32,9 + 8,8 + 1,9 + 1,9 + 0,5 ms
= 70,4 ms
The round trip delay shall therefore not exceed:
143,9 ms (see clause 126.96.36.199).
Annex E (informative):
Adaptive gain control
In noisy surroundings the speaker automatically raises his voice to compensate for the noise. The sending sensitivity can therefore be reduced, giving the same voice output to the line as under quiet conditions and at the same time reducing the output line noise level.
On the other hand the receiving sensitivity has to be increased under noisy conditions to give a better perceived sound quality.
The Sending and receiving sensitivities may be modified automatically by implementing an Adaptive gain control. The gain variation in the set corresponds to a gain in the receiving path and to a symmetrical attenuation in the sending path for increased ambient noise level.
The following table presents, for guidance and illustration only, three examples of gain variation characteristics.
Table E.1: Gain variation characteristics
Ambient noise level
Relative Gain variation
between –44 and –39 dBPa(A)
from –64 to –44 dBPa(A)
from –44 to –24 dBPa(A)
Annex F (informative):
Change Request History
New specification for handsfree performance
Noise reduction Specification for the GSM MS.
Inclusion of the possibility to use a type 3.2 leaky ear for handset tests.
Update of references
CR to 03.50 Correction of wrong reference to ITU-T Recommendation
Removal of inconsistency in RLR for Handsetand Headset MS
Correction of reference in Section 1.1
Release 1997 version
Acoustic testing of MS over the air interface
Upgrade to Release 1998 version 7.0.0
Update to version 7.0.1 for Publication
Release 1999 version 8.0.0
Addition of a new optional artificial ear Type for acoustic tests
Update to Version 8.1.1 for Publication