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25.446 | MBMS synchronisation protocol (SYNC) | TS | 18.0.0 | R3 | https://www.3gpp.org/ftp/Specs/archive/25_series/25.446/25446-i00.zip | The present document specifies the MBMS Synchronisation Protocol. For the release of this specification it is used on Iu towards UTRAN and M1 towards E-UTRAN. |
25.450 | UTRAN Iupc interface general aspects and principles | TS | 18.0.0 | R3 | https://www.3gpp.org/ftp/Specs/archive/25_series/25.450/25450-i00.zip | The present document is an introduction to the TSG RAN TS 25.45z series of UMTS Technical Specifications that define the Iupc Interface. The Iupc interface is a logical interface for the interconnection of Stand-Alone SMLC (SAS) and Radio Network Controller (RNC) components of the Universal Terrestrial Radio Access Network (UTRAN) for the UMTS system. |
25.451 | UTRAN Iupc interface layer 1 | TS | 18.0.0 | R3 | https://www.3gpp.org/ftp/Specs/archive/25_series/25.451/25451-i00.zip | The present document specifies the standards allowed to implement Layer 1 on the Iupc interface.
The specification of transmission delay requirements and O&M requirements is not in the scope of the present document.
In the following "Layer 1" and "Physical Layer" are assumed to be synonymous. |
25.452 | UTRAN Iupc interface: signalling transport | TS | 18.0.0 | R3 | https://www.3gpp.org/ftp/Specs/archive/25_series/25.452/25452-i00.zip | The present document specifies the signalling transport related to PCAP signalling to be used across the Iupc interface. The Iupc interface is a logical interface for the interconnection of Stand-Alone SMLC (SAS) and Radio Network Controller (RNC) components of the Universal Terrestrial Radio Access Network (UTRAN) for the UMTS system. The radio network control signalling between these nodes is based upon the Position Calculation Application Part (PCAP). |
25.453 | UTRAN Iupc interface Positioning Calculation Application Part (PCAP) signalling | TS | 18.0.0 | R3 | https://www.3gpp.org/ftp/Specs/archive/25_series/25.453/25453-i00.zip | The present document specifies the Positioning Calculation Application Part (PCAP) between the Radio Network Controller (RNC) and the Stand-Alone SMLC (SAS). It fulfills the RNC-SAS communication requirements specified in TS 25.305 [6] and thus defines the Iupc interface and its associated signaling procedures. |
25.460 | UTRAN Iuant interface: General aspects and principles | TS | 18.0.0 | R3 | https://www.3gpp.org/ftp/Specs/archive/25_series/25.460/25460-i00.zip | See TS 37.460 [8]. |
25.461 | UTRAN Iuant interface: Layer 1 | TS | 18.0.0 | R3 | https://www.3gpp.org/ftp/Specs/archive/25_series/25.461/25461-i00.zip | See TS 37.461 [7]. |
25.462 | UTRAN Iuant interface: Signalling transport | TS | 18.0.0 | R3 | https://www.3gpp.org/ftp/Specs/archive/25_series/25.462/25462-i00.zip | See TS 37.462 [5]. |
25.466 | UTRAN Iuant interface: Application part | TS | 18.0.0 | R3 | https://www.3gpp.org/ftp/Specs/archive/25_series/25.466/25466-i00.zip | See TS 37.466 [5]. |
25.467 | UTRAN architecture for 3G Home Node B (HNB); Stage 2 | TS | 18.0.0 | R3 | https://www.3gpp.org/ftp/Specs/archive/25_series/25.467/25467-i00.zip | The present document specifies the UTRAN architecture for 3G Home Node B (HNB).
It covers specification of the functions for UEs not supporting Closed Subscriber Groups (CSG) and UEs supporting CSGs. It also covers HNB specific requirements for O&M. |
25.468 | UTRAN Iuh Interface RANAP User Adaption (RUA) signalling | TS | 18.0.0 | R3 | https://www.3gpp.org/ftp/Specs/archive/25_series/25.468/25468-i00.zip | The present document specifies the RANAP User Adaption (RUA) between the Home Node B (HNB) and the Home Node B Gateway (HNB-GW). It fulfils the HNB- HNB-GW communication requirements specified in TS 25.467 [3] and is defined over the Iuh – reference point. It provides transparent transport for RANAP messages. |
25.469 | UTRAN Iuh interface Home Node B (HNB) Application Part (HNBAP) signalling | TS | 18.0.0 | R3 | https://www.3gpp.org/ftp/Specs/archive/25_series/25.469/25469-i00.zip | The present document specifies the Home Node B Application Part (HNBAP) between the Home Node B (HNB) and the Home Node B Gateway (HNB-GW). It fulfils the HNB- HNB-GW communication requirements specified in TS 25.467 [3] and is defined over the Iuh – reference point. It provides control and management procedures between HNB and HNB-GW. |
25.470 | UTRAN Iuh Interface PCAP User Adaption (PUA) signalling | TS | 18.0.0 | R3 | https://www.3gpp.org/ftp/Specs/archive/25_series/25.470/25470-i00.zip | The present document specifies the PCAP User Adaption (PUA) between the Home Node B (HNB) and the Home Node B Gateway (HNB-GW). It fulfils the HNB- HNB-GW communication requirements specified in TS 25.467 [5] and is defined over the Iuh – reference point. It provides transport for PCAP messages. |
25.471 | UTRAN Iurh interface Radio Network Subsystem Application Part (RNSAP) User Adaption (RNA) signalling | TS | 18.0.0 | R3 | https://www.3gpp.org/ftp/Specs/archive/25_series/25.471/25471-i00.zip | The present document specifies the RNSAP User Adaption (RNA) supporting Iurh-connectivity between HNBs as specified in TS 25.467 [3] by adapting the services made available by the Iurh signalling transport layer to the needs of RNSAP. It provides transparent transport for RNSAP messages in connection-oriented and connectionless mode and an Iurh setup function. |
25.484 | Automatic Neighbour Relation (ANR) for UTRAN; Stage 2 | TS | 18.0.0 | R3 | https://www.3gpp.org/ftp/Specs/archive/25_series/25.484/25484-i00.zip | The present document is a technical specification of the overall support of Automatic Neighbour Relation (ANR) Function in UTRA. |
25.903 | Continuous connectivity for packet data users | TR | 18.0.0 | RP | https://www.3gpp.org/ftp/Specs/archive/25_series/25.903/25903-i00.zip | The present document summarizes the work done under the WI "Continuous Connectivity for Packet Data Users" defined in [1] by listing technical concepts addressing the objectives of the work item (see below), analysing these technical concepts and selecting the best solution (which might be a combination of technical concepts).
“The objective of this work item is to reduce the uplink noise rise from physical control channels of packet data users, e.g. for users which have temporarily no data transmission.
This is intended to significantly increase the number of packet data users (i.e. HS-DSCH/E-DCH users without UL DPDCH) in the UMTS FDD system that can stay in CELL_DCH state over a long time period, without degrading cell throughput, and that can restart transmission after a period of inactivity with a much shorter delay (<50ms) than would be necessary for reestablishment of a new connection
The objective covers also schemes which could allow improving the achievable UL capacity for VoIP users with its inherent periodic transmission through reducing the overhead of the control channels.
Mobility and downlink transmission should not be impacted for these users.”
The present document provides the base for the following preparation of change requests to the corresponding RAN specifications. |
25.906 | Dynamically reconfiguring a Frequency Division Duplex (FDD) User Equipment (UE) receiver to reduce power consumption when desired Quality of Service (QoS) is met | TR | 18.0.0 | R4 | https://www.3gpp.org/ftp/Specs/archive/25_series/25.906/25906-i00.zip | The objectives of this study are:
a) RAN4 to identify whether there are situations in which individual UE receiver performance reduction has no, or minimal impact to the overall UTRAN system level performance or user experience. RAN4 should also identify scenarios in which UE receiver performance reduction cannot safely be performed.
b) RAN4 to investigate scenarios for the identified situations where the UE could reduce its performance. The purpose of these scenarios is to ensure that UE performance is not degraded when conditions are not suitable.
c) RAN2 to investigate additional signalling which may be beneficial to support Ues in the decision making process for reducing their performance, for example quality thresholds which assist the UE in determining that conditions are suitable to reduce receiver performance. |
25.912 | Feasibility study for evolved Universal Terrestrial Radio Access (UTRA) and Universal Terrestrial Radio Access Network (UTRAN) | TR | 18.0.0 | RP | https://www.3gpp.org/ftp/Specs/archive/25_series/25.912/25912-i00.zip | This present document is the technical report for the study item "Evolved UTRA and UTRAN" [1]. The objective of the study item is to develop a framework for the evolution of the 3GPP radio-access technology towards a high-data-rate, low-latency and packet-optimized radio access technology. |
25.914 | Measurements of radio performances for UMTS terminals in speech mode | TR | 18.0.0 | R4 | https://www.3gpp.org/ftp/Specs/archive/25_series/25.914/25914-i00.zip | The present document describes the methods to be used in order to assess the radio performances of the 3G user equipment/mobile stations (UE/MS) in active mode in both the up- and the downlink. The test procedure is based on the test method developed as a result of COST 273 Sub-Working Group (SWG) 2.2 members' contributions and the first draft was published in [1]. Background work has also been made in the former COST259 project [2] [3]. |
25.929 | Continuous connectivity for packet data users; 1.28 Mcps TDD | TR | 18.0.0 | RP | https://www.3gpp.org/ftp/Specs/archive/25_series/25.929/25929-i00.zip | The present document summarizes the work done under the WI "Continuous Connectivity for Packet Data Users for 1.28Mcps TDD" defined in [1] by listing technical concepts addressing the objectives of the work item (see below), analysing these technical concepts and selecting the best solution (which might be a combination of technical concepts).
The objective of this work item is to reduce the code consumption (e.g. overhead of physical control channels or related signaling messages) of packet data users for both real-time (e.g. VoIP) and non real-time services, e.g. for users which have temporarily no data transmission in either uplink or downlink. Packet data users as considered in this work item are using only HS-DSCH/E-DCH channels without UL DPCH and DL DPCH.
The aim is to increase the number of packet data users in the UMTS 1.28Mcps TDD system that can be kept efficiently in CELL_DCH state over a longer time period and that can restart transmission after a period of temporary inactivity with a much shorter delay (for example, <100ms) than would be necessary for reestablishment of a new connection.
Another aim is to reduce UE power consumption in CELL_DCH state over a long period by DTX and DRX.
The present document provides the base for the following preparation of change requests to the corresponding RAN specifications. |
25.931 | UTRAN functions, examples on signalling procedures | TR | 18.0.0 | R3 | https://www.3gpp.org/ftp/Specs/archive/25_series/25.931/25931-i00.zip | The present document describes the UTRAN functions by means of signalling procedure examples (Message Sequence Charts). The signalling procedure examples show the interaction between the UE, the different UTRAN nodes and the CN to perform system functions. This gives an overall understanding of how the UTRAN works in example scenarios. |
25.942 | Radio Frequency (RF) system scenarios | TR | 18.0.0 | R4 | https://www.3gpp.org/ftp/Specs/archive/25_series/25.942/25942-i00.zip | During the UTRA standards development, the physical layer parameters will be decided using system scenarios, together with implementation issues, reflecting the environments that UTRA will be designed to operate in. |
25.943 | Deployment aspects | TR | 18.0.0 | R4 | https://www.3gpp.org/ftp/Specs/archive/25_series/25.943/25943-i00.zip | The present document establishes channel models to be used for deployment evaluation. |
25.951 | FDD Base Station (BS) classification | TR | 18.0.0 | R4 | https://www.3gpp.org/ftp/Specs/archive/25_series/25.951/25951-i00.zip | This document is a Technical Report on Release 6 work item “FDD Base Station Classification”. |
25.956 | Universal Terrestrial Radio Access (UTRA) repeater planning guidelines and system analysis | TR | 18.0.0 | R4 | https://www.3gpp.org/ftp/Specs/archive/25_series/25.956/25956-i00.zip | The purpose of the following document is to describe planning guidelines and system scenarios for UTRA repeaters. In addition it also contains simulations and analysis of the usage of repeaters in UMTS networks. |
25.963 | Feasibility study on interference cancellation for UTRA FDD User Equipment (UE) | TR | 18.0.0 | R4 | https://www.3gpp.org/ftp/Specs/archive/25_series/25.963/25963-i00.zip | The objective of this study is to evaluate the feasibility and potential performance improvements of interference cancellation/mitigation techniques for UTRA FDD UE receivers, based on realistic network scenarios. Scope of the work includes:
- Determine realistic network scenarios.
- Determine suitable interference models for 'other cell' interference.
- Evaluate the feasibility of two-branch interference cancellation receivers through link and system level analysis and simulations.
- Evaluate feasibility of one-branch interference cancellation receivers through link and system level analysis and simulations. |
25.967 | Home Node B (HNB) Radio Frequency (RF) requirements (FDD) | TR | 18.0.0 | R4 | https://www.3gpp.org/ftp/Specs/archive/25_series/25.967/25967-i00.zip | This document is a technical report which was requested in the Objective 2 of the RAN4 work item description “FDD Home NodeB RF requirements” [5]. The goal of this technical report is to describe the agreed approach towards the RF related issues raised in [5]:
A) The existing UTRA BS classes did not fully address the RF requirements of the HNB application. Proposals for changes to radio performance requirement specifications TS 25.104 are therefore provided in this report, together with the proposals for the test specification TS 25.141. Most of the HNB-specific additions to TS 25.104 / 25.141 were accommodated in a manner similar to the other BS classes.
Editors note:
- Where square bracketed values are suggested in 3GPP TR 25.820, to conduct further work as required to agree appropriate values.
- Where it is suggested that performance values in 3GPP TS 25.104 may be subject to change to conduct further work as required to see if this is necessary.
B) The report intends to ensure that operators are provided with sufficient information to fully understand the issues concerning the deployment of HNBs:
- Deployment scenarios and their potential bottlenecks.
- Guidance on how to control the interference to surrounding macro networks and provide good coverage for the HNB
- Testing of the HNB. |
25.968 | 1.28 Mcps TDD Home NodeB Radio Frequency (RF) | TR | 18.0.0 | R4 | https://www.3gpp.org/ftp/Specs/archive/25_series/25.968/25968-i00.zip | This document is a technical report which was requested in the Objective of the RAN4 work item description “1.28Mcps TDD Home NodeB RF requirements” [1]. The goal of this technical report is to describe the agreed approach towards the RF related issues raised in [1]:
A) The existing 1.28Mcps BS classes did not fully address the RF requirements of the Home NodeB application. Proposals for changes to radio performance requirement specifications TS 25.105 are therefore provided in this report, together with the proposals for the test specification TS 25.142.
B) The report intends to provide guidance to mitigate interference and clarify some interference cases |
25.992 | Multimedia Broadcast/Multicast Service (MBMS); UTRAN/GERAN requirements | TR | 18.0.0 | RP | https://www.3gpp.org/ftp/Specs/archive/25_series/25.992/25992-i00.zip | |
25.993 | Typical examples of Radio Access Bearers (RABs) and Radio Bearers (RBs) supported by Universal Terrestrial Radio Access (UTRA) | TR | 18.0.0 | RP | https://www.3gpp.org/ftp/Specs/archive/25_series/25.993/25993-i00.zip | The present document provides a list of examples of RABs and RAB combinations which are supported by UTRA with examples of radio interface mapping for these RABs onto Radio Bearers and Signalling Radio Bearers.
This list of examples describes typical parameters, and should only be understood as possible configurations i.e. any other configuration supported by the Core Specifications and consistent with a given UE capability shall also be supported by this UE.
The present document addresses the FDD mode as well as the TDD mode.
This report is a release independent report. This means that the latest release applicable to 3GPP is the reference that this TR is defined upon, and contains information on all previous releases. Actual release where a given example applies is indicated in the relevant section. |
25.996 | Spatial channel model for Multiple Input Multiple Output (MIMO) simulations | TR | 18.0.0 | RP | https://www.3gpp.org/ftp/Specs/archive/25_series/25.996/25996-i00.zip | The present document details the output of the combined 3GPP-3GPP2 spatial channel model (SCM) ad-hoc group (AHG).
The scope of the 3GPP-3GPP2 SCM AHG is to develop and specify parameters and methods associated with the spatial channel modelling that are common to the needs of the 3GPP and 3GPP2 organizations. The scope includes development of specifications for:
System level evaluation.
Within this category, a list of four focus areas are identified, however the emphasis of the SCM AHG work is on items a and b.
a) Physical parameters (e.g. power delay profiles, angle spreads, dependencies between parameters)
b) System evaluation methodology.
c) Antenna arrangements, reference cases and definition of minimum requirements.
d) Some framework (air interface) dependent parameters.
Link level evaluation.
The link level models are defined only for calibration purposes. It is a common view within the group that the link level simulation assumptions will not be used for evaluation and comparison of proposals. |
26.071 | Mandatory speech CODEC speech processing functions; AMR speech Codec; General description | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.071/26071-i00.zip | The present document is an introduction to the speech processing parts of the narrowband telephony speech service employing the Adaptive Multi-Rate (AMR) speech coder. A general overview of the speech processing functions is given, with reference to the documents where each function is specified in detail. |
26.073 | ANSI-C code for the Adaptive Multi Rate (AMR) speech codec | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.073/26073-i00.zip | The present document contains an electronic copy of the ANSI‑C code for the Adaptive Multi-Rate codec. The ANSI‑C code is necessary for a bit exact implementation of the Adaptive Multi Rate speech transcoder (TS 26.090 [2]), Voice Activity Detection (TS 26.094 [6]), comfort noise (TS 26.092 [4]), source controlled rate operation (TS 26.093 [5]) and example solutions for substituting and muting of lost frames (TS 26.091 [3]). |
26.090 | Mandatory Speech Codec speech processing functions; Adaptive Multi-Rate (AMR) speech codec; Transcoding functions | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.090/26090-i00.zip | The present document describes the detailed mapping from input blocks of 160 speech samples in 13‑bit uniform PCM format to encoded blocks of 95, 103, 118, 134, 148, 159, 204, and 244 bits and from encoded blocks of 95, 103, 118, 134, 148, 159, 204, and 244 bits to output blocks of 160 reconstructed speech samples. The sampling rate is 8 000 samples/s leading to a bit rate for the encoded bit stream of 4.75, 5.15, 5.90, 6.70, 7.40, 7.95, 10.2 or 12.2 kbit/s. The coding scheme for the multi-rate coding modes is the so‑called Algebraic Code Excited Linear Prediction Coder, hereafter referred to as ACELP. The multi-rate ACELP coder is referred to as MR-ACELP.
In the case of discrepancy between the requirements described in the present document and the fixed point computational description (ANSI‑C code) of these requirements contained in [4], the description in [4] will prevail. The ANSI‑C code is not described in the present document, see [4] for a description of the ANSI‑C code.
The transcoding procedure specified in the present document is mandatory for systems using the AMR speech codec. |
26.077 | Minimum performance requirements for Noise Suppresser; Application to the Adaptive Multi-Rate (AMR) speech encoder | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.077/26077-i00.zip | The present document specifies recommended minimum performance requirements for noise suppression algorithms intended for application in conjunction with the AMR speech encoder. This specification is for guidance purposes. Noise Suppression is intended to enhance the speech signal corrupted by acoustic noise at the input to the AMR speech encoder.
The use of this recommended minimum performance requirements specification is not mandatory except for those solutions intended to be endorsed by SMG11.
It is the intention of SMG11 to perform analysis and validation of any AMR noise suppression solution which is voluntarily brought to the attention of SMG11 in the future, using the requirements set out in this specification to facilitate such an analysis. In order for SMG11 to endorse such a solution, SMG11 must confirm that all the recommended minimum performance requirements are met. |
26.074 | Mandatory speech codec speech processing functions; Adaptive Multi-Rate (AMR) speech codec test sequences | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.074/26074-i00.zip | The present document specifies the digital test sequences for the adaptive multi-rate (AMR) speech codec. These sequences test for a bit exact implementation of the adaptive multi-rate speech transcoder (TS 26.090 [2]), voice activity detection (TS 26.094 [5]), comfort noise (TS 26.092 [3]), and source controlled rate operation (TS 26.093 [4]). |
26.091 | Mandatory Speech Codec speech processing functions; Adaptive Multi-Rate (AMR) speech codec; Error concealment of lost frames | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.091/26091-i00.zip | The present document defines an error concealment procedure, also termed frame substitution and muting procedure, which shall be used by the AMR speech codec receiving end when one or more lost speech or lost Silence Descriptor (SID) frames are received.
The requirements of the present document are mandatory for implementation in all networks and User Equipment (UE)s capable of supporting the AMR speech codec. It is not mandatory to follow the bit exact implementation outlined in the present document and the corresponding C source code. |
26.092 | Mandatory speech codec speech processing functions; Adaptive Multi-Rate (AMR) speech codec; Comfort noise aspects | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.092/26092-i00.zip | The present document gives the detailed requirements for the correct operation of the background acoustic noise evaluation, noise parameter encoding/decoding and comfort noise generation for the AMR speech codec during Source Controlled Rate (SCR) operation.
The requirements described in the present document are mandatory for implementation in all UEs capable of supporting the AMR speech codec.
The receiver requirements are mandatory for implementation in all networks capable of supporting the AMR speech codec, the transmitter requirements only for those where downlink SCR will be used.
In case of discrepancy between the requirements described in the present document and the fixed point computational description of these requirements contained in [1], the description in [1] will prevail. |
26.093 | Mandatory speech codec speech processing functions Adaptive Multi-Rate (AMR) speech codec; Source controlled rate operation | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.093/26093-i00.zip | This document describes the Source Controlled Rate (SCR)operation of the Adaptive Multi-Rate speech Codec in Codec Types UMTS_AMR and UMTS_AMR2 for the UMTS system. The implementation of this SCR operation is mandatory in all UMTS equipment.
The description is structured according to the block diagram in figure 1. This structure of distributing the various functions between system entities is not mandatory for implementation, as long as the operation on the speech decoder output remains the same.
Annex A describes the Discontinuous Transmission (DTX) operation of the Adaptive Multi-Rate speech Codec in Codec Types FR_AMR, HR_AMR and OHR_AMR for GERAN. This annex is the former GSM 06.93 (Release 98).
Annexes B to E describe the SCR operation of the Adaptive Multi-Rate speech Codec in Codec Types GSM_EFR, TDMA_EFR, TDMA_US1 and PDC_EFR for the UMTS system. |
26.094 | Mandatory speech codec speech processing functions; Adaptive Multi-Rate (AMR) speech codec; Voice Activity Detector (VAD) | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.094/26094-i00.zip | The present document specifies two alternatives for the Voice Activity Detector (VAD) to be used in the Discontinuous Transmission (DTX) as described in [3]. Implementors of mobile station and infrastructure equipment conforming to the AMR specifications can choose which of the two VAD options to implement. There are no interoperability factors associated with this choice.
The requirements are mandatory on any VAD to be used either in User Equipment (UE) or Base Station Systems (BSS)s that utilize the AMR speech codec. |
26.101 | Mandatory speech codec speech processing functions; Adaptive Multi-Rate (AMR) speech codec frame structure | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.101/26101-i00.zip | The present document describes a generic frame format for the Adaptive Multi-Rate (AMR) speech codec and the Enhanced Full Rate (GSM-EFR) speech codec. This format shall be used as a common reference point when interfacing speech frames between different elements of the 3G system and between different systems. Appropriate mappings to and from this generic frame format will be used within and between each system element.
Annex A describes a second frame format which shall be used when octet alignment of AMR frames is required. |
26.102 | Mandatory speech codec; Adaptive Multi-Rate (AMR) speech codec; Interface to Iu, Uu and Nb | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.102/26102-i00.zip | The present document specifies the mapping of the AMR generic frame format (3GPP TS 26.101) to the Iu Interface (3GPP TS 25.415 [7]), the Uu Interface and the Nb Interface (3GPP TS 29.415). It further specifies the mapping of Enhanced Full Rate (GSM_EFR) coded speech and of PCM 64 kBit/s (ITU-T G.711 [9]) coded speech to the Nb Interface in a BICC-based circuit switched core network.
The present document also specifies the mapping of Full Rate (GSM_FR) coded speech and of Half Rate (GSM_HR) coded speech to the Nb Interface in a BICC-based circuit switched core network.
The present document also specifies the transport of the AMR Codec Types, the AMR-WB Codec Types, the GSM_EFR Codec, the GSM_FR Codec, the GSM_HR Codec and the ITU-T G.711 Codec over the A-Interface over IP (3GPP TS 48.002 [11]) and the Nb-Interface in a SIP-I -based circuit switched core network (3GPP TS 23.231 [12]). |
26.103 | Speech codec list for GSM and UMTS | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.103/26103-i00.zip | The present Technical Specification outlines the Codec Lists in 3GPP including both systems, GSM and UMTS, to be used by the Out of Band Transcoder Control (OoBTC) protocol to set up a call or modify a call in Transcoder Free Operation (TrFO) and in "transcoder at the edge" scenarios.
The TS also specifies the SDP description of 3GPP Codecs to be used within a SIP-I -based circuit switched core network as specifies in 3GPP TS 23.231 [14].
The TS further specifies the coding of the Supported Codec List Information Elements for the UMTS radio access technology.
The TS further reserves the Code Point for the CSData (dummy) Codec Type for the negotiation of A-Interface Type and the RTP redundancy for CS Data and Fax services, see 3GPP TS 48.008 [23].
The Supported Codec List IE includes Codec_Types from the TDMA and PDC systems, to support TFO or TrFO between UMTS and TDMA, or UMTS and PDC. |
26.104 | ANSI-C code for the floating-point Adaptive Multi-Rate (AMR) speech codec | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.104/26104-i00.zip | This Technical Standard (TS) contains an electronic copy of the ANSI-C code for a floating-point implementation of the Adaptive Multi-Rate codec. This floating-point codec specification is mainly targeted to be used in multimedia applications such as the 3G-324M terminal specified in 3GPP TS 26.110, or in packet-based (e.g., H.323) applications. The bit-exact fixed-point ANSI-C code in 3GPP TS 26.073 remains the preferred implementation for all applications, but the floating-point codec may be used instead of the fixed-point codec when the implementation platform is better suited for a floating-point implementation. It has been verified that the fixed-point and floating-point codecs interoperate with each other without any artefacts.
The floating-point ANSI‑C code in this specification is the only standard conforming non-bit-exact implementation of the Adaptive Multi Rate speech transcoder (3GPP TS 26.090 [2]), Voice Activity Detection (3GPP TS 26.094 [6]), comfort noise generation (3GPP TS 26.092 [4]), and source controlled rate operation (3GPP TS 26.093 [5]). The floating-point code also contains example solutions for substituting and muting of lost frames (3GPP TS 26.091 [3]).
The fixed-point specification in 26.073 shall remain the only allowed implementation for the 3G mandatory speech service and the use of the floating-point codec is strictly limited to other services.
The floating-point encoder in this specification is a non-bit-exact implementation of the fixed-point encoder producing quality indistinguishable from that of the fixed-point encoder. The decoder in this specification is functionally a bit‑exact implementation of the fixed-point decoder, but the code has been optimized for speed and the standard fixed-point libraries are not used as such. |
26.110 | Codec for circuit switched multimedia telephony service; General description | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.110/26110-i00.zip | This specification introduces the set of specifications which apply to 3G-324M multimedia terminals. |
26.111 | Codec for circuit switched multimedia telephony service; Modifications to H.324 | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.111/26111-i00.zip | In ITU-T Recommendation H.324 [10] with annex C describes a generic multimedia codec for use in error-prone, wireless networks. The scope of the present document are the changes, deletions, and additions to those texts necessary to fully specify a multimedia codec for use in 3GPP networks. Note that this implicitly excludes the network interface and call setup procedures. Also excluded are any general introductions to the system components. |
26.113 | Real-Time Media Communication; Protocols and APIs | TS | 18.3.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.113/26113-i30.zip | |
26.114 | IP Multimedia Subsystem (IMS); Multimedia Telephony; Media handling and interaction | TS | 18.11.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.114/26114-ib0.zip | |
26.115 | Echo control for speech and multimedia services | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.115/26115-i00.zip | The present document specifies minimum performance requirements for the gateway echo control of 3G speech and multi-media services. The present document is applicable to any narrow band speech telephony or multimedia service. |
26.116 | Television (TV) over 3GPP services; Video profiles | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.116/26116-i00.zip | The present document specifies requirements and guidelines on video source formats (frame rate, resolution, aspect ratio, colorimetry, bit depth…) and encoding parameters (codec format, random access point period, SEI messages…) for different types of TV services, including linear TV, catch-up TV or on-demand services. A limited set of Operation Points (e.g. SDTV, HDTV, UHD, 8K UHD, …) are defined to provide confidence to content providers/broadcasters on the quality of experience offered by 3GPP services when used for TV-like distribution. Operation Points define format and encoding restrictions but may also be viewed as compatibility points for UEs.
In particular, the Operation Points defined in the present document may serve as the primary tested configurations for TV centric video distribution. The Operation Points are defined based on the analysis and findings in the technical report TR 26.949 [2].
In addition, in the context of DASH operations, not only the main distribution formats are defined, but also a subset of spatial and temporal resolutions. In order to minimize testing for seamless switching experience, suitable lower resolutions of distribution formats are defined. Furthermore, to compensate congestion situations, a minimum service quality is defined in order to provide service continuity.
The specification is aligned with the Common Media Application Format (CMAF) as defined in ISO/IEC 23000-19 [13] to a large extent. Differences and further restrictions compared to CMAF baseline formats as well as media profiles are highlighted. |
26.117 | 5G Media Streaming (5GMS); Speech and audio profiles | TS | 18.4.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.117/26117-i40.zip | The present document specifies speech and audio media capabilities, operation points and media profiles for 5G Media Streaming in the context of 3GPP services and deployments. Speech and audio media capabilities, operation points and media profiles are also provided for usage in other streaming applications. |
26.118 | Virtual Reality (VR) profiles for streaming applications | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.118/26118-i00.zip | The present document defines interoperable formats for Virtual Reality for streaming services. Specifically, the present document defines operation points, media profiles and presentation profiles for Virtual Reality. The present document builds on the findings and conclusions in TR 26.918 [2]. |
26.119 | Media Capabilities for Augmented Reality | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.119/26119-i00.zip | The present document defines the supported media formats, codecs, processing functions for XR Devices in UE per XR device type category. The present document addresses the interoperability gaps identified in the conclusions of TR 26.998 [3]. |
26.130 | Speech/Audio Codec RTP Payload Format Conformance for UE Testing | TS | 18.1.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.130/26130-i10.zip | The present document is applicable to any terminal capable of supporting narrowband, wideband, super-wideband or fullband telephony, either as a stand-alone service or as the telephony component of a multimedia service. The present document specifies requirements and test methods to verify correct implementations of the RTP payload format for 3GPP codecs in UE. The focus is on conversational services in LTE, NR and WLAN terminals when used to provide narrowband, wideband, super-wideband or fullband telephony. |
26.131 | Terminal acoustic characteristics for telephony; Requirements | TS | 18.1.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.131/26131-i10.zip | The present document is applicable to any terminal capable of supporting narrowband, wideband, super-wideband or fullband telephony, either as a stand-alone service or as the telephony component of a multimedia service. The present document specifies minimum performance requirements for the electro-acoustic characteristics of 3G, LTE, NR and WLAN terminals when used to provide narrowband, wideband, super-wideband or fullband telephony.
The set of minimum performance requirements enables a guaranteed level of speech quality while taking possible physical limits of the terminal design into account. Some performance objectives are also defined, if such design limits can be overcome. Care must be taken in applying performance objectives in isolation, not to degrade overall end-user speech quality. |
26.132 | Speech and video telephony terminal acoustic test specification | TS | 18.2.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.132/26132-i20.zip | The present document is applicable to any terminal capable of supporting narrowband, wideband, super-wideband or fullband telephony, either as a stand-alone service or as the telephony component of a multimedia service. The present document specifies test methods to allow the minimum performance requirements for the electro-acoustic characteristics of GSM, 3G, LTE, NR and WLAN terminals when used to provide narrowband, wideband, super-wideband or fullband telephony to be assessed.
NOTE For 3G, LTE, NR and WLAN, acoustic requirements are specified in TS 26.131, test methods are specified in TS 26.132. For GSM, most acoustic requirements are specified in TS 43.050, test methods are specified in TS 51.010. These specifications are in many cases harmonized with or even refer to TS 26.131 and TS 26.132. See TS 43.050 and TS 51.010 for details. The reason for including GSM, UMTS, LTE, NR and WLAN terminals within the scope of the present specification is to avoid, whenever possible, duplication of test method descriptions for terminals supporting multiple access technologies.. |
26.139 | Real-time Transport Protocol (RTP) / RTP Control Protocol (RTCP) verification procedures | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.139/26139-i00.zip | The present document describes:
- Test cases needed to ensure an adequate level of RTP operation and RTP stream monitoring.
- Test methods capable to verify that information contained in the RTP header and in RTCP is correct and consistent with the observed characteristics of the related RTP streams:
- Between RTP/RTCP within the scope of a single RTP stream (e.g. between an RTP stream and the corresponding RTCP reporting from the remote party, or between an RTP stream and the corresponding RTCP metadata, e.g. for sampling clock accuracy compensation between RTP sender and RTP receiver).
- Between RTP/RTCP across RTP streams in the same RTP session (e.g. between sent and received RTP streams, or between audio RTP streams and video RTP streams).
- Requirements on what constitutes acceptable RTP/RTCP protocol field values, including RTP payload header and RTP payload length, based on the observed characteristics of the related RTP streams.
- A method for an RTP/RTCP implementation to announce on the network that it has passed the necessary tests and conforms to the new specification. |
26.140 | Multimedia Messaging Service (MMS); Media formats and codecs | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.140/26140-i00.zip | The present document specifies message bodies for MMS that include different media types, formats and codecs within the 3GPP system. The scope of the present document extends to codecs for speech, audio, video, still images, bitmap graphics, 3D scenes and assets, and other media in general, as well as scene description, multimedia integration and synchronization schemes. |
26.141 | IP Multimedia System (IMS) Messaging and Presence; Media formats and codecs | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.141/26141-i00.zip | The present document specifies the basic media formats and codecs to be used in the IMS Messaging and Presence services, including CSI. It defines the mandatory 'baseline' set of media types for the services. Additionally, it also targets to allow possible message content type enhancements, either 3GPP-standardized or other generally used media types, in a flexible way. |
26.142 | Dynamic and Interactive Multimedia Scenes (DIMS) | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.142/26142-i00.zip | DIMS defines a dynamic rich-media system, including a media type, its packaging, delivery, and interaction with the local terminal, user, and other local and remote sub-systems. Enhanced end-user experiences are provided by the coordinated management and synchronization of media and events, combined with end-user interaction.
The DIMS media type can be used as a generic media type, allowing creating dynamic interactive rich-media services and can also benefit, or be used in association with other media types (e.g.: audio codecs, video codecs, XHTML browser, etc.). |
26.143 | Messaging Media profiles | TS | 18.2.1 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.143/26143-i21.zip | The present document specifies the media types, formats, codecs capabilities and profiles for the messaging applications used over the 5G System. The scope of the present document extends to codecs for speech, audio, video, still images, bitmap graphics, 3D scenes and assets, and other media in general, as well as scene description. |
26.150 | Syndicated Feed Reception (SFR) within 3GPP environments; Protocols and codecs | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.150/26150-i00.zip | The present document defines a set of media codecs, formats and transport/application protocols to enable syndicated feed reception within the 3GPP system.
The present document includes information applicable to network operators, service providers and manufacturers. |
26.171 | Speech codec speech processing functions; Adaptive Multi-Rate - Wideband (AMR-WB) speech codec; General description | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.171/26171-i00.zip | The present document is an introduction to the speech processing parts of the wideband telephony speech service employing the Adaptive Multi-Rate Wideband (AMR-WB) speech coder. A general overview of the speech processing functions is given, with reference to the documents where each function is specified in detail. |
26.173 | ANSI-C code for the Adaptive Multi-Rate - Wideband (AMR-WB) speech codec | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.173/26173-i00.zip | The present document contains an electronic copy of the ANSI‑C code for the Adaptive Multi-Rate Wideband codec. The ANSI‑C code is necessary for a bit exact implementation of the Adaptive Multi Rate Wideband speech transcoder (3GPP TS 26.190 [2]), Voice Activity Detection (3GPP TS 26.194 [6]), comfort noise (3GPP TS 26.192 [4]), source controlled rate operation (3GPP TS 26.193 [5]) and example solutions for substituting and muting of lost frames (3GPP TS 26.191 [3]). |
26.174 | Speech codec speech processing functions; Adaptive Multi-Rate - Wideband (AMR-WB) speech codec test sequences | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.174/26174-i00.zip | The present document specifies the digital test sequences for the adaptive multi-rate wideband (AMR-WB) speech codec. These sequences test for a bit-exact implementation of the adaptive multi-rate wideband (AMR-WB) speech transcoder (TS 26.190 [2]), voice activity detection (TS 26.194 [5]), comfort noise (TS 26.192 [3]), and source controlled rate operation (TS 26.193 [4]). |
26.177 | Speech Enabled Services (SES); Distributed Speech Recognition (DSR) extended advanced front-end test sequences | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.177/26177-i00.zip | The present document specifies the digital test sequences for the DSR Extended Advanced Front-end speech codec. These sequences can be used to test for a bit exact implementation of the DSR Advanced Front-end codec and quantization (3GPP TS 26.243). |
26.179 | Mission Critical Push To Talk (MCPTT); Codecs and media handling | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.179/26179-i00.zip | The present document specifies the codecs and media handling for MCPTT. The corresponding service requirements are defined in 3GPP TS 22.179 [2]. The corresponding functional architecture, procedures and information flows are defined in 3GPP TS 23.179 [3]. |
26.190 | Speech codec speech processing functions; Adaptive Multi-Rate - Wideband (AMR-WB) speech codec; Transcoding functions | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.190/26190-i00.zip | This Telecommunication Standard (TS) describes the detailed mapping from input blocks of 320 speech samples in 16‑bit uniform PCM format to encoded blocks of 132, 177, 253, 285, 317, 365, 397, 461 and 477 bits and from encoded blocks of 132, 177, 253, 285, 317, 365, 397, 461 and 477 bits to output blocks of 320 reconstructed speech samples. The sampling rate is 16 000 samples/s leading to a bit rate for the encoded bit stream of 6.60, 8.85, 12.65, 14.25, 15.85, 18.25, 19.85, 23.05 or 23.85 kbit/s. The coding scheme for the multi-rate coding modes is the so‑called Algebraic Code Excited Linear Prediction Coder, hereafter referred to as ACELP. The multi-rate wideband ACELP coder is referred to as MRWB-ACELP. |
26.191 | Speech codec speech processing functions; Adaptive Multi-Rate - Wideband (AMR-WB) speech codec; Error concealment of erroneous or lost frames | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.191/26191-i00.zip | This specification defines an error concealment procedure, also termed frame substitution and muting procedure, which shall be used by the AMR-WB speech codec receiving end when one or more erroneous/lost speech or lost Silence Descriptor (SID) frames are received.
The requirements of this document are mandatory for implementation in all networks and User Equipment (UE)s capable of supporting the AMR-WB speech codec. It is not mandatory to follow the bit exact implementation outlined in this document and the corresponding C source code. |
26.192 | Speech codec speech processing functions; Adaptive Multi-Rate - Wideband (AMR-WB) speech codec; Comfort noise aspects | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.192/26192-i00.zip | This document gives the detailed requirements for the correct operation of the background acoustic noise evaluation, noise parameter encoding/decoding and comfort noise generation for the AMR Wideband (AMR-WB) speech codec during Source Controlled Rate (SCR) operation.
The requirements described in this document are mandatory for implementation in all UEs capable of supporting the AMR-WB speech codec.
The receiver requirements are mandatory for implementation in all networks capable of supporting the AMR-WB speech codec, the transmitter requirements only for those where downlink SCR will be used.
In case of discrepancy between the requirements described in this document and the fixed point computational description of these requirements contained in [1], the description in [1] will prevail. |
26.193 | Speech codec speech processing functions; Adaptive Multi-Rate - Wideband (AMR-WB) speech codec; Source controlled rate operation | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.193/26193-i00.zip | This document describes the Source Controlled Rate (SCR) operation of the Adaptive Multi-Rate Wideband speech Codec. The implementation of this SCR operation is mandatory in all UMTS equipment implementing the Adaptive Multi-Rate Wideband speech Codec.
The description is structured according to the block diagram in Figure 1. This structure of distributing the various functions between system entities is not mandatory for implementation, as long as the operation on the speech decoder output remains the same.
Annex A describes the Discontinuous Transmission (DTX) operation of the Adaptive Multi-Rate Wideband speech Codec in Codec Type FR_AMR-WB for the GSM system. |
26.194 | Speech codec speech processing functions; Adaptive Multi-Rate - Wideband (AMR-WB) speech codec; Voice Activity Detector (VAD) | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.194/26194-i00.zip | This document specifies the Voice Activity Detector (VAD) to be used in the Discontinuous Transmission (DTX) as described in [3].
The requirements are mandatory on any VAD to be used either in User Equipment (UE) or Base Station Systems (BSS)s that utilize the AMR wideband speech codec. |
26.201 | Speech codec speech processing functions; Adaptive Multi-Rate - Wideband (AMR-WB) speech codec; Frame structure | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.201/26201-i00.zip | The present document describes a generic frame format for the Adaptive Multi-Rate Wideband (AMR-WB) speech codec. This format shall be used as a common reference point when interfacing speech frames between different elements of the 3G system and between different systems. Appropriate mappings to and from this generic frame format will be used within and between each system element.
Annex A describes a second frame format which shall be used when octet alignment of AMR-WB frames is required. |
26.202 | Speech codec speech processing functions; Adaptive Multi-Rate - Wideband (AMR-WB) speech codec; Interface to Iu, Uu and Nb | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.202/26202-i00.zip | The present document specifies the mapping of the AMR wideband generic frame format (3GPP TS 26.201) to the Iu Interface (3GPP TS 25.415), the Uu Interface and the Nb Interface (3GPP TS 29.415) of a BICC-based circuit switched core network. It further specifies the mapping of PCM 64 kBit/s (ITU-T G.711) coded speech to the Nb Interface of a BICC-based circuit switched core network.
The mapping of the AMR wideband generic frame format to RTP for the A-Interface and the Nb Interface for a SIP-I -based circuit switched core network is described in TS 26.102. |
26.204 | Speech codec speech processing functions; Adaptive Multi-Rate - Wideband (AMR-WB) speech codec; ANSI-C code | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.204/26204-i00.zip | The present document contains an electronic copy of the ANSI‑C code for the Floating-point Adaptive Multi-Rate Wideband codec. This floating-point codec specification is mainly targeted to be used in multimedia applications or in packet-based applications. The bit-exact fixed-point ANSI-C code in 3GPP TS 26.173 remains the preferred implementation for all applications, but the floating-point codec may be used instead of the fixed-point codec when the implementation platform is better suited for a floating-point implementation. It has been verified that the fixed-point and floating-point codecs interoperate with each other without any artifacts.
The floating-point ANSI‑C code in the present document is the only standard conforming non-bit-exact implementation of the Adaptive Multi-Rate Wideband speech transcoder (3GPP TS 26.190 [2]), Voice Activity Detection (3GPP TS 26.194 [6]), comfort noise generation (3GPP TS 26.192 [4]), and source controlled rate operation (3GPP TS 26.193 [5]). The floating-point code also contains example solutions for substituting and muting of lost frames (3GPP TS 26.191 [3]).
The fixed-point specification in 26.173 shall remain the only allowed implementation for the 3G AMR-WB speech service and the use of the floating-point codec is strictly limited to other services.
The floating-point encoder in the present document is a non-bit-exact implementation of the fixed-point encoder producing quality indistinguishable from that of the fixed-point encoder. The decoder in the present document is functionally a bit‑exact implementation of the fixed-point decoder, but the code has been optimized for speed and the standard fixed-point libraries are not used as such. |
26.223 | Telepresence using the IP Multimedia Subsystem (IMS); Media handling and interaction | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.223/26223-i00.zip | The present document specifies a client for the IMS-based telepresence service supporting conversational speech, video and text transported over RTP. Telepresence is defined as a conference with interactive audio-visual communications experience between remote locations, where the users enjoy a strong sense of realism and presence between all participants (i.e. as if they are in same location) by optimizing a variety of attributes such as audio and video quality, eye contact, body language, spatial audio, coordinated environments and natural image size. A telepresence system is defined as a set of functions, devices and network elements which are able to capture, deliver, manage and render multiple high quality interactive audio and video signals in a telepresence conference. An appropriate number of devices (e.g. cameras, screens, loudspeakers, microphones, codecs) and environmental characteristics are used to establish telepresence.
The media handling capabilities of a telepresence client (TP UE) are specified in the present document. A TP UE supports Multimedia Telephony Service for IMS (MTSI) UE media handling capabilities [2], but it also supports more advanced media handling capabilities. The media handling aspects of a TP UE within the scope of the present document include media codecs, media configuration and session control, data transport, audio/video parameters, and interworking with MTSI. |
26.226 | Cellular text telephone modem; General description | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.226/26226-i00.zip | This Technical Specification (TS) concerns the Cellular Text Telephone Modem (CTM). CTM allows reliable transmission of a text telephone conversation alternating with a speech conversation through the existing speech communication paths in cellular mobile phone systems. This reliability is achieved by an improved modulation technique, including error protection, interleaving and synchronization. Together with recommendations ITU-T V.18 and T.140, CTM may serve for worldwide applications in text telephony.
A general overview and explanations of possible implementation architectures is provided. CTM is intended for use in end terminals (on the mobile or fixed side) and within the network for the adaptation between CTM and existing traditional text telephone standards.
The CTM transmitter is fully specified and a bit exact C-code reference is provided. An implementation of an example CTM receiver is also described. |
26.230 | Cellular text telephone modem; Transmitter bit exact C-code | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.230/26230-i00.zip | This Technical Standard (TS) contains an electronic copy of the ANSI‑C code for the Cellular Text Telephone Modem (CTM) for reliable transmission of text telephone text via the speech channel of cellular networks. While CTM is generally usable with text in UCS coding, the example application linked to CTM in this document is limited to use the signals and character set of the Baudot type. |
26.231 | Cellular text telephone modem; Minimum performance requirements | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.231/26231-i00.zip | This Technical Standard (TS) describes the minimum performance requirements for the Cellular Text Telephone Modem (CTM) for reliable transmission of text telephone text via the speech channel of cellular or PSTN networks. The transmitting parts of the Cellular Text Telephone Modem are specified in [1].
CTM is a general technology, independent of text telephone types. The tests are made only for one specific type of text telephone, the Baudot type. The tests are applicable only to a combination of a Baudot codec and CTM and tests the combined performance. A bit-exact implementation of the CTM transmitter as well as an example implementation of the remaining functions of such a combination are provided in [2].
The test scripts and test vectors required to perform this testing are included in a supplement, which is located in the zip archive ctm_testing.zip. The path and file names given in this specification refer to the file structures associated with this supplement. A second supplement (zip archive ctm_score.zip) provides the scoring program that is described in clause 6. |
26.234 | Transparent end-to-end Packet-switched Streaming Service (PSS); Protocols and codecs | TS | 18.1.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.234/26234-i10.zip | The present document specifies the protocols and codecs for the PSS within the 3GPP system. Protocols for control signalling, capability exchange, media transport, rate adaptation and protection are specified. Codecs for speech, natural and synthetic audio, video, still images, bitmap graphics, vector graphics, timed text, text, 360/VR video and 3D/VR audio are specified.
The present document is applicable to IP-based packet-switched networks. |
26.237 | IP Multimedia Subsystem (IMS) based Packet Switch Streaming (PSS) and Multimedia Broadcast/Multicast Service (MBMS) User Service; Protocols | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.237/26237-i00.zip | The present document specifies the usage of IMS protocols to initiate and control PSS and MBMS Streaming and Download User Services based applications. It applies to IMS enabled UEs that also implement PSS and/or MBMS clients. Existing protocols that are used are described in reference to relevant specifications.
The present document is applicable to IP-based packet-switched networks over 3GPP systems.
The present document includes information applicable to network operators, service providers and manufacturers. |
26.238 | Uplink streaming | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.238/26238-i00.zip | The present document defines a FLUS source entity and a FLUS sink entity that can support point-to-point transmission of speech/audio, video, and text. It defines media handling (e.g., signalling, transport, packet-loss handling, and adaptation). The goal is to ensure a reliable and interoperable service with a predictable media quality while allowing for flexibility in the service offerings.
A FLUS source entity, which may be embedded in a single UE, or distributed among a UE and separate audio-visual capture devices, may support all or a subset of the features specified in this document.
When used as a generic framework, only the F-C procedures for establishing the FLUS session are required to be supported by the FLUS source and the FLUS sink entities, and no other feature or procedure specified in this document is mandated. Impact on the service quality and network capacity is left to the discretion of the implementation and the service utilizing the framework. For example, configuration of media formats and codecs follows the requirements of the respective service.
When offered as part of a 3GPP IMS/MTSI service, the FLUS source and the FLUS sink entities are required to support the IMS control plane and media plane procedures, and the service quality is determined by the MTSI service policy.
The specification is written in a forward-compatible way in order to allow additions of media components and functionality in releases beyond Release 15.
. |
26.243 | ANSI-C code for the fixed-point distributed speech recognition extended advanced front-end | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.243/26243-i00.zip | The present document contains an electronic copy of the ANSI‑C code for DSR Extended Advanced Front-end. The ANSI‑C code is necessary for a bit exact implementation of DSR Extended Advanced Front-end. |
26.244 | Transparent end-to-end packet switched streaming service (PSS); 3GPP file format (3GP) | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.244/26244-i00.zip | The present document defines the 3GPP file format (3GP) as an instance of the ISO base media file format. The definition addresses 3GPP specific features such as codec registration and conformance within the MMS, PSS and MBMS services. |
26.245 | Transparent end-to-end Packet switched Streaming Service (PSS); Timed text format | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.245/26245-i00.zip | The present document defines the timed text format relative to the 3GPP file format. This specification defines the format of timed text in downloaded files. |
26.246 | Transparent end-to-end Packet-switched Streaming Service (PSS); 3GPP SMIL language profile | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.246/26246-i00.zip | The present document includes the specification of the 3GPP SMIL Language Profile. The 3GPP SMIL Language Profile is also referred to as "3GPP PSS SMIL Language Profile" [3] and also just "3GPP SMIL".
The 3GPP SMIL Language Profile is based on SMIL 2.0 Basic [7] and SMIL Scalability Framework. It is a clean subset of SMIL 2.0 Full profile [7], and a clear superset of SMIL 2.0 Basic [7].
The 3GPP SMIL Language Profile is used by the PSS [2][3] and MMS [6] services.
The 3GPP SMIL Language Profile is in no way restricted to be used with only these services, but can also be used for other services. |
26.247 | Transparent end-to-end Packet-switched Streaming Service (PSS); Progressive Download and Dynamic Adaptive Streaming over HTTP (3GP-DASH) | TS | 18.4.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.247/26247-i40.zip | The present document specifies Progressive Download and Dynamic Adaptive Streaming over HTTP (3GP-DASH). This specification is part of Packet-switched Streaming Service (PSS) and 5G Media Streaming. HTTP-based progressive download and dynamic adaptive streaming had initially been separated from TS 26.234 to differentiate from RTP-based streaming that is maintained in TS 26.234. HTTP-based progressive download and dynamic adaptive streaming may be deployed independently from RTP-based PSS, for example by using standard HTTP/1.1 servers for hosting data formatted as defined in the present document, and in particular together with 5G Media Streaming. |
26.249 | Immersive Audio for Split Rendering Scenarios; Detailed Algorithmic Description of Split Rendering Functions | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.249/26249-i00.zip | The present document is a detailed algorithmic description of Split Rendering functions (ISAR) addressing Immersive Audio for Split Rendering Scenarios and that are applicable to a broad range of immersive audio coding systems and renderers. Functional solutions are described on an algorithmic level. Annexes of this document specify APIs, RTP payload format and SDP parameters as well as source code and test vectors. |
26.250 | Codec for Immersive Voice and Audio Services (IVAS); General overview | TS | 18.2.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.250/26250-i20.zip | The present document is an introduction to the audio processing parts and auxiliary functions of the codec for Immersive Voice and Audio Services (IVAS codec). A general overview of the audio processing and auxiliary functions is given, with reference to the documents where each function is specified in detail. |
26.251 | Codec for Immersive Voice and Audio Services (IVAS); C code (fixed-point) | TS | 1.0.2 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.251/26251-102.zip | Attached to this document is an electronic copy of the fixed-point C code for the Immersive Voice and Audio Services (IVAS) Codec. This C code is the unique alternative reference specification besides the floating-point C code for the IVAS Codec (3GPP TS 26.258) for a standard compliant implementation of the IVAS Codec (3GPP TS 26.253), Rendering (3GPP TS 26.254), Error Concealment of Lost Packets (3GPP TS 26.255) and Jitter Buffer Management (JBM) (3GPP TS 26.256).
The bit-exact fixed-point C code is the preferred implementation for all applications, but the floating-point codec as specified in 3GPP TS 26.258 may be used instead of the fixed-point codec when the implementation platform is better suited for a floating-point implementation.
Requirements for any implementation of the IVAS codec to be standard compliant are specified in 3GPP TS 26.252 (Test sequences). |
26.252 | Codec for Immersive Voice and Audio Services (IVAS); Test sequences | TS | 18.2.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.252/26252-i20.zip | The present document specifies the digital test sequences for the Immersive Voice and Audio Services (IVAS) codec. These sequences shall be used in conformance testing for implementations of the IVAS codec (3GPP TS 26.253), Rendering (3GPP TS 26.254), Error Concealment of Lost Packets (3GPP TS 26.255) and Jitter Buffer Management (JBM) (3GPP TS 26.256) and its reference C code specification 3GPP TS 26.258 (floating-point). In addition, the present document specifies procedures for conformance testing. |
26.253 | Codec for Immersive Voice and Audio Services (IVAS); Detailed Algorithmic Description including RTP payload format and SDP parameter definitions | TS | 18.5.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.253/26253-i50.zip | The present document is a detailed description of the signal processing algorithms of the Immersive Voice and Audio Services (IVAS) coder including the IVAS renderer. |
26.254 | Codec for Immersive Voice and Audio Services (IVAS); Rendering | TS | 18.1.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.254/26254-i10.zip | The present document provides a comprehensive description of the rendering functions of the decoder/renderer for Immersive Voice and Audio Services (IVAS codec). |
26.255 | Codec for Immersive Voice and Audio Services (IVAS); Error concealment of lost packets | TS | 18.2.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.255/26255-i20.zip | The present document defines a frame loss concealment procedure, also termed frame substitution and muting procedure, which is executed by the Immersive Voice and Audio Services (IVAS) decoder when one or more frames (speech or audio or SID frames) are unavailable for decoding due to e.g. packet loss, corruption of a packet or late arrival of a packet. |
26.256 | Codec for Immersive Voice and Audio Services (IVAS); Jitter Buffer Management | TS | 18.1.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.256/26256-i10.zip | The present document defines the Jitter Buffer Management (JBM) solution for the Immersive Voice and Audio Services (IVAS) codec [2]. Jitter Buffers are required in packet-based communications, such as 3GPP MTSI [7], to smooth the inter-arrival jitter of incoming media packets for uninterrupted playout.
The procedure of the present document is recommended for implementation in all network entities and UEs supporting the IVAS codec; procedures described in [4] and used in this document, such as multi-channel time-scale modification, metadata adaptation and rendering are mandatory for implementations in all network entities and UEs supporting the IVAS codec.
The present document does not describe the C code of this procedure. For a description of the floating-point C code implementation see [3].
In the case of discrepancy between the Jitter Buffer Management described in the present document and its C code specification contained in [3], the procedure defined by [3] prevails. |
26.258 | Codec for Immersive Voice and Audio Services (IVAS); C code (floating-point) | TS | 18.2.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.258/26258-i20.zip | Attached to this document is an electronic copy of the floating-point C code for the Immersive Voice and Audio Services (IVAS) Codec. This C code is the unique reference specification for a standard compliant implementation of the IVAS Codec (3GPP TS 26.253), Rendering (3GPP TS 26.254), Error Concealment of Lost Packets (3GPP TS 26.255) and Jitter Buffer Management (JBM) (3GPP TS 26.256).
Requirements for any implementation of the IVAS codec to be standard compliant are specified in 3GPP TS 26.252 (Test sequences). |
26.259 | Subjective test methodologies for the evaluation of immersive audio systems | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.259/26259-i00.zip | The present document specifies subjective test methodologies for 3GPP immersive audio systems including channel-based, object-based, scene-based and hybrids of these formats. The subjective evaluation methods described in the present document are applicable to audio capture, coding, transmission and rendering as indicated in their corresponding clauses. |
26.260 | Objective test methodologies for the evaluation of immersive audio systems | TS | 18.1.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.260/26260-i10.zip | The present document specifies objective test methodologies for 3GPP immersive audio systems including channel based, object based, scene-based, parametric and hybrids of these formats. The objective evaluation methods described in the present document are applicable to audio capture, coding, transmission and rendering as indicated in their corresponding clauses. They also include testing of IVAS-based UEs [26]. |
26.261 | Terminal audio quality performance requirements for immersive audio services | TS | 18.0.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.261/26261-i00.zip | The present document specifies minimum performance requirements for the electro-acoustic characteristics of LTE, NR and WLAN terminal. It is applicable to any terminal capable of supporting wideband, super-wideband or fullband immersive services.
The set of minimum performance requirements enables a guaranteed level of speech quality while taking possible physical limits of the terminal design into account. Some performance objectives are also defined if such design limits can be overcome.
The present document covers both conversational services based on MTSI / telepresence and non-conversational services. |
26.264 | IMS-based AR Real-Time Communication | TS | 18.2.0 | S4 | https://www.3gpp.org/ftp/Specs/archive/26_series/26.264/26264-i20.zip | The present document focuses on IMS-based conversational AR (Augmented reality) services. AR services can overlay media (e.g., video, audio, text, etc.) on top of the user’s real perception. Conversational AR services as described by the present document typically include a bidirectional conversational A/V connection in addition to other non-real-time AR media for collaboration or communication between two or more users. |
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