The present document provides speech transmission performance requirements for narrowband and wideband media gateways from a QoS perspective as perceived by the user. Media gateways can be network or home based, they may include a transcoding function. The present document covers the following types of IP-based media gateways:
• ATA (Analogue Terminal Adapter), home gateway IP to POTS
• ITA (ISDN Terminal Adapter), home gateway IP to ISDN
• IAD (Integrated Access device), home router including ATA or ITA
• Network based ATA and ITA
• Carrier grade media gateway, network gateway IP to TDM
• IP-to-IP media gateway, network gateway with transcoding and/or other media processing
• New Generation DECT Fixed part with IP interface (only parameters not covered by New Generation DECT) Interfaces of media gateways used together with terminals as a system (i.e. connected via Ethernet or with a proprietary interface) are excluded in the present document and should be measured according to the relevant terminal standard. If a media gateway includes more than one interface type (e.g. POTS and ISDN), each interface has to be dealt with differently. The requirements available in the present document will ensure a high compatibility with IP- and TDM-based fixed and wireless terminals and networks, including DECT and mobile terminals. It is the aim to optimize interoperability, the listening and talking quality and the conversational performance. Related requirements and test methods are defined in the present document. The present document does not apply to media gateways with 4-wire analogue interfaces. The requirements for MGWs with respect to voiceband data (VBD) are out of scope in the present document. These requirements are covered in ETSI TS 102 929 [i.4].

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The present document aims to identify and define testing methodologies which can be used to objectively evaluate the
performance of narrowband and wideband terminals and systems for speech communication in the presence of
background noise.
Background noise is a problem in mostly all situations and conditions and need to be taken into account in both,
terminals and networks. The present document provides information about the testing methods applicable to objectively
evaluate the speech quality in the presence of background noise. The present document includes:
• The description of the experts post evaluation process chosen to select the subjective test data being within the
scope of the objective methods.
• The results of the performance evaluation of the currently existing methods described in Recommendations
ITU-T P.862 [i.16] and P.862.1 [i.17] and in TOSQA2001 [i.19] which is chosen for the evaluation of
terminals in the framework of ETSI VoIP speech quality test events [i.8], [i.9], [i.10] and [i.11].
• The method which is applicable to objectively determine the different parameters influencing the speech
quality in the presence of background noise taking into account:
- the speech quality;
- the background noise transmission quality;
- the overall quality.
• The present document is to be used in conjunction with:
- ETSI ES 202 396-1 [i.1] which describes a recording and reproduction setup for realistic simulation of
background noise scenarios in lab-type environments for the performance evaluation of terminals and
communication systems.
- ETSI EG 202 396-2 [i.2] which describes the simulation of network impairments and how to simulate
realistic transmission network scenarios and which contains the methodology and results of the
subjective scoring for the data forming the basis of the present document.
- French speech sentences as defined in Recommendation ITU-T P.501 [i.13] for wideband and English
speech sentences as defined in Recommendation ITU-T P.501 [i.13] for narrowband.

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The quality of background noise transmission is an important factor, which significantly contributes to the perceived
overall quality of speech. Existing and even more the new generation of terminals, networks and system configurations
including broadband services can be greatly improved with a proper design of terminals and systems in the presence of
background noise. The present document:
• describes a noise simulation environment using realistic background noise scenarios for laboratory use;
• contains a database including the relevant background noise samples for subjective and objective evaluation.
The present document provides information about the recording techniques needed for background noise recordings and
discusses the advantages and drawbacks of existing methods. The present document describes the requirements for
laboratory conditions. The loudspeaker setup and the loudspeaker calibration and equalization procedure are described.
The simulation environment specified can be used for the evaluation and optimization of terminals and of complex
configurations including terminals, networks and other configurations. The main application areas should be: office,
home and car environment.
The setup and database as described in the present document are applicable for:
• Objective performance evaluation of terminals in different (simulated) background noise environments.
• Speech processing evaluation by using the pre-processed speech signal in the presence of background noise,
recorded by a terminal.
• Subjective evaluation of terminals by performing conversational tests, specific double talk tests or talking and
listening tests in the presence of background noise.
• Subjective evaluation in third party listening tests by recording the speech samples of terminals in the presence
of background noise.

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The present document provides speech transmission performance requirements for narrowband and wideband media
gateways from a QoS perspective as perceived by the user. Media gateways can be network or home based, they may
include a transcoding function. The present document covers the following types of IP-based media gateways:
• ATA (Analogue Terminal Adapter), home gateway IP to POTS
• ITA (ISDN Terminal Adapter), home gateway IP to ISDN
• IAD (Integrated Access device), home router including ATA or ITA
• Network based ATA and ITA
• Carrier grade media gateway, network gateway IP to TDM
• IP-to-IP media gateway, network gateway with transcoding and/or other media processing
• New Generation DECT Fixed part with IP interface (only parameters not covered by New Generation DECT)
Interfaces of media gateways used together with terminals as a system (i.e. connected via Ethernet or with a proprietary
interface) are excluded in the present document and should be measured according to the relevant terminal standard.
If a media gateway includes more than one interface type (e.g. POTS and ISDN), each interface has to be dealt with
differently.
The requirements available in the present document will ensure a high compatibility with IP-and TDM-based fixed and
wireless terminals and networks, including DECT and mobile terminals.
It is the aim to optimize interoperability, the listening and talking quality and the conversational performance. Related
requirements and test methods are defined in the present document.
The present document does not apply to media gateways with 4-wire analogue interfaces.
The requirements for MGWs with respect to voiceband data (VBD) are out of scope in the present document. These
requirements are covered in ETSI TS 102 929 [i.5].

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The present document aims to identify and define testing methodologies which can be used to objectively evaluate the
performance of narrowband and wideband terminals and systems for speech communication in the presence of
background noise.
Background noise is a problem in mostly all situations and conditions and need to be taken into account in both,
terminals and networks. The present document provides information about the testing methods applicable to objectively
evaluate the speech quality in the presence of background noise. The present document includes:
• The description of the experts post evaluation process chosen to select the subjective test data being within the
scope of the objective methods.
• The results of the performance evaluation of the currently existing methods described in Recommendations
ITU-T P.862 [i.16] and P.862.1 [i.17] and in TOSQA2001 [i.19] which is chosen for the evaluation of
terminals in the framework of ETSI VoIP speech quality test events [i.8], [i.9], [i.10] and [i.11].
• The method which is applicable to objectively determine the different parameters influencing the speech
quality in the presence of background noise taking into account:
- the speech quality;
- the background noise transmission quality;
- the overall quality.
• The present document is to be used in conjunction with:
- ETSI ES 202 396-1 [i.1] which describes a recording and reproduction setup for realistic simulation of
background noise scenarios in lab-type environments for the performance evaluation of terminals and
communication systems.
- ETSI EG 202 396-2 [i.2] which describes the simulation of network impairments and how to simulate
realistic transmission network scenarios and which contains the methodology and results of the
subjective scoring for the data forming the basis of the present document.
- French speech sentences as defined in Recommendation ITU-T P.501 [i.13] for wideband and English
speech sentences as defined in Recommendation ITU-T P.501 [i.13] for narrowband.

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The document will provide the umbrella for the different parts of this multi-part document as these are service specific (e.g. voice) on the one hand and transport network specific (e.g. IP) on the other hand.

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Revision to take into account the MV comments. This part 2 is moved from EG to ES  by TB decision.

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The aim of DEG/STQ-00104-4 is to identify and define indicators and methodologies for a use in a context of end-user quality characterisation and supervision of multiplay services for Internet access, IPTV, VoD and for services associated to telephony offers.

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The present document provides an overview of the common metric definitions and measurement method specifications
upon which the interoperability of network performance measurement (also called QoS measurement) is based. Two
different standardisation bodies, the Internet Engineering Task Force (IETF) and the International Telecommunication
Union - Telecommunication Standardization Sector (ITU - T), have addressed this issue. The present document
addresses the following points:
• Survey the existing network performance related IETF standards and how these standards can be applied to
end-to-end network performance measurements. The scope of this work is also to discuss the relationship of
those standards to those of ITU-T and ETSI.
• Discuss and compare definitions of metrics used to specify and assess performance in IP networks. The
metrics addressed in the present document are those defined by the IETF IPPM working group and ITU-T
Study Group 12. Besides comparing the different definitions, the present document gives applicability
guidelines on which metric is more appropriate for a particular application, configuration or scenario.
• Define measurement methods for selected performance metrics in IP networks, addressing both active and
passive methods. Clarifying guidelines are given.
NOTE: All text sections in the remainder of the present document which are enclosed in quotation marks (") and
formatted in italic style denote citations taken verbatim from referenced documents.

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The present document aims to identify and define testing methodologies which can be used to objectively evaluate the
performance of narrowband and wideband terminals and systems for speech communication in the presence of
background noise.
Background noise is a problem in mostly all situations and conditions and need to be taken into account in both,
terminals and networks. The present document provides information about the testing methods applicable to objectively
evaluate the speech quality in the presence of background noise. The present document includes:
• The description of the experts post evaluation process chosen to select the subjective test data being within the
scope of the objective methods.
• The results of the performance evaluation of the currently existing methods described in Recommendations
ITU-T P.862 [i.16] and P.862.1 [i.17] and in TOSQA2001 [i.19] which is chosen for the evaluation of
terminals in the framework of ETSI VoIP speech quality test events [i.8], [i.9], [i.10] and [i.11].
• The method which is applicable to objectively determine the different parameters influencing the speech
quality in the presence of background noise taking into account:
- the speech quality;
- the background noise transmission quality;
- the overall quality.
• The present document is to be used in conjunction with:
- ETSI ES 202 396-1 [i.1] which describes a recording and reproduction setup for realistic simulation of
background noise scenarios in lab-type environments for the performance evaluation of terminals and
communication systems.
- ETSI EG 202 396-2 [i.2] which describes the simulation of network impairments and how to simulate
realistic transmission network scenarios and which contains the methodology and results of the
subjective scoring for the data forming the basis of the present document.
- French speech sentences as defined in Recommendation ITU-T P.501 [i.13] for wideband and English
speech sentences as defined in Recommendation ITU-T P.501 [i.13] for narrowband.

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The present document provides speech transmission performance requirements for narrowband and wideband media gateways from a QoS perspective as perceived by the user. Media gateways can be network or home based, they may include a transcoding function. The present document covers the following types of IP-based media gateways:
• ATA (Analogue Terminal Adapter), home gateway IP to POTS
• ITA (ISDN Terminal Adapter), home gateway IP to ISDN
• IAD (Integrated Access device), home router including ATA or ITA
• Network based ATA and ITA
• Carrier grade media gateway, network gateway IP to TDM
• IP-to-IP media gateway, network gateway with transcoding and/or other media processing DECT interfaces of media gateways are excluded from the present document and should be measured according to the relevant DECT standards. Interfaces of media gateways used together with terminals as a system (i.e. connected via Ethernet or with a proprietary interface) are excluded in the present document and should be measured according to the relevant terminal standard. If a media gateway includes more than one interface type (e.g. POTS and ISDN), each interface has to be dealt with
differently. The requirements available in the present document will ensure a high compatibility with IP-and TDM-based fixed and wireless terminals and networks, including DECT and mobile terminals. It is the aim to optimize interoperability, the listening and talking quality and the conversational performance. Related requirements and test methods are defined in the present document. The present document does not apply to media gateways with 4-wire analogue interfaces.

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The present document contains definitions and measurement methods for a range of user perceivable Quality of Service (QoS) parameters. The purpose of these parameters is to define objective and comparable measures of the QoS delivered to users/customers for use by users/customers. The present document applies to any telecommunication service, however, some parameters may have a limited application. The present document is intended to provide a menu from which individual items can be selected. There is no obligation to use any or all of the parameters.
The QoS parameters are related primarily to services and service features and not to the technology used to provide the services. Therefore the parameters should be capable of use when the services are provided on new technologies such as IP and ATM or other packet switched technologies as well as on circuit switched technologies.
The establishment of target values for QoS is beyond the scope of the present document. The QoS parameters listed in the present document are also not intended to assess the complete QoS of a telecommunication service. The present document provides a set of QoS parameters that covers specific user related QoS aspects rather than a complete list of QoS parameters. This set has been chosen to address areas where monitoring of QoS is likely to be most worthwhile, i.e. the areas that are most likely to be affected by any QoS problems. If stakeholders wish to examine other QoS aspects they are recommended to follow the general approach of the present document - as far as practicable - as a basis for the development of definitions and measurement methods for new
specific QoS parameters.
The set of QoS parameters is designed to be understood by the users of various telecommunications services. Sub-sets of these parameters can be selected for use in different circumstances. For example a specific parameter might be relevant for many users in some countries or markets but the same parameter might not be of relevance in others.
Therefore stakeholders - users, customers, regulators, service providers, network operators and other parties interested in the use of QoS parameters - should decide in co-operation, which parameters and which measures should be used in their particular situation. This decision should take account of:
• The precise purpose for which they will be used.
• The general level of quality achieved by most operators/providers.
• The degree to which the parameters will provide a reliable comparison of performance.
• The cost of measuring and reporting each parameter.

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The present document aims to identify and define testing methodologies which can be used to objectively evaluate the performance of narrowband and wideband terminals and systems for speech communication in the presence of background noise.
Background noise is a problem in mostly all situations and conditions and need to be taken into account in both, terminals and networks. The present document provides information about the testing methods applicable to objectively evaluate the speech quality in the presence of background noise. The present document includes:
• The description of the experts post evaluation process chosen to select the subjective test data being within the scope of the objective methods.
• The results of the performance evaluation of the currently existing methods described in ITU-T
Recommendation P.862 [i.16], [i.17] and in TOSQA2001 [i.19] which is chosen for the evaluation of terminals in the framework of ETSI VoIP speech quality test events [i.8], [i.9], [i.10] and [i.11].
• The method which is applicable to objectively determine the different parameters influencing the speech quality in the presence of background noise taking into account:
- the speech quality;
- the background noise transmission quality;
- the overall quality.
• The document is to be used in conjunction with:
- EG 202 396-1 [i.1] which describes a recording and reproduction setup for realistic simulation of
background noise scenarios in lab-type environments for the performance evaluation of terminals and communication systems.
- EG 202 396-2 [i.2] which describes the simulation of network impairments and how to simulate realistic transmission network scenarios and which contains the methodology and results of the subjective scoring for the data forming the basis of the present document.
- French speech sentences as defined in ITU-T Recommendation P.501 [i.13] for wideband and English speech sentences as defined in ITU-T Recommendation P.501 [i.13] for narrowband.

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The present document is part 1 of a series of documents on the specification and measurement of mouth-to-ear (also end-to-end) speech transmission quality. Its main objective is to describe objective comparison-based methods and systems for measuring mouth-to-ear speech quality in networks. Apart from this, it gives an overview on other important aspects of mouth-to-ear speech quality. As the need arises, these other aspects will be covered in more detail in subsequent parts of the present document. Although some of the models described in the present document are superseded the description of the models is kept for information. The present document gives an overview of the methods available for measuring one-way speech transmission quality.
Its purpose is to give information and guidance primarily for operators, users, consumer organizations and regulators who wish to measure or compare the speech transmission quality provided by different networks. The need for the present document has been increased by:
the liberalization of voice services, which has introduced alternative competing providers of voice services;
the introduction of new mobile and IP based technologies;
which has increased the range of services and cost/quality options for users.
The present document applies to both fixed and mobile networks with or without terminal equipment connected to the network. It applies only for narrowband (i.e. between 300 Hz and 3 400 Hz) communications. In principle, comparison methods can be used for IP-based (internet protocol-based) networks, but further work is needed on the calibration of the methods for such networks. The present document describes:
methods for measurements of individual impairments or combinations of impairments to be made at acoustic or electrical interfaces;
methods for combining measures of different impairments into a single objective measure;
methods for predicting the subjective effect of impairments that would be perceived by users.
The methods in the present document assume that subjects with normal hearing have been involved in the test. Therefore, the instrumental methods estimate the perceived speech quality of persons with normal hearing. For each method, the guide contains a general description to highlight the main points, and provides references for more detailed information. The present document does not contain detailed specifications of the individual methods. The present document concentrates on one-way speech quality in networks. It gives no guidance on how to evaluate systems that include equipment such as echo cancellers or in which interactive impairments such as talker echo are significant. The perceived quality in such cases depends not only on the one-way performance, but very much on the behaviour of the equipment under duplex conditions; specifically, the influence of double-talk and delay needs to be considered. Although all assessments of overall speech quality are ultimately subjective because they depend on the user's opinion, a distinction is made between:
subjective methods, which involve real time user assessment; and
objective methods, which use stored information on the user's assessment and therefore involve some degree of calibration. Objective methods for the evaluation of speech quality fall into three categories:
a) Comparison Methods: Methods based on the comparison of transmitted speech signal and a known reference.
b) Absolute Estimation Methods: Methods based on the absolute estimation of the speech quality (i.e. there is no known reference signal); e.g. INMD (ITU-T Recommendation P.561.
c) Transmission Rating Models: Methods that derive a value for the expected speech quality from knowledge about the network; e.g. ETSI Model ETR 250, ITU-T Recommendation G.107.

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The present document addresses mouth-to-ear (i.e. end-to-end speech quality for 3,1 kHz telephony). It both:
a) summarizes and gives guidance about the main factors that affect speech quality in end-to-end scenarios; and
b) specifies test methods for end-to-end speech quality testing.
The test methods can be used both for the complete transmission from mouth-to-ear and also for testing individual sections of a connection. The end-end (mouth-to-ear) test methods specified in the present document are independent of the technology used in the network and the terminals. However when practical considerations make it necessary to test at electrical interfaces within or between equipments the present document explains how to handle the most common current technologies. The present document is designed to be used by:
terminal and terminal component (e.g. soundcard) developers who wish to evaluate the end-to-end performance of networks and their terminals (or components); or
network designers who wish to evaluate the end-to-end performance of their networks with typical terminals. And therefore it gives advice on how networks and representative terminals (respectively) can be selected or simulated for use in the end-to-end tests. The test methods described allow the evaluation of all conversational situations such as single talk and double talk by means of objective procedures. The present document takes account of:
a) all types of terminals, including handsets, headsets and dedicated hands-free arrangements such as are provided with some mobile terminals and PC based terminals;
b) both circuit switched and packet based networks, including IP and ATM.
The present document is not generally suitable for wideband telephony or other forms of wideband communication although the parametric approach and the measurement procedures for some of the parameters described in the present document are applicable for wideband communication as well.

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The quality of background noise transmission is an important factor, which significantly contributes to the perceived overall quality of speech. Existing and even more the new generation of terminals, networks and system configurations including broadband services can be greatly improved with a proper design of terminals and systems in the presence of background noise. The present document:
- describes a noise simulation environment using realistic background noise scenarios for laboratory use;
- contains a database including the relevant background noise samples for subjective and objective evaluation. The present document provides information about the recording techniques needed for background noise recordings and discusses the advantages and drawbacks of existing methods. The present document describes the requirements for laboratory conditions. The loudspeaker setup and the loudspeaker calibration and equalization procedure are described. The simulation environment specified can be used for the evaluation and optimization of terminals and of complex configurations including terminals, networks and other configurations. The main application areas should be: office, home and car environment. The setup and database as described in the present document are applicable for:
- Objective performance evaluation of terminals in different (simulated) background noise environments.
- Speech processing evaluation by using the pre-processed speech signal in the presence of background noise, recorded by a terminal.
- Subjective evaluation of terminals by performing conversational tests, specific double talk tests or talking and listening tests in the presence of background noise.
- Subjective evaluation in third party listening tests by recording the speech samples of terminals in the presence of background noise.

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The present document specifies the technical characteristics (electrical and acoustic requirements and measurement methods) to be provided by a single, handset telephony, terminal equipment which is intended for connection by 2 wires to an analogue interface of a PSTN. This interface is characterized by a d.c. loop to indicate seizure and clearing, low frequency a.c. ringing signals below the speech passband to indicate an incoming call and the transmission phase having an approximate bandwidth of 3 kHz at the network terminating point. The objective of the present documen is to ensure minimum speech quality when interworking via the public network between two single items of equipment. The present document only applies to terminal equipment supporting handset telephony. The present document is applicable to handset telephony function. In the case of multiple functions provided in the same terminal equipment, the present document does not apply when those other functions are active in conjunction with handset telephony. The present document also applies to any type of analogue handset terminal intended to be connected to a gateway. The present document specifies the functions necessary to provide 2-way real-time speech conversation. Where a function is indicated as optional, it needs not to be provided, but, where such a function is provided, the terminal needs to conform to the requirements and tests specified in the present document. A test is given for each requirement in the present document including measurement methods. The terminal equipment may be stimulated to perform the tests by additional equipment if necessary. The present document gives requirements for new test methods based upon use of HATS and new tests signals In an annex, requirements with test methods corresponding to previous tests methods (LRGP) and test signals are given. The application of the present document is intended also for handset telephony function employing a radio link (e.g. DECT); The application of the present document is not intended for:
- a handset telephony function with speech transmission performance specially designed for the less abled (e.g. with amplification of received speech as an aid for the hard of hearing);
- a handset telephony function with speech transmission performance specifically designed to cater for hostile environments;
- any handsfree or loudspeaking voice telephony function;
- a handset telephony function employing speech processing techniques other than coding.

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The present document addresses combination network performance parameters and user perceived media (audio and video) quality parameters for audiovisual communications on IP networks. The access technologies covered include both wired (e.g. xDSL) and wireless (e.g. UMTS, WLAN) technologies. The display size range covered is from those of small mobile terminals (e.g. 2") up to large TV sets (e.g. 40" or more). It is applicable to:
- Broadcasting and streaming applications such as IPTV and VoD.
- Interactive point-to-pint applications such as videotelephony and videoconferencing.
Where the media coding standards define two or more profiles, the baseline profile is addressed in the normative part of the standard. Informative annexes present an overview of network QoS mechanisms and the effects on connection performance as well as guidance on terminal parameters that may influence the user perceived media performance.

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The present document contains definitions and measurement methods for a range of user perceivable Quality of Service (QoS) parameters. The purpose of these parameters is to define objective and comparable measures of the QoS delivered to users/customers for use by users/customers. The present document applies to any telecommunication service, however, some parameters may have a limited application. The present document is intended to provide a menu from which individual items can be selected. There is no obligation to use any or all of the parameters. The QoS parameters are related primarily to services and service features and not to the technology used to provide the services. Therefore the parameters should be capable of use when the services are provided on new technologies such as IP and ATM or other packet switched technologies as well as on circuit switched technologies. The establishment of target values for QoS is beyond the scope of the present document. The QoS parameters listed in the present document are also not intended to assess the complete QoS of a telecommunication service. The present document provides a set of QoS parameters that covers specific user related QoS aspects rather than a complete list of QoS parameters. This set has been chosen to address areas where monitoring of QoS is likely to be most worthwhile, i.e. the areas that are most likely to be affected by any QoS problems. If stakeholders wish to examine other QoS aspects they are recommended to follow the general approach of the present document - as far as practicable - as a basis for the development of definitions and measurement methods for new specific QoS parameters. The set of QoS parameters is designed to be understood by the users of various telecommunications services. Sub-sets of these parameters can be selected for use in different circumstances. For example a specific parameter might be relevant for many users in some countries or markets but the same parameter might not be of relevance in others. Therefore stakeholders - users, customers, regulators, service providers, network operators and other parties interested in the use of QoS parameters - should decide in co-operation, which parameters and which measures should be used in their particular situation. This decision should take account of:
- The precise purpose for which they will be used.
- The general level of quality achieved by most operators/providers.
- The degree to which the parameters will provide a reliable comparison of performance.
- The cost of measuring and reporting each parameter.

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The present document aims at identifying and defining indicators and methodologies for a use in a context of end-user quality characterization and supervision of voice telephony services. In this context the measurements and metric determinations are perform by analysing signals accessible on user-end services and not on the network. In order to mirror the reality in terms of access to the services at the user-end measurements and analysis are perform on electrical signal that exclude the electro-acoustic part of the end equipment but the probe adaptation to electric interface of the end user equipment much take into account the electro-acoustic characteristics of this terminal.

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ISO/IEC 22535:2009 specifies tunnelling of "QSIG" over the Session Initiation Protocol (SIP) within a corporate telecommunication network (CN). The tunnelling of QSIG through a public IP network employing SIP is outside the scope of ISO/IEC 22535:2009. However, the functionality specified in this International Standard is in principle applicable to such a scenario when deployed in conjunction with other relevant functionality (e.g. address translation, security functions, etc.). ISO/IEC 22535:2009 is applicable to any interworking unit that can act as a gateway between a PISN employing QSIG and a corporate IP network employing SIP, with QSIG tunnelled within SIP requests and responses.

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The present document contains definitions and measurement methods for a range of user perceivable Quality of Service (QoS) parameters. The purpose of these parameters is to define objective and comparable measures of the QoS delivered to users/customers for use by users/customers. The present document applies to any telecommunication service, however, some parameters may have a limited application. The present document is intended to provide a menu from which individual items can be selected. There is no obligation to use any or all of the parameters. The QoS parameters are related primarily to services and service features and not to the technology used to provide the services. Therefore the parameters should be capable of use when the services are provided on new technologies such as IP and ATM or other packet switched technologies as well as on circuit switched technologies. The establishment of target values for QoS is beyond the scope of the present document. The QoS parameters listed in the present document are also not intended to assess the complete QoS of a telecommunication service. The present document provides a set of QoS parameters that covers specific user related QoS aspects rather than a complete list of QoS parameters. This set has been chosen to address areas where monitoring of QoS is likely to be most worthwhile, i.e. the areas that are most likely to be affected by any QoS problems. If stakeholders wish to examine other QoS aspects they are recommended to follow the general approach of the present document - as far as practicable - as a basis for the development of definitions and measurement methods for new specific QoS parameters. The set of QoS parameters is designed to be understood by the users of various telecommunications services. Sub-sets of these parameters can be selected for use in different circumstances. For example a specific parameter might be relevant for many users in some countries or markets but the same parameter might not be of relevance in others. Therefore stakeholders - users, customers, regulators, service providers, network operators and other parties interested in the use of QoS parameters - should decide in co-operation, which parameters and which measures should be used in their particular situation. This decision should take account of:
- The precise purpose for which they will be used.
- The general level of quality achieved by most operators.
- The degree to which the parameters will provide a reliable comparison of performance.
- The cost of measuring and reporting each parameter.

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The present document specifies a simple set of objectives, principles and responsibilities for the transmission performance of multiple interconnected networks that provide "traditional quality (POTS)" circuit switched telephony services. The objectives, principles and responsibilities take account of the liberalization of telephony services and the interconnection of several separate networks, each with different topologies, in the provision of telephony connections. The present document applies to:
- national and international networks;
- Digital Networks and Integrated Digital Networks (i.e. networks where the only analogue component may be the local loop). The present document applies in the cases where the telephony service is:
- contracted between the network operator and the customer/end user; and
- contracted through a service provider.
It applies where:
- the caller pays for the call;
- the recipient pays for the call (e.g. the 800 service); or
- the cost of the call is shared by the caller and the recipient.
The present document does not apply to any segment of calls where either the calling or called terminal is a mobile "terminating network".

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The present document contains definitions and measurement methods for a range of user perceivable Quality of Service (QoS) parameters. The purpose of these parameters is to define objective and comparable measures of the QoS delivered to users/customers for use by users/customers. The present document applies to any telecommunication service however some parameters may have a limited application. The present document is intended to provide a menu from which individual items can be selected. There is no obligation to use any or all of the parameters. The QoS parameters are related primarily to services and service features and not to the technology used to provide the services. Therefore the parameters should be capable of use when the services are provided on new technologies such as IP and ATM or other packet switched technologies as well as on circuit switched technologies. The establishment of target values for QoS is outside the scope of the present document. The QoS parameters listed in the present document are also not intended to assess the complete QoS of a telecommunication service. The present document provides a set of QoS parameters that covers specific user related QoS aspects rather than a complete list of QoS parameters. This set has been chosen to address areas where monitoring of QoS is likely to be most worthwhile, i.e. the areas that are most likely to be affected by any QoS problems. If stakeholders wish to examine other QoS aspects they are recommended to follow the general approach of the present document - as far as practicable - as a basis for the development of definitions and measurement methods for new specific QoS parameters. The set of QoS parameters is designed to be understood by the users of various telecommunications services. Sub-sets of these parameters can be selected for use in different circumstances. For example a specific parameter might be relevant for many users in some countries or markets but the same parameter might not be of relevance in others. Therefore stakeholders - users, customers, regulators, service providers, network operators and other parties interested in the use of QoS parameters - should decide in co-operation, which parameters should be used in their particular situation. This decision should take account of:
- The precise purpose for which they will be used.
- The general level of quality achieved by most operators.
- The degree to which the parameters will provide a reliable comparison of performance.
- The cost of measuring and reporting each parameter.
The parameters defined in the present document are applicable to any kind of Internet access independently of the underlying technology.

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The present document aims to identify and define testing methodologies which can be used to objectively evaluate the performance of narrowband and wideband terminals and systems for speech communication in the presence of background noise. Background noise is a problem in mostly all situations and conditions and need to be taken into account in both, terminals and networks. The present document provides information about the testing methods applicable to objectively evaluate the speech quality in the presence of background noise. The present document includes:
-The description of the experts post evaluation process chosen to select the subjective test data being within the scope of the objective methods.
-The results of the performance evaluation of the currently existing methods described in ITU-T
Recommendation P.862 [i.16], [i.17] and in TOSQA2001 [i.19] which is chosen for the evaluation of terminals in the framework of ETSI VoIP speech quality test events [i.8], [i.9], [i.10] and [i.11].
-The method which is applicable to objectively determine the different parameters influencing the speech quality in the presence of background noise taking into account:
- the speech quality;
- the background noise transmission quality;
- the overall quality.
-The document is to be used in conjunction with:
- EG 202 396-1 [i.1] which describes a recording and reproduction setup for realistic simulation of background noise scenarios in lab-type environments for the performance evaluation of terminals and communication systems.
- EG 202 396-2 [i.2] which describes the simulation of network impairments and how to simulate realistic transmission network scenarios and which contains the methodology and results of the subjective scoring for the data forming the basis of the present document.
- French speech sentences as defined in ITU-T Recommendation P.501 [i.13] for wideband and English speech sentences as defined in ITU-T Recommendation P.501 [i.13] for narrowband.

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The quality of background noise transmission is an important factor, which significantly contributes to the perceived overall quality of speech. Existing and even more the new generation of terminals, networks and system configurations including broadband services can be greatly improved with a proper design of terminals and systems in the presence of background noise. The present document:
-describes a noise simulation environment using realistic background noise scenarios for laboratory use;
-contains a database including the relevant background noise samples for subjective and objective evaluation.
The present document provides information about the recording techniques needed for background noise recordings and discusses the advantages and drawbacks of existing methods. The present document describes the requirements for laboratory conditions. The loudspeaker setup and the loudspeaker calibration and equalization procedure are described. The simulation environment specified can be used for the evaluation and optimization of terminals and of complex configurations including terminals, networks and other configurations. The main application areas should be: office, home and car environment. The setup and database as described in the present document are applicable for:
- Objective performance evaluation of terminals in different (simulated) background noise environments.
- Speech processing evaluation by using the pre-processed speech signal in the presence of background noise, recorded by a terminal.
- Subjective evaluation of terminals by performing conversational tests, specific double talk tests or talking and listening tests in the presence of background noise.
- Subjective evaluation in third party listening tests by recording the speech samples of terminals in the presence of background noise.

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ISO/IEC 17343:2007 specifies signalling interworking between QSIG and the Session Initiation Protocol (SIP) in support of basic services within a corporate telecommunication network (CN) (also known as enterprise network). QSIG is a signalling protocol that operates between Private Integrated services Network eXchanges (PINX) within a Private Integrated Services Network (PISN). A PISN provides circuit-switched basic services and supplementary services to its users. QSIG is specified in other Standards. NOTE The name QSIG was derived from the fact that it is used for signalling at the Q reference point. The Q reference point is a point of demarcation between two PINXs. SIP is an application-layer protocol for establishing, terminating, and modifying multimedia sessions. It is typically carried over IP. Telephone calls are considered a type of multimedia session where just audio is exchanged. As the support of telephony within corporate networks evolves from circuit-switched technology to Internet technology, the two technologies will coexist in many networks for a period, perhaps several years. Therefore, there is a need to be able to establish, modify, and terminate sessions involving a participant in the SIP network and a participant in the QSIG network. Such calls are supported by gateways that perform interworking between SIP and QSIG. ISO/IEC 17343:2007 specifies SIP-QSIG signalling interworking for basic services that provide a bi-directional transfer capability for speech, DTMF, facsimile, and modem media between a PISN employing QSIG and a corporate IP network employing SIP. Other aspects of interworking, e.g., the use of RTP and SDP, will differ according to the type of media concerned and are outside the scope of ISO/IEC 17343:2007. Call-related and call-independent signalling in support of supplementary services is outside the scope of ISO/IEC 17343:2007, but support for certain supplementary services (e.g., call transfer, call diversion) could be the subject of future work. Interworking between QSIG and SIP permits a call originating at a user of a PISN to terminate at a user of a corporate IP network, or a call originating at a user of a corporate IP network to terminate at a user of a PISN. Interworking between a PISN employing QSIG and a public IP network employing SIP is outside the scope of ISO/IEC 17343:2007. However, the functionality specified in ISO/IEC 17343:2007 is in principle applicable to such a scenario when deployed in conjunction with other relevant functionality (e.g., number translation, security functions, etc.). ISO/IEC 17343:2007 is applicable to any interworking unit that can act as a gateway between a PISN employing QSIG and a corporate IP network employing SIP.

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The present document provides the ETSI endorsement of the ITU-T Bearer Independent Call Control (BICC) Capability Set 2 protocol Recommendations Q.1902.1 [1], Q.1902.2 [2], Q.1902.3 [3], Q.1902.4 [4], Q.1902.5 [5], Q.1902.6 [6], Q.765.5 [7] Amendment 1 [8],Q.1912.1 [9], Q.1912.2 [10], Q.1912.3 [11], Q.1912.4 [12], Q.1922.2 [13], Q.1950 [14], Q.1970 [15], Q.1990 [16], Q.2150.0 [17], Q.2150.1 [18], Q.2150.2 [19], Q.2150.3 [20]. Formats, codes and procedures marked for national use or as network option are included for informative purposes for the international interface specification. If these items so marked are supported within a national network and operator's network, then it is proposed that they shall be supported in this manner.

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ISO/IEC 23916:2005 specifies call transfer interworking between the Session Initiation Protocol (SIP) and "QSIG" within corporate telecommunication networks (CN), also known as enterprise networks. "QSIG" is a signalling protocol that operates between Private Integrated services Network eXchanges (PINX) within a Private Integrated Services Network (PISN). A PISN provides circuit-switched basic services and supplementary services to its users. SIP is an application layer protocol for establishing, terminating and modifying multimedia sessions. It is typically carried over IP. Telephone calls are considered as a type of multimedia session where just audio is exchanged. As the support of telephony within corporate networks evolves from circuit-switched technology to Internet technology, the two technologies will co-exist in many networks for a period, perhaps several years. Therefore, there is a need to be able to establish, modify, terminate and transfer sessions involving participants in the SIP network and participants in the QSIG network. Such calls are supported by gateways that perform interworking between SIP and QSIG. ISO/IEC 23916:2005 specifies SIP-QSIG signalling interworking for transfer services between a PISN employing QSIG and a corporate IP network employing SIP.

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ISO/IEC 23915:2005 specifies signalling interworking between "QSIG" and the Session Initiation Protocol (SIP) in support of call diversion within corporate telecommunication networks (CN), also known as enterprise networks. "QSIG" is a signalling protocol that operates between Private Integrated services Network eXchanges (PINX) within a Private Integrated Services Network (PISN). A PISN provides circuit-switched basic services and supplementary services to its users. SIP is an application layer protocol for establishing, terminating and modifying multimedia sessions. It is typically carried over IP. ISO/IEC 23915:2005 specifies signalling interworking for call diversion during the establishment of calls between a PISN employing QSIG and a corporate IP network employing SIP. It covers both the impact on SIP of call diversion in the QSIG network and the impact on QSIG of request retargeting in the SIP network. ISO/IEC 23915:2005 is applicable to any interworking unit that can act as a gateway between a PISN employing QSIG and a corporate IP network employing SIP.

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This ETS addresses the configuration management area, covering the following aspects: -  a management architecture that shows how the X-interface is to be used between network and service providers; -  the management services and functions needed to manage ATM connections, which span over several administrative domains. These management services and functions cover the configuration management requirements for the X-interface; -  the management information crossing the X-interface (using GDMO formalisms as described in ITU-T Recommendatio X.722).

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Contains stage 3 description of the calI inter. supplementary service. See ECMA DTR/ECMA-00002 (ECMA-TR/SVC) for a description of the call int supplementary service. See ENV 41 005 for definition of stages.

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This European Telecommunication Standard (ETS) specifies the signalling protocol for the support of Call Diversion supplementary services (SS-DIV) at the Q reference point between Private Telecommunication Network eXchanges (PTNX) connected together within a Private Telecommunication Network (PTN). The Call Diversion supplementary services are Calling Forwarding Unconditional (SS-CFU), Call Forwarding Busy (SS-CFB), Call Forwarding No Reply (SS-CFNR) and Call Deflection (SS-CD).

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Contains stage 1 and 2 description of the explicit call transfer supplementary service. See ECMA DTR/ECMA-0002 (ECMA-TR/SVC) for a description of the explict call transfer supplementary service. See ENV 41 005 for definition of stages 1 and 2.

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This European Telecommunication Standard (ETS) defines the signalling protocol for the control of Supplementary Services and Additional Network Features (ANF) at the Q reference point. The protocol is part of QSIG. The Q reference point exists between Private Telecommunication Network eXchanges (PTNX) connected together within a Private Telecommunication Network (PTN). The Q reference point is defined in ETS 300 475-1. Detailed procedures applicable to individual Supplementary Services and ANFs are beyond the scope of this ETS and will be specified by other standards for those services which are standardised and by individual manufacturers for proprietary services using the capabilities defined in this ETS.

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This European Normes (EN) contains the specification, functional model and information flows for the Call priority interruption supplementar

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Contains stages 1 and stage 2 description of the intrusion supplementary service. See ECMA DTR/ECMA-0002 (ECMA-TR/SVC) for a description of the intrusion supplementary service. See ENv 41 005 for definition of stages 1 and 2.

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Conformance Test Specification (TTCN) for the protocol for layer 3 signalling between exchanges of private integrated services networks for the control of circuit-switched calls.

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This enropean Telecommunication Standard (ETS) specifies the signalling protocol for the support of the Call completion supplementary services (SS-CC) at the Q reference point bet-ween Private Telecommunication Network eXchanges (PTNX) connected together within a Private Telecommunication Network (PTN). The Call completion supplementary services are Call Completion on Busy Subscriber (SS-CCBS) and Call Completion on No Reply (SS-CCNR).

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This European Telecommunication Standard (ETS) specifies the signalling protocol for the support of the Path Replacement additional network feature (ANF-PR) at the Q reference point between Private Telecommunication Network eXchanges (PTNX) connected together within a Private Telecommunication Network (PTN).

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Contains stage 1 and 2 description of the Call interception additional network feature. See DTR/ECMA-00002 (ECMA-TR/SVC) for a description of the call interception additional network feature. See ENV 41 005 for definition of stage 1 and 2.

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This European Telecommunication Standard (ETS) specifies the Path Replacement additional network feature (ANF-PR), which is applicable to various basic services supported by Private Integrated Services Networks (PISN).

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This standard defines the Inter-Exchange Signalling Protocol for the support of Private User Mobility - Call Handling (PUM-CH) in Private Integrated Services Networks (PISNs) for incoming and outgoing calls.

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Stage 3 of call forwarding uncond/no-reply/busy sup services at Q ref pt. See DTR/ECMA-0002 (ECMA-TR/SVC) for service description. See ENV41005 for definition of stage 3. See ENV41004 for definition of Q reference point. To contain PICS proforma.

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Contains stage 1 and stage 2 description of the (CFU) sup service (CFNR) sup service and (CFB) supplementary service. See ECMA DTR/ECMA-0002 (ECMA-TR/SVC) for a description of these supplementary services. See ENV 41 005 for definition of stage 1 and 2

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Reference document identifying usage & descriptions of address info used in private telecoms networks.

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Stage 3 of basic call at Q ref pt. See ENV41005 for definition of stage 3. See ENV41004 for definition of Q reference point. Second edition to contain PICS proforma. (Not PICS)

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Contains stage 1 and 2 description of the do not disturb supplementary service and the do not disturb override supplementary service. See ECMA DTR/ECMA-0002 (ECMA-TR/SVC) for a description of these supplementary services. See ENV 41 005 for definition of stage 1 and 2

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This European Telecommunication Standard (ETS) specifies the signalling protocol for the support of the Call transfer supplementary service (SS-CT) at the Q reference point between Private Telecommunication Network eXchanges (PTNX) connected together within a Private Telecommunication Network (PTN).

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