ETSI TS 103 222-1 V1.3.1 (2019-03)

Speech and multimedia Transmission Quality (STQ); Reference benchmarking, background traffic profiles and KPIs; Part 1: Reference benchmarking, background traffic profiles and KPIs for VoIP and FoIP in fixed networks

RTS/STQ-280-1

General Information

• Status: Published
• Publication Date: 14-Mar-2019
• Technical Committee: STQ - Speech and multimedia Transmission Quality
• Current Stage: 12 - Completion
• Due Date: 26-Mar-2019
• Completion Date: 15-Mar-2019

Standards Content (Sample)

TECHNICAL SPECIFICATION
Speech and multimedia Transmission Quality (STQ);
Reference benchmarking,
background traffic profiles and KPIs;
Part 1: Reference benchmarking, background traffic profiles
and KPIs for VoIP and FoIP in fixed networks

2 ETSI TS 103 222-1 V1.3.1 (2019-03)

Reference
RTS/STQ-280-1
Keywords
KPI, QoS
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3 ETSI TS 103 222-1 V1.3.1 (2019-03)
Contents
Intellectual Property Rights . 5
Foreword . 5
Modal verbs terminology . 5
Introduction . 5
1 Scope . 6
2 References . 6
2.1 Normative references . 6
2.2 Informative references . 7
3 Definition of terms, symbols and abbreviations . 8
3.1 Terms . 8
3.2 Symbols . 8
3.3 Abbreviations . 8
4 Management Summary. 10
4.1 Introduction . 10
4.2 Scope of functionality . 10
5 Technical concept . 11
5.1 Voice over IP . 11
5.2 Call set-up delay and Session initiation call set-up delay . 12
5.3 Call set-up time (post dialling delay) . 14
5.4 Premature release probability (call failure rate; telephony service non accessibility) . 16
5.5 Telephony Cut-off Call Ratio [%] (Call drop rate) . 16
5.6 Media establishment delay . 17
5.7 Level of active speech signal in receive direction . 17
5.8 Noise level in receive direction . 18
5.9 Signal to noise ratio in receive direction . 18
5.10 Speech signal attenuation (or gain) . 18
5.11 Talker echo delay . 18
5.12 Double talk performance . 18
5.13 Interrupted voice transmission . 19
5.14 Listening speech quality . 20
5.14.1 General aspects of Listening Speech Quality. 20
5.14.2 General aspects of voice channel test calls . 20
5.14.3 Connections without parallel data transfer . 22
5.14.3.1 Connections with one voice channel . 22
5.14.3.2 Multiple voice channel access . 24
5.14.4 Connections with parallel data transfer . 27
5.14.4.0 Introduction . 27
5.14.4.1 Quality measurement of one voice channel and parallel data transfer . 27
5.14.4.2 Parallel quality measurement of one voice channel and data transmission speed . 31
5.14.4.3 Quality measurement of multiple voice channels and data transfer . 33
5.14.4.4 Parallel quality measurement of multiple voice channels and data transmission speed . 34
5.15 Listening speech quality stability . 37
5.16 End-to-end audio delay . 37
5.17 End -to-end audio delay variation . 38
5.18 Frequency response in receive direction . 39
5.19 Fax transmission with Recommendations ITU-T T.30 and T.38 . 39
5.19.1 General considerations . 39
5.19.2 Fax set-up duration . 41
5.19.3 Fax transmission duration . 41
5.19.4 Fax failure ratio . 42
5.19.5 Test case descriptions . 42
5.19.5.1 Quality measurement of one fax channel . 42
5.19.5.2 Quality measurement of one fax channel and parallel data transfer . 43
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5.20 Early media listening speech quality . 50
5.20.1 Introduction. 50
5.20.2 Early media generated by the called party . 50
5.21 Jitter Buffer and IP prioritization response time. 51
5.21.1 Jitter Buffer and IP prioritization response time without data transfer . 51
5.21.2 Jitter Buffer and IP prioritization response time with data transfer . 52
5.22 Stability of the de-jitter buffer delay adjustments for VBD calls during IP data transfer. 53
6 Internet related load generation methods for VoIP and FoIP tests . 56
6.1 Introduction to Load generation alternatives . 56
6.2 TCP load generation with flow-control enabled . 56
6.3 TCP load generation with no flow-control . 56
6.4 Monitoring the generated load . 57
6.5 QoS IP transmission layer parameter tests . 57
6.5.0 Introduction. 57
6.5.1 VoIP (Voice over IP) . 57
6.5.2 TCP Parameters . 58
6.5.3 UDP Parameters . 58
6.5.4 Web Page . 59
6.5.5 Unmodified content . 59
6.5.6 Transparent connection . 59
6.5.7 DNS . 60
6.5.8 Traceroute . 60
Annex A (informative): Bibliography . 61
History . 62

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5 ETSI TS 103 222-1 V1.3.1 (2019-03)
Intellectual Property Rights
Essential patents
IPRs essential or potentially essential to normative deliverables may have been declared to ETSI. The information
pertaining to these essential IPRs, if any, is publicly available for ETSI members and non-members, and can be found
in ETSI SR 000 314: "Intellectual Property Rights (IPRs); Essential, or potentially Essential, IPRs notified to ETSI in
respect of ETSI standards", which is available from the ETSI Secretariat. Latest updates are available on the ETSI Web
server (https://ipr.etsi.org/).
Pursuant to the ETSI IPR Policy, no investigation, including IPR searches, has been carried out by ETSI. No guarantee
can be given as to the existence of other IPRs not referenced in ETSI SR 000 314 (or the updates on the ETSI Web
server) which are, or may be, or may become, essential to the present document.
Trademarks
The present document may include trademarks and/or tradenames which are asserted and/or registered by their owners.
ETSI claims no ownership of these except for any which are indicated as being the property of ETSI, and conveys no
right to use or reproduce any trademark and/or tradename. Mention of those trademarks in the present document does
not constitute an endorsement by ETSI of products, services or organizations associated with those trademarks.
Foreword
This Technical Specification (TS) has been produced by ETSI Technical Committee Speech and multimedia
Transmission Quality (STQ).
The present document is part 1 of a multi-part deliverable covering the Reference benchmarking, background traffic
profiles and KPIs, as identified below:
Part 1: "Reference benchmarking, background traffic profiles and KPIs for VoIP and FoIP in fixed
networks";
Part 2: "Reference benchmarking and KPIs for High speed internet";
Part 3: "Reference benchmarking, background traffic profiles and KPIs for UMTS and VoLTE";
Part 4: "Reference benchmarking for IPTV, Web TV and RCS-e Video Share".
Modal verbs terminology
In the present document "shall", "shall not", "should", "should not", "may", "need not", "will", "will not", "can" and
"cannot" are to be interpreted as described in clause 3.2 of the ETSI Drafting Rules (Verbal forms for the expression of
provisions).
"must" and "must not" are NOT allowed in ETSI deliverables except when used in direct citation.
Introduction
The present document describes possible key performance indicators for VoIP and FoIP as well as framework
requirements for reference benchmarking particularly with regard to background traffic. The latest version replaces
clause 6 with load generation methods that are appropriate for the agreed scope of the present document.

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1 Scope
The present document:
• identifies and defines possible key performance indicators for voice and fax telephony services;
• defines benchmarking methods for the spectrum of potential applications.
The offer of new NGN services requires new KPIs, QoS measurement and benchmarking methods which are needed to
ensure the quality of new services. To ensure the comparability of test results, reference benchmarking methods and
background traffic load profiles are needed.
The scope of the defined testing procedures is the evaluation of the network access by VoIP and FoIP fixed-network
services. The measurements are conducted stationary between a subscriber access-point to a measurement point
emulating an idealized termination point in the core network. All access technologies offered by the operator under test
are considered. In this context the measurements and key performance indicators determinations are performed by
analysing signals accessible on the network.
The present document is the first part of the multi-part deliverable which consists of four parts.
The present document contains possible KPIs for VoIP and FoIP as well as framework requirements for reference
benchmarking particularly with regard to background traffic profiles.
2 References
2.1 Normative references
References are either specific (identified by date of publication and/or edition number or version number) or
non-specific. For specific references, only the cited version applies. For non-specific references, the latest version of the
referenced document (including any amendments) applies.
Referenced documents which are not found to be publicly available in the expected location might be found at
https://docbox.etsi.org/Reference/.
NOTE: While any hyperlinks included in this clause were valid at the time of publication, ETSI cannot guarantee
their long term validity.
The following referenced documents are necessary for the application of the present document.
[1] Recommendation ITU-T E.800 (09-2008): "Definitions of terms related to quality of service".
[2] Recommendation ITU-T P.863 (03-2018): "Perceptual objective listening quality prediction".
[3] ETSI TS 101 563 (V1.3.1): "Speech and multimedia Transmission Quality (STQ);
IMS/PES/VoLTE exchange performance requirements".
[4] Recommendation ITU-T Q.543 (03-1993): "Digital exchange performance design objectives".
[5] ETSI ES 202 765-2 (V1.2.1): "Speech and multimedia Transmission Quality (STQ); QoS and
network performance metrics and measurement methods; Part 2: Transmission Quality Indicator
combining Voice Quality Metrics".
[6] Recommendation ITU-T G.131 (11-2003): "Talker echo and its control".
[7] ETSI ES 203 021-3 (V2.1.2): "Access and Terminals (AT); Harmonized basic attachment
requirements for Terminals for connection to analogue interfaces of the Telephone Networks;
Update of the technical contents of TBR 021, EN 301 437, TBR 015, TBR 017; Part 3: Basic
Interworking with the Public Telephone Networks".
[8] ETSI TBR 003 (Edition 1) (11-1995): "Integrated Services Digital Network (ISDN); Attachment
requirements for terminal equipment to connect to an ISDN using ISDN basic access".
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7 ETSI TS 103 222-1 V1.3.1 (2019-03)
[9] ETSI TBR 004 (Edition 1) (11-1995): "Integrated Services Digital Network (ISDN); Attachment
requirements for terminal equipment to connect to an ISDN using ISDN primary rate access".
[10] ETSI EN 300 175-8: "Digital Enhanced Cordless Telecommunications (DECT); Common
Interface (CI); Part 8: Speech and audio coding and transmission".
[11] Recommendation ITU-T O.41 (10-1994): "Psophometer for use on telephone-type circuits".
[12] Recommendation ITU-T P.56 (12-2011): "Objective measurement of active speech level".
[13] Recommendation ITU-T P.501 (03-2017): "Test signals for use in telephonometry".
[14] ETSI ES 202 737 (V1.4.1): "Speech and multimedia Transmission Quality (STQ); Transmission
requirements for narrowband VoIP terminals (handset and headset) from a QoS perspective as
perceived by the user".
[15] ETSI ES 202 739 (V1.4.1): "Speech and multimedia Transmission Quality (STQ); Transmission
requirements for wideband VoIP terminals (handset and headset) from a QoS perspective as
perceived by the user".
[16] Recommendation ITU-T P.340 (05-2000): "Transmission characteristics and speech quality
parameters of hands-free terminals".
[17] Recommendation ITU-T P.502 (05-2000): "Objective test methods for speech communication
systems using complex test signals".
[18] Recommendation ITU-T P.863.1 (09-2014): "Application guide for Recommendation ITU-T
P.863".
[19] Recommendation ITU-T E.458 (02-1996): "Figure of merit for facsimile transmission
performance".
[20] Recommendation ITU-T E.453 (08-1994): "Facsimile image quality as corrupted by transmission-
induced scan line errors".
[21] ETSI TS 102 250-2 (V2.3.1): "Speech and multimedia Transmission Quality (STQ); QoS aspects
for popular services in mobile networks; Part 2: Definition of Quality of Service parameters and
their computation".
[22] Recommendation ITU-T Y.1541: "Network performance objectives for IP-based services".
[23] Recommendation ITU-T G.711: "Pulse code modulation (PCM) of voice frequencies".
[24] Recommendation ITU-T V.34: "A modem operating at data signalling rates of up to 33 600 bit/s
for use on the general switched telephone network and on leased point-to-point 2-wire telephone-
type circuits".
[25] Recommendation ITU-T V.17: "A 2-wire modem for facsimile applications with rates up to
14 400 bit/s".
2.2 Informative references
References are either specific (identified by date of publication and/or edition number or version number) or
non-specific. For specific references, only the cited version applies. For non-specific references, the latest version of the
referenced document (including any amendments) applies.
NOTE: While any hyperlinks included in this clause were valid at the time of publication, ETSI cannot guarantee
their long term validity.
The following referenced documents are not necessary for the application of the present document but they assist the
user with regard to a particular subject area.
[i.1] Void.
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[i.2] ETSI ETR 138 (12-1997): "Network Aspects (NA); Quality of service indicators for Open
Network Provision (ONP) of voice telephony and Integrated Services Digital Network (ISDN)".
[i.3] ETSI EG 202 425 (V1.1.1): "Speech Processing, Transmission and Quality Aspects (STQ);
Definition and implementation of VoIP reference point".
[i.4] ETSI EG 202 057-2: "Speech and multimedia Transmission Quality (STQ); User related QoS
parameter definitions and measurements; Part 2: Voice telephony, Group 3 fax, modem data
services and SMS".
[i.5] ETSI TR 103 138: "Speech and multimedia Transmission Quality (STQ); Speech samples and
their use for QoS testing".
[i.6] IEC 61260:1995: "Electroacoustics - Octave-band and fractional-octave-band filters".
[i.7] Recommendation ITU-T T.30 (09-2005): "Procedures for document facsimile transmission in the
general switched telephone network".
[i.8] Recommendation ITU-T T.38 (09-2010): "Procedures for real-time Group 3 facsimile
communication over IP networks".
[i.9] Recommendation ITU-T T.24: "Standardized digitized image set".
[i.10] Recommendation ITU-T G.168 (04-2015): "Digital network echo cancellers".
[i.11] Recommendation ITU-T V.25: "Automatic answering equipment and general procedures for
automatic calling equipment on the general switched telephone network including procedures for
disabling of echo control devices for both manually and automatically established calls".
[i.12] IETF RFC 8337 (March 2018): "Model-Based Metrics for Bulk Transport Capacity", M.Mathis
and A.Morton.
NOTE: Available from https://tools.ietf.org/html/rfc8337.
[i.13] IETF RFC 4122 (July 2005): "A Universally Unique IDentifier (UUID) URN Namespace".
3 Definition of terms, symbols and abbreviations
3.1 Terms
For the purposes of the present document, the following terms apply:
benchmark: evaluation of performance value/s of a parameter or set of parameters for the purpose of establishing
value/s as the norm against which future performance achievements may be compared or assessed
NOTE: The definition is taken from Recommendation ITU-T E.800 [1].
3.2 Symbols
Void.
3.3 Abbreviations
For the purposes of the present document, the following abbreviations apply:
AB Direction of call establishment User B to user A
ACK ACKnowledgement
AGCF Access Gateway Control Function
AGW Access GateWay
AIMD Additive Increase, Multiplicative Decrease
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ANS ANswer Tone
AS Application Server
BA Direction of call establishment User A to user B
BRI Basic Rate Interface
BST Broadband Speed Test
CED CallED station identification tone
CFR Call Failure Rate
CI Common Interface
CLI Calling Line Identification
CNG CalliNG tone
CPE Customer Premises Equipment
CSCF Call Session Control Function
CSS Composite Source Signal
DL DownLink
DNS Domain Name System
DSS1 Digital subscriber Signalling System No. 1
DTMF Dual-Tone Multi-Frequency signalling
ECM Error Correction Mode
ERP Ear Reference Point
FE Functional Entity
FM Feature Manager
FoIP Fax over IP
FOM Figure Of Merit
FTP File Transfer Protocol
HTTP Hypertext Transfer Protocol
HTTPS Hypertext Transfer Protocol Secure
IAD Integrated Access Device
IEC International Electrotechnical Commission
IETF Internet Engineering Task Force
IMAP Internet Message Access Protocol
IMAPS Internet Message Access Protocol Secure
IMS Internet Multimedia Subsystem
IP Internet Protocol
IPTV Internet Protocol TeleVision
ISDN Integrated Services Digital Network
ITU-T International Telecommunication Union - Telecommunication standardization sector
IVR Interactive Voice Response
KPI Key Performance Indicator
LQO Listening Quality Objective
MGC Media Gateway Controller
MGW Media GateWay
MMTel MultiMedia Telephony service
MOS Mean Opinion Score
MRP Mouth Reference Point
MSAN Multi-Service Access Nodes
NGN New Generation Network
NTP Network Time Protocol
OVL OVerLoad point
PCMA Pulse Code Modulation A-law
P-CSCF Proxy - Call Session Control Function
PES PSTN Emulation Subsystem
PRI Primary Rate Interface
PSTN Public Switched Telephone Network
PT ProTocol
QoS Quality of Service
RDP Remote Desktop Protocol
RFC Request For Comments
RTCP Real Time Control Protocol
RTP Real Time Protocol
RTSP Real Time Streaming Protocol
S-CSCF Service - Call Session Control Function
SDP Session Description Protocol
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SIP Session Initiation Protocol
SMTP Simple Mail Transfer Protocol
SMTPS Simple Mail Transfer Protocol Secure
SNR Speech signal level/Noise level
SSH Secure SHell
SSL Secure Sockets Layer
SWB Super Wide Band
TCP Transmission Control Protocol
TELR Talker Echo Loudness Rating
TLS Transport Layer Security
TOR Terminal Owning Region
TR Technical Report
TV TeleVision
UA User Agent
UAS User Agent Server
UDP User Datagram Protocol
UE User Equipment
UL UpLink
UMTS Universal Mobile Telecommunications System
UNiA User Network interface A
UNiB User Network interface B
UUID Universally Unique IDentifier
NOTE: As described in IETF RFC 4122 [i.13].
VBD Voice Band Data
VGW Voice GateWay
VoIP Voice over IP
VoLTE Voice over Long Term Evolution
4 Management Summary
4.1 Introduction
The spectrum of potential applications of a benchmarking platform requires measurements including but not limited to
the following: analogue (a/b), ISDN, VoIP (including SIP trunking) and high-speed internet.
The performance data which are collected will be relevant for a real-world environment encompassing a mix of
technologies. The scope of the defined testing procedures is the evaluation of the network access by VoIP and FoIP
fix-network services. The measurements are conducted stationary between a subscriber access-point to a measurement
point emulating an idealized termination point in the core network.
4.2 Scope of functionality
A benchmarking platform can be distributed across a larger region or an entire country. In this case several server
systems should be also part of the setup, including: a business intelligence platform; a data warehouse, a management
system and a system for evaluating of media (e.g. video, audio and voice) quality.
The measurement systems at the user premises are connected electrically to ISDN ports via a VGW (IAD) or directly to
a CPE or Ethernet port (e.g. MMTel fixed access).
The test system (QoS control and data server) is connected through ISDN connections (via IMS PES with AGCF (or
PSTN or ISDN Access) or IMS PES with VGW) or MMTel (IMS) fixed access lines for voice quality measurements.
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Figure 1: Setup of the benchmarking platform
5 Technical concept
5.1 Voice over IP
The conduction of voice quality measurements is following the descriptions that can be found in ETSI
EG 202 057-2 [i.4], Recommendation ITU-T Q.543 [4], ETSI TS 101 563 [3] and ETSI TS 102 250-2 [21],
clauses 6.6.3.1 and 6.6.3.2.
The access points of the test equipment which are used for inserting or retrieving the signals needed for determining the
speech quality parameters shall conform to the reference characteristics as laid down in the following relevant
standards:
• ETSI EG 202 425 [i.3] for VoIP access;
• ETSI ES 203 021-3 [7] for analogue access;
• ETSI TBR 003 [8] for ISDN BRI access;
• ETSI TBR 004 [9] for ISDN PRI access;
• ETSI EN 300 175-8 [10].
The properties of the test equipment shall be known and the values measured for each parameter shall be corrected
accordingly by the impairments introduced by the test equipment. Especially any delay introduced by the test equipment
shall be known and the measurement results shall be corrected by the delay introduced by the test equipment.
The simultaneous transmission of voice and data through uploads, downloads or IPTV use is an additional user related
scenario. For this reason voice quality measurements have been included where in parallel to the voice connection
active upload and download of data is simulated. This provides information about any potential prioritization of voice
data when the entire bandwidth is being utilized.
The KPI listed in table 1 are recorded as part of the voice quality measurements.
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Table 1: Overview of KPI for voice quality measurements
1. Call set-up delay [4] and session initiation call set-up delay [3]
2. Call set-up time (Post Dialling Delay) [5]
3. Premature release probability (Call Failure Rate), see clause 5.4
4. Telephony Cut-off Call Ratio [%] (Call drop rate), see clause 5.5
5. Media establishment delay, see clause 5.6
6. Level of active speech signal, see clause 5.7
7. Noise level, see clause 5.8
8. Signal to Noise ratio, see clause 5.9
9. Speech signal attenuation, see clause 5.10
10. Talker echo delay, see clause 5.11
11. Double talk, see clause 5.12
12. Interrupted voice transmission, see clause 5.13
13. Listening speech quality, see clause 5.14
14. Listening speech quality stability, see clause 5.15
15. End-to-end audio delay, see clause 5.16
16. End-to-end audio delay variation, see clause 5.17
17. Frequency response, see clause 5.18
18. Fax transmission T.30 (Fax, bit rate ≤ 14,4 kbit/s and Fax, bit rate ≥ 14,4 kbit/s), see clause 5.19
19. Early media, see clause 5.20
20. Jitter Buffer and IP periodization response time, see clause 5.21

5.2 Call set-up delay and Session initiation call set-up delay
The testing methodology for the call set-up delay is described in ETSI TS 101 563 [3].
Call set-up delay is defined as the interval from the instant when the signalling information required for outgoing circuit
selection is received from the incoming signalling system until the instant when the corresponding signalling
information is passed to the outgoing signalling system.
For SIP (e.g. SIP Trunking, IMS) Session initiation call set-up delay is defined as the interval from the instant when the
INVITE signalling information is received from the calling user on the originating Gm interface until the instant when
the corresponding INVITE signalling information is passed on the terminating Gm interface to the called user.
Figure 2 depicts some of the call set up measurement options between AGCF/VGW and the Gm Interface.
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Figure 2: Call set up delay and Session initiation call set-up delay: en-bloc sending is used
Table 2 gives an overview of the call set-up delay configuration options.
Table 2: Call set-up delay configurations
From To
MMTel (IMS) fixed access MMTel (IMS) fixed access
MMTel (IMS) fixed access IMS PES with AGW (PSTN or ISDN Access)
MMTel (IMS) fixed access IMS PES with VGW
Call set up delay and
IMS PES with AGW (PSTN or ISDN Access) MMTel (IMS) fixed access
Session initiation call
IMS PES with AGW (PSTN or ISDN Access) IMS PES with AGW(PSTN or ISDN Access)
set-up delay
IMS PES with AGW (PSTN or ISDN Access) IMS PES with VGW
IMS PES with VGW IMS PES with VGW
IMS PES with VGW IMS PES with AGW (PSTN or ISDN Access)
IMS PES with VGW IMS PES with VGW
NOTE: The Call set-up delay values are specified in ETSI TS 101 563 [3].

Figure 3 illustrates the session processing model used by the AGCF and VGW functional entities.
An AGCF is modelled as comprising H.248 Media Gateway Controller (MGC), Feature Manager (FM), and SIP UA
functionality. An AGCF interfaces to a Media Gateway (MGW) and also to the S-CSCF (via P1 and Mw reference
points respectively).
A functional modelling of the VGW contains an entity similar to H.248 Media Gateway Controller, a Feature Manager,
a SIP UA, and MGW functionality. The VGW interfaces to the P-CSCF using the Gm reference point.
The SIP UA functionality provides the interface to the other components of the IMS-based architecture. It is involved in
registration and session processing as well as in event subscription/notification procedures with application servers.
The MGC functionality enables the session processing functionality to interface with existing line signalling such as
analogue signalling or DSS1.
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14 ETSI TS 103 222-1 V1.3.1 (2019-03)
Session and registration processing in the AGCF or VGW involves two halves: H.248 based MGC processing and SIP
User Agent (UA) processing (see figure 3). MGC processing focuses on the interactions with the media gateway
functions, while SIP UA processing focuses on the interactions with the IMS components. The Feature Manager (FM)
coordinates the two processing activities.

Figure 3: AGCF/VGW session processing models
5.3 Call set-up time (post dialling delay)
Call set-up time: the period starting when the address information required for setting up a call is received by the
network (e.g. recognized on the calling user's access line) and finishing when the called party busy tone or ringing tone
or answer signal is received by the calling party (e.g. recognized on the calling user's access line) (see ETSI
ETR 138 [i.2]).
To determine the call setup time in an ISDN implementation, the time in seconds from the sending of the DSS1 SETUP
signal through the "A" side (calling number + "Sending complete" information) until the receipt of the DSS1
CONNECT signal is measured on the "A" side is measured, or the time in seconds from the sending of the DSS1
SETUP signal through the "A" side (calling number + "Sending complete" information) until the receipt of the DSS1
ALERTING signal is measured on the "A" side is measured. In figures 4 and 5, this time is indicated by the green
arrow.
For ANALOGUE SUBSCRIBER LINES the Post Dialling Delay shall be used. It is the time interval between the end
of dialling by the caller and the reception back by him of the appropriate ringing tone or recorded announcement.
To determine the call setup time in a VoIP implementation, the time in seconds from the sending of the INVITE signal
through the "A" side until the receipt of the 200 OK signal is measured on the "A" side is measured, or the time in
seconds from the sending of the INVITE signal through the "A" side until the receipt of the 180 Ringing signal on the
"A" side is recorded. In figures 6 and 7, this time is indicated by the grey arrow.
Table 3 gives an overview of the call set-up time configurations options.
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15 ETSI TS 103 222-1 V1.3.1 (2019-03)
Table 3: Call set-up time configurations
From To
MMTel (IMS) fixed access MMTel (IMS) fixed access
MMTel (IMS) fixed access IMS PES with AGW (PSTN or ISDN Access)
MMTel (IMS) fixed access IMS PES with VGW
IMS PES with AGW (PSTN or ISDN Access) MMTel (IMS) fixed access
Call set up time
IMS PES with AGW (PSTN or ISDN Access) IMS PES with AGW(PSTN or ISDN Access)
IMS PES with AGW (PSTN or ISDN Access) IMS PES with VGW
IMS PES with VGW IMS PES with VGW
IMS PES with VGW IMS PES with AGW (PSTN or ISDN Access)
IMS PES with VGW IMS PES with VGW

Figure 4: Measurement of the call setup duration, option A with CONNECT

Figure 5: Measurement of the call setup duration, option B with ALERTING
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16 ETSI TS 103 222-1 V1.3.1 (2019-03)

Figure 6: VoIP Measurement of the call setup duration, option A with 200 OK

Figure 7: VoIP Measurement of the call setup duration, option B with 180 Ringing
5.4 Premature release probability (call failure rate; telephony
service non accessibility)
Premature release probability call failure rate (CFR) is based on the measurement of the number of released
communications in comparison with the number of established communications. Released communications are defined
as communications released before voluntary action from one of the ends of the transmission. See ETSI
TS 102 250-2 [21] for the formula.
5.5 Telephony Cut-off Call Ratio [%] (Call drop rate)
The Cut-off Call Ratio (Call drop rate) is the percentage of number of calls that are dropped after connection to the
system or network has been established. See ETSI TS 102 250-2 [21] for the formula.
In an ISDN implementation a call is completely established with the Connect message [21] and is considered dropped if
the call is not ended intentionally. See figure 8 for details.
In a SIP implementation a call is completely established with the arrival of the INVITE 200 OK on the caller side and is
considered dropped if the call is not terminated intentionally. See figure 9 for details.
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17 ETSI TS 103 222-1 V1.3.1 (2019-03)

Figure 8: Determination of the call drop ratio

Figure 9: VoIP: determination of the call drop ratio
5.6 Media establishment delay
The Media establishment delay determines on one of the two access of the communication, between off hock of the
called and the beginning of voice signal receive. The detailed testing method is described in ETSI ES 202 765-2 [5].
5.7 Level of active speech signal in receive direction
A typical method for the measurement of this parameter, based on a sample by sample approach and a moving threshold
between noise and speech, is given in Recommendation ITU-T P.56 [12].
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18 ETSI TS 103 222-1 V1.3.1 (2019-03)
5.8 Noise level in receive direction
Level of noise determined in receive direction in the non-speech segments of a speech sample. For the actual
measurement the noise in between speech signals (idle noise) is analysed. The analysis window length needs to be
adapted accordingly.
The noise level is measured in the frequency range from 100 Hz to 4 kHz in narrowband and from 100 Hz to 8 kHz in
wideband. The analysis window is applied directly to the end of a speech signal until the start of the following speech
signal. The averaging time is determined by the length of this segment.
In narrowband, the noise level is measured in dBm0p (psophometric weighting, see Recommendation
ITU-T O.41 [11]). In wideband the noise level is determined in dBm0 (A).
5.9 Signal to noise ratio in receive direction
The noise to signal ratio in receive direction is defined as the difference between the active speech level and the level of
noise in receive direction (Speech signal level/noise level = SNR).
The signal level is the average level of the complete speech signal. The signal level is measured using a speech level
voltmeter according to Recommendation ITU-T P.56 [12]. This level is the speech signal level.
The noise level in receive direction is determined as described in clause 5.8.
The weighted noise signal level is referenced to the speech signal level.
5.10 Speech signal attenuation (or gain)
The speech signal attenuation is the difference between the active speech level at the receiving and at the sending point.
5.11 Talker echo delay
In telecommunications, the term talker echo describes delayed and undesired feedback from the send signal into the
receive path. The so-called echo source is the reflection point between send and receive directions. Talker echo delay is
the round-trip delay of the echo path. The impact of user perception of talker echo in conjunction with delay is
explained in Recommendation ITU-T G.131 [6]. The detailed test description is to be found in ETSI ES 202 765-2 [5].
In general the test of talker echo delay can be based on cross correlation between the speech signal inserted and the echo
signal received. The measurement is corrected by delays which are caused by the test equipment. The maximum of the
cross-correlation function is used for the determination. However, it shall be noted that such measurements can only be
made in case the echo signal is sufficiently high to allow a reliable calculation of the cross correlation.
NOTE: In case the talker echo is received at a very low level, the echo loss might be artificially decreased in
order to allow for the calculation of talker echo delay.
5.12 Double talk performance
This parameter looks into the situation when the talk spurts of both partners of a conversation overlap for a period of
time. Degradations due to bad double talk performance can be perceived as very annoying because this impairment has
a potential to frequently interrupt the flow of the conversation.
During double talk the speech is mainly determined by two parameters: Impairment caused by echo during double talk
and level variation between single and double talk (attenuation
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