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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. A benchmark shall be the evaluation of performance value/s of a parameter or set of parameters for the purpose ...
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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...
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5 Technical concept
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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 sig...
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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...
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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...
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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 ...
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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].
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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]. ETSI ETSI TS 103 222-1 V1.4.1 (2025-02) 18
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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 na...
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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 (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-...
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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.
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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...
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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...
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5.13 Interrupted voice transmission
A call is defined as interrupted if the duration of the interruption of the voice transmission is > 1 s and the call connection is maintained. ETSI ETSI TS 103 222-1 V1.4.1 (2025-02) 20
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5.14 Listening speech quality
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5.14.1 General aspects of Listening Speech Quality
The listening speech quality represents the intrinsic quality of speech signal as perceived by the user at the receiving end. This indicator takes into account the impairments introduced by the transmission system. The MOS-LQO score is obtained by comparing speech samples: • the original undistorted reference speech si...
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5.14.2 General aspects of voice channel test calls
For the all voice channel tests, an aligned structure of the voice call shall be used. In this call sentence pairs (male/female) fulfilling the requirements of Recommendation ITU-T P.863.1 [18] shall be transmitted from A to B and from B to A. Speech files especially tested for the use with Recommendation ITU-T P.863 [...
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5.14.3 Connections without parallel data transfer
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5.14.3.1 Connections with one voice channel
For the single voice channel Test, a test call consisting of the three following parts should be used: • Channel Convergence Quality test. • Listening Speech Quality test. • DTMF test. Figures 11 to 13 depict the detailed description of the single voice channel test. The general technical aspects are described in claus...
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5.14.3.2 Multiple voice channel access
In the case of multiple voice channel access or SIP trunking, during the complete testing phase the first call (single voice channel test) from user A to user B is active (see figure 14). The single voice channel test consists of the following parts: • Convergence Quality test. • Listening Speech Quality test. • DTMF t...
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5.14.4 Connections with parallel data transfer
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5.14.4.0 Introduction
The data traffic should be parameterized according to respective typical service usages. This refers to the overall load as well as to the composition with respect to the number of sockets and temporal shape. The use of multiple parallel socket connections may be required either to reach the desired load, or to meet th...
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5.14.4.1 Quality measurement of one voice channel and parallel data transfer
In the case when the access link is used for voice and data application the voice quality measurement sequence with parallel upload/download shall be used. Table 5 gives an overview about the connections options with parallel data transfer, figures 17 and 18 depict the measurement of voice quality with parallel data lo...
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5.14.4.3 Quality measurement of multiple voice channels and data transfer
In the case of multiple voice channel access or SIP Trunking, during the complete testing phase the first call from user A to user B is active. During the parallel download and upload the data size should be between 80 % and 100 % of the nominal data capacity of the link or respectively the maximal data capacity which ...
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5.15 Listening speech quality stability
The listening speech quality stability should be analysed all along the duration of the call. This indicator takes into account the degradations generated on the signal by the transmission links. Several measurements of MOS-LQO score performed with Recommendation ITU-T P.863 [2] are performed in series within the same ...
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5.16 End-to-end audio delay
This parameter represents the global delay from one user to the other one. This indicator takes into account the transmission delay of networks but also processing delay in sending and receiving terminals. The end-to-end delay can be measured acoustically from mouth to ear, from one access point to the other one, see f...
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5.17 End -to-end audio delay variation
The test signal consists of a series of CSS-signals using a nominal network level of -16 dBm0 with a total duration of 120 s. The pause of the CSS-sequence should be 150 ms. The delay of every CSS-signal should be measured. The delay variation for each CSS-signal D(i) compared to the first CSS signal (as described in R...
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5.18 Frequency response in receive direction
Narrowband telephony should transmit signals between 300 Hz and 3 400 Hz. Wideband telephony should transmit signals between 50 Hz and 7 000 Hz. The objective of this measurement is to see which bandwidth is used, and also to see whether a partial and unwanted bandwidth limitation is present. The frequency response is ...
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5.19.1 General considerations
This test applies to Fax bit rates ≤ 14,4 kbit/s and Fax bit rates ≥ 14,4 kbit/s in accordance with Recommendations ITU-T T.30 [i.7] and T.38 [i.8]. Figure 28 gives an overview about the Fax stack for FoIP. Figure 28: FAX Stack for FoIP Following is the list of electronic versions of the test pages related to the test ...
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5.19.2 Fax set-up duration
To determine the fax set-up duration, the time in seconds is measured from the sending of the dialling information by the "A" side to the start of transmission of the fax page on the "A" side (see green arrow in figure 29). Figure 29: Fax setup duration
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5.19.3 Fax transmission duration
This value measured shows the transmission time of a fax page in seconds. The fax transmission duration is defined in the context of the present document as the time that elapses from the start of transmission of the fax page by the "A" side until the complete transfer of the fax page to the "B" side (see green arrow i...
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5.19.4 Fax failure ratio
The fax failure ratio is defined as the ratio of failed fax transmissions and all fax transmissions initiated, see figure 31. A fax transmission is considered to have failed if the fax connection setup or the fax transmission is unsuccessful, or if fax connection setup and fax were not completed within 180 s. Figure 31...
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5.19.5 Test case descriptions
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5.19.5.1 Quality measurement of one fax channel
Table 8 gives an overview about the connections options without parallel data transfer. Figure 32 gives an overview a detailed overview of a single FAX channel test. ETSI ETSI TS 103 222-1 V1.4.1 (2025-02) 43 Table 8: Transmission options without additional data traffic No. From Payload type assigned To G.711 [23] Mode...
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5.19.5.2 Quality measurement of one fax channel and parallel data transfer
The data traffic should be parameterized according to respective typical service usages. This refers to the overall load as well as to the composition with respect to the number of sockets and temporal shape. The use of multiple parallel socket connections may be required either to reach the desired load, or to meet th...
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5.20 Early media listening speech quality
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5.20.1 Introduction
Early media refers to media (e.g. audio and video) which are exchanged before a particular session is accepted by the called user (in terms of the signalling). Within a dialogue, early media occurs from the moment the initial INVITE is sent until the User Agent Server (UAS) generates a final response. It may be unidire...
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5.20.2 Early media generated by the called party
To test the early media, the listening speech quality test from B to A after the '183 Session Progress' message is sent. In case of IMS implementations the '183 Session Progress' should contain the P-Early - media header. In case that the called user is an ISDN user an ALERTING with Progress Indicator #8 or a progress ...
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5.21 Jitter Buffer and IP prioritization response time
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5.21.1 Jitter Buffer and IP prioritization response time without data transfer
To test the jitter buffer and IP prioritization response time the test starts directly with a Listening Speech Quality test from the called party to the calling party directly after the connection is established. After the call is released the tests is repeated in the opposite direction. However, only one speech sample...
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5.21.2 Jitter Buffer and IP prioritization response time with data transfer
To test the jitter buffer and IP prioritization response time the test starts with a Listening Speech Quality test from the called party to the calling party. After the call is released the tests is repeated in the opposite direction. The flow of the voice call is the same as in clause 5.21.1, there is just a surroundi...
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5.23.1 Introduction
This requirement is dealing with transcoding between G.711 without telephony events and G.711 with telephony events (and back), while also processing a DTMF signal with the following characteristics: • DTMF Signal Duration: 70 ms. • Pause between DTMF Signals: 70 ms. (This pause is not directly relevant for the transco...
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5.23.4 DTMF Signal Duration
The DTMF signal itself lasts for 70 ms, and this time is considered part of the total delay due to the signal being processed during transcoding. DTMF signal delay: • 70 ms (DTMF signal duration) = 70 ms. Total Pure Transcoding Delay Calculation: • The Total Pure Transcoding Delay is the sum of: - The transcoding delay...
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6.1 Introduction to Load generation alternatives
There are multiple methods to generate load that competes with the applications of interest. Earlier clauses specify TCP connection details to introduce the load, such as the number of connections used. Use of TCP connections with active flow control has both advantages and disadvantages that are described below, and t...
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6.2 TCP load generation with flow-control enabled
Many testing methods (and corresponding tools) have been developed to initiate one or more TCP connections between test hosts, and to launch traffic intended to deliver test files or maintain connection activity for a specified time duration (while other tests run to completion). For example, the latest generation of t...
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6.3 TCP load generation with no flow-control
It is possible to disable the usual phases of TCP slow-start and congestion-avoidance, and to introduce TCP packet streams with typical "bursty" characteristics. At the same time, the packet streams can be designed to deliver a fixed capacity of bits per second as the test load. This is a clear advantage of this method...
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6.4 Monitoring the generated load
The various methods have built-in measurements of the average throughput achieved, averaged over several connections where applicable. This is useful to monitor for the flow-control enabled methods, particularly because the generated load may be unexpectedly high or low, and influence the results for the applications o...
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6.5 QoS IP transmission layer parameter tests
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6.5.0 Introduction
During the transmission the IP traffic from the user is passing several Firewalls and Session Border Controllers which are in some cases not transparent. To ensure a transparent E2E transmission the following tests should be executed. Table 11: Transparent E2E transmission tests 1 VoIP. 2 TCP Port: TCP is an important ...
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6.5.1 VoIP (Voice over IP)
To ensure a transparent E2E transmission of VoIP, the following Quality of Service (QoS) parameters defined in Recommendation ITU-T Y.1541 [22] should be tested with a simulated VoIP call. Table 12: Network performance objectives test for VoIP Simulated VoIP call 1 Sample rate 8 000, bits per sample: 8 2 Call duration ...
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6.5.2 TCP Parameters
TCP is an important connection oriented Internet protocol. It is used for example for web pages or e-mail. To ensure a transparent E2E transmission the following TCP port tests should be executed. Table 13: TCP port tests 1 Web site protocol (HTTP alternate, TCP port 8080 incoming) 2 Secure control of communication ses...
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6.5.3 UDP Parameters
UDP is an important connectionless Internet protocol. It is used for real-time communications, e.g. for VoIP and video. To ensure a transparent E2E transmission the following UDP port tests should be executed. Table 14: UDP port tests 1 Microsoft Terminal Server (RDP, UDP port 3389 incoming) 2 Streaming of audio and vi...
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6.5.4 Web Page
The website test downloads a reference web page (mobile Kepler page by ETSI). It is verified, if the page can be transferred and how long the download of the page takes.
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6.5.5 Unmodified content
This test downloads a test web resource (e.g. image) and checks if it was modified during transport. EXAMPLE: Target: 'http://webtest.nettest.at/qostest/reference01.jpg' Range: bytes=1000000-1004999 Duration: 0,1 s ETSI ETSI TS 103 222-1 V1.4.1 (2025-02) 60 Length: 5 000 Status code: 206 Hash: fc563e1e80b8cb964d712982f...
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6.5.6 Transparent connection
This test checks if a request is modified by a proxy or other middlebox Table 15: Transparent port test 1 Port: 45963, Request: GET/HTTR/7.9 2 Port: 80, Request: GET 3 Port: 25, Request: SMTP Transparent 4 Port: 23329, Request: GET 5 Port: 80, Request: GET/HTTR/7.9
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6.5.7 DNS
DNS is a fundamental Internet service. It is used to translate domain names to IP addresses. Depending on the test it is checked if the service is available, if the answers are correct and how fast the server responds. ETSI ETSI TS 103 222-1 V1.4.1 (2025-02) 61 Table 16: DNS test 1 google.com 2 youtube.com 3 facebook.c...
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6.5.8 Traceroute
Traceroute is a tool for displaying the route across IP based networks. ETSI ETSI TS 103 222-1 V1.4.1 (2025-02) 62 Annex A (informative): Bibliography • Recommendation ITU-T P.800.1 (07-2006): "Mean Opinion Score (MOS) terminology". ETSI ETSI TS 103 222-1 V1.4.1 (2025-02) 63 History Document history V1.1.1 November 201...
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1 Scope
The present document defines an electronic interface for the transmission of intercepted information as part of Lawful Interception. This interface is used from points of interception to LI mediation functions. Typical reference models for LI define an interface between Law Enforcement Agencies (LEAs) and Communication...
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2 References
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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 a...
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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 i...
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3 Definition of terms, symbols and abbreviations
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3.1 Terms
For the purposes of the present document, the terms given in ETSI TS 103 221-1 [1] apply.
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3.2 Symbols
Void. ETSI ETSI TS 103 221-2 V1.10.1 (2026-03) 9
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3.3 Abbreviations
For the purposes of the present document, the abbreviations given in ETSI TS 103 221-1 [1] and the following apply: 3GPP 3rd Generation Partnership Project AXRI Additional XID Related Information CSP Communications Service Provider DHCP Dynamic Host Configuration Protocol DID Domain IDentifier GTP GPRS Tunnelling Proto...
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4 Introduction and reference model
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4.1 Reference model
The X2/X3 interface is based on communication between: a) The Point Of Interception (POI), which performs interception. b) The Mediation and Delivery Function (MDF), which performs the necessary translation, correlation and mediation for onward handover over material to LEAs via the HI2 and HI3 interfaces. The X2/X3 re...
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4.2 Assumptions
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4.2.1 Architecture
The present document makes minimal assumptions about the LI architecture in which the X2/X3 interfaces are deployed. The X2/X3 interface is intended to be sufficiently flexible to be used as part of LI architectures defined elsewhere and assumes that the POI and MDF are deployed following an LI architecture defined sep...
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4.2.2 Implementation/realization
The present document assumes that implementations of an LI architecture which utilize X1, X2 and X3 can be described by the high-level model as shown in figure 4.2.2-1. Implementation NF POI POI POI POI X3 X2 X3 X2 NF X1 Control Function Control Function Figure 4.2.2-1: Assumed Implementation Model The model consists o...
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4.2.3 Deployment infrastructure
The present document assumes that the transport infrastructure between POI/NF and MDF is untrusted (see clause A.2.8) but assumes that the platform on which the POI, NF and MDF are realized is appropriately secured. It does not make any specific assumptions about whether either the platform or transport infrastructure ...
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4.2.4 Regulatory assumptions
The present document assumes that material delivered over X2/X3 may be used as evidence in court. As such, it assumes that the X2/X3 interface is capable of indicating when data has been lost over the X2/X3 interface (see clause A.1.8), but recovery of this data (e.g. by buffering and retransmission) are out of scope (...
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4.3 Relationship to other standards
The present document forms part of an overall set of standards together with ETSI TS 103 221-1 [1] (X1) and ETSI TS 104 000 [26] (X0). Some models of LI (e.g. 3GPP TS 33.108 [15], 3GPP TS 33.128 [20]) define interfaces for the purposes described in clause 4.1 (e.g. X2, X3 defined by 3GPP TS 33.108 [15] or LI_X2, LI_X3 ...
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5 Message contents and parameters
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5.1 Overview
The POI sends data to the MDF as a binary stream of X2/X3 Protocol Data Units (PDUs). Each PDU is formatted as described in the following clauses. Each X2/X3 PDU consists of three main sections: • A set of mandatory PDU header fields containing identifiers, routing and correlation information - see clause 5.2. • A set ...
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5.2 PDU Header Fields
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5.2.1 Version
The POI shall populate the Version field with the version of the specification used to create the PDU, given as a 16-bit unsigned integer. The Version field consists of a major and minor version number. ETSI ETSI TS 103 221-2 V1.10.1 (2026-03) 13 For PDUs created according to the present document, the Version field is ...
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5.2.2 PDU Type
The POI shall populate the PDU Type field to indicate the type of PDU. The PDU Type field shall take one of the values described in table 5.2.2-1. Table 5.2.2-1: PDU Types Value Meaning 1 X2 PDU 2 X3 PDU 3 Keepalive 4 Keepalive Acknowledgement
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5.2.3 Header Length
The POI shall populate the Header Length field with the length of the header in octets, including the mandatory and any conditional fields that have been populated. For PDUs created against the present document, this value has a minimum value of 40 octets.
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5.2.4 Payload Length
The POI shall populate the Payload Length field with the length of the Payload field in octets. Keepalive and Keepalive Acknowledgement PDU shall set the field to zero.
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5.2.5 Payload Format
The POI shall indicate the format and encoding of the Payload field by setting the Payload Format field to the appropriate value. A list of valid values, and their definitions, is given in clause 5.4.
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5.2.6 Payload Direction
The POI shall populate the Payload Direction field with an indication of the direction of the intercepted data or event contained in the PDU. The Payload Direction field shall take one of the values described in table 5.2.6-1. ETSI ETSI TS 103 221-2 V1.10.1 (2026-03) 14 Table 5.2.6-1: Payload Direction Direction Value ...
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5.2.7 XID
The POI shall populate the XID field with the XID associated with the intercepted product, as assigned by the relevant X1 interface (see ETSI TS 103 221-1 [1]). When the X1 task has been provisioned with an optional ProductID (see ETSI TS 103 221-1 [1] clause 6.2.1.2, table 6.2.1.2-1), the XID shall be populated with t...
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5.2.8 Correlation ID
Where the POI correlates X2/X3 PDUs, the POI shall ensure that PDUs associated with the same communication session are given the same Correlation ID value. The value shall uniquely identify the communication within a given context. The scheme for generating Correlation IDs is defined by the relevant LI architecture. Th...
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5.3 Conditional Attribute Fields
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5.3.1 General structure
The POI may provide a number of conditional attributes at the end of the header, as directed by the relevant LI architecture. Each of these conditional attributes has the Type-Length-Value (TLV) structure described in table 5.3.1-1. Table 5.3.1-1: General Conditional Attribute Structure Field Name Description Format At...
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5.3.2 ETSI TS 102 232-1 Defined Attribute
If used, this conditional attribute shall contain an attribute defined by ETSI TS 102 232-1 [2]. Multiple occurrences of this conditional attribute are permitted.
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5.3.3 3GPP TS 33.128 Defined Attribute
If used, this conditional attribute shall contain an attribute defined by 3GPP TS 33.128 [20]. Multiple occurrences of this conditional attribute are permitted.
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5.3.4 3GPP TS 33.108 Defined Attribute
If used, this conditional attribute shall contain an attribute defined by 3GPP TS 33.108 [15]. Multiple occurrences of this conditional attribute are permitted.
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5.3.5 Proprietary Attribute
If used, this conditional attribute shall contain an attribute encoded as described in table 5.3.5-1. Table 5.3.5-1: Proprietary Conditional Attribute Structure Field Name Description Format Attribute Owner Length Indicates the length of the Attribute Owner field. 8-bit unsigned integer Attribute Owner FQDN indicating ...
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5.3.6 Domain ID (DID)
If used, this conditional attribute shall contain a value that identifies the CSP, network or resource group in which the NF or POI exists. The format and content of this conditional attribute is left to the relevant LI architecture to define.
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5.3.7 Network Function ID (NFID)
If used, this conditional attribute shall contain a value that identifies the NF associated with the POI to the MDF. The format and content of this conditional attribute is left to the relevant LI architecture to define. Depending on the architecture and deployment scenarios, it may be used as a way to identify the typ...
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5.3.8 Interception Point ID (IPID)
If used, this conditional attribute shall contain a value that identifies the POI within the NF. The format and content of this conditional attribute is left to the relevant LI architecture and implementation to define.
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5.3.9 Sequence Number
If used, this conditional attribute shall contain the sequence number of the X2/X3 PDU. If used, the Sequence Number shall start at zero and increment by one for each X2/X3 PDU with the same XID, DID, NFID, IPID and Correlation ID context. A separate sequence shall be maintained for X2 and X3 PDUs within the same conte...
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5.3.10 Timestamp
If used, this conditional attribute shall contain the time that the content for the PDU was intercepted. The time shall be given in POSIX.1-2017 [3] timespec format (i.e. the number of elapsed seconds since the start of the Unix epoch in UTC). The value shall be given as two successive 32-bit unsigned integers, with th...
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5.3.11 Source IPv4 Address
If used, this conditional attribute shall contain the source IPv4 address related to the Payload field given as four octets in network byte order (i.e. most significant octet first). This conditional attribute is primarily intended for use when using payload formats that strip off the IP-layer encapsulation (e.g. SIP M...
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5.3.12 Destination IPv4 Address
If used, this conditional attribute shall contain the destination IPv4 address related to the Payload field given as four octets in network byte order (i.e. most significant octet first). This conditional attribute is primarily intended for use when using payload formats that strip off the IP-layer encapsulation (e.g. ...
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5.3.13 Source IPv6 Address
If used, this conditional attribute shall contain the source IPv6 address related to the Payload field given as sixteen octets in network byte order (i.e. most significant octet first). This conditional attribute is primarily intended for use when using payload formats that strip off the IP-layer encapsulation (e.g. SI...
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5.3.14 Destination IPv6 Address
If used, this conditional attribute shall contain the destination IPv6 address related to the Payload field given as sixteen octets in network byte order (i.e. most significant octet first). This conditional attribute is primarily intended for use when using payload formats that strip off the IP-layer encapsulation (e....