ETSI TS 103 389 V3.2.1 (2019-12)
Railway Telecommunications (RT); Global System for Mobile communications (GSM); Usage of Session Initiation Protocol (SIP) on the Network Switching Subsystem (NSS) to Fixed Terminal Subsystem (FTS) interface for GSM Operation on Railways
Railway Telecommunications (RT); Global System for Mobile communications (GSM); Usage of Session Initiation Protocol (SIP) on the Network Switching Subsystem (NSS) to Fixed Terminal Subsystem (FTS) interface for GSM Operation on Railways
RTS/RT-0053
General Information
Standards Content (Sample)
TECHNICAL SPECIFICATION
Railway Telecommunications (RT);
Global System for Mobile communications (GSM);
Usage of Session Initiation Protocol (SIP)
on the Network Switching Subsystem (NSS)
to Fixed Terminal Subsystem (FTS)
interface for GSM Operation on Railways
2 ETSI TS 103 389 V3.2.1 (2019-12)
Reference
RTS/RT-0053
Keywords
GSM-R, railways
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3 ETSI TS 103 389 V3.2.1 (2019-12)
Contents
Intellectual Property Rights . 6
Foreword . 6
Modal verbs terminology . 6
Introduction . 6
1 Scope . 8
2 References . 8
2.1 Normative references . 8
2.2 Informative references . 10
3 Definition of terms, symbols and abbreviations . 11
3.1 Terms . 11
3.2 Symbols . 13
3.3 Abbreviations . 13
4 Reference System Architecture . 15
5 Interface Functionality . 16
5.0 General Description . 16
5.1 Basic Call . 16
5.1.0 Primary Function . 16
5.1.1 Progress Indication . 16
5.1.2 Early Media . 16
5.2 Connected Parties Identity Information . 16
5.3 Call Hold . 16
5.4 Multi Level Precedence and Pre-emption . 17
5.5 Voice Group Call and Broadcast Call Control . 17
5.6 User-User-Information-Element Transport . 17
5.7 Reason Transport . 17
5.8 Call Transfer Notification . 17
5.9 Conferencing . 18
5.10 Call Forwarding . 18
5.11 Call Waiting . 18
6 Signalling Interface . 18
6.1 Network Layer Protocol . 18
6.2 Transport Layer Protocol . 18
6.3 Signalling Protocol . 19
6.3.0 General Provisions . 19
6.3.1 SIP Entities . 19
6.3.1.0 SIP networks . 19
6.3.1.1 SIP User Agent . 19
6.3.1.2 SIP Proxy . 19
6.3.2 SIP Request Methods . 20
6.3.3 SIP Responses . 20
6.3.4 SIP Header Fields . 20
6.3.5 SIP Bodies . 23
6.3.6 SIP URI Convention . 23
6.3.6.0 General provisions . 23
6.3.6.1 Display Name . 24
6.3.6.2 User Part . 24
6.3.6.3 Host Part . 24
6.3.6.4 URI Parameters . 25
6.3.6.5 Use . 25
6.3.6.6 Examples . 26
6.3.7 Option Tags . 27
6.3.8 Feature Parameter . 27
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4 ETSI TS 103 389 V3.2.1 (2019-12)
6.4 Interface Functionality to Signalling Interface Mapping . 27
6.4.0 General . 27
6.4.1 Basic Call . 27
6.4.2 Connected Parties Identity Information . 29
6.4.3 Media Session Renegotiation and Call Hold . 31
6.4.4 Early Media . 33
6.4.5 Multi Level Precedence and Pre-emption . 35
6.4.5.0 General Provisions . 35
6.4.5.1 Resource Priority . 36
6.4.5.2 Reason Indication for Precedence and Pre-emption Events . 36
6.4.5.3 Signalling Procedure for Precedence Call Blocking . 36
6.4.5.4 Signalling Procedure for Pre-emption . 37
6.4.6 Group Call and Broadcast Call Control . 38
6.4.7 User-to-User-Information-Element Transport . 38
6.4.8 Release Cause Transport . 38
6.4.9 SIP Session Timer . 39
6.4.10 OPTIONS Processing . 40
6.4.10.0 General Provisions . 40
6.4.10.1 OPTIONS Heartbeating . 41
6.4.11 Signalling for Group Call and Broadcast Call Control . 41
6.4.12 Call Transfer Notification . 43
6.4.13 Conferencing. 46
6.4.14 Call Forwarding . 49
6.4.15 Call Waiting . 51
7 Media Interface . 52
7.1 Network Layer Protocol . 52
7.2 Transport Layer Protocol . 52
7.3 Real-Time Transport Protocol . 52
7.3.0 General Provisions . 52
7.3.1 Media inactivity detection . 53
7.4 Media Codecs . 53
7.4.0 General Provisions . 53
7.4.1 DTMF . 53
8 Recorder Interface . 54
8.1 Reference Architecture . 54
8.2 Interface Functionality . 55
8.2.0 General . 55
8.2.1 Recording Session . 55
8.3 Signalling Interface . 55
8.3.0 General Description . 55
8.3.1 SIP Request Methods . 55
8.3.2 SIP Responses . 56
8.3.3 SIP Header Fields . 56
8.3.4 SIP Bodies . 58
8.3.5 SIP URI Convention . 59
8.3.6 Option Tags . 59
8.3.7 Feature Parameter . 59
8.3.8 Handling of Recording Session . 60
8.4 Media interface . 62
8.4.0 General . 62
8.4.1 Media mixing . 62
8.4.2 Multiple streams . 62
Annex A (normative): Locating SIP Entities . 63
Annex B (informative): Quality of Service framework . 66
Annex C (informative): Security Framework . 67
Annex D (informative): Mapping of EIRENE to Interface Features . 68
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5 ETSI TS 103 389 V3.2.1 (2019-12)
Annex E (informative): Group Call Control Scenarios . 70
Annex F (informative): Bibliography . 72
History . 73
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6 ETSI TS 103 389 V3.2.1 (2019-12)
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 Railway Telecommunications
(RT).
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
While a number of interoperability specifications for various interfaces at various layers of GSM-R systems exist, the
interface between the Network Switching Subsystem (NSS) and the Fixed Terminal Subsystem (FTS) has not yet been
addressed by any interoperability specification activity.
In most of the GSM-R system deployments available at the time of the creation of the present document, the Network
Switching Subsystem and the Fixed Terminal Subsystem are interconnected using TDM based interfaces such as
DSS1 [i.2].
ETSI TS 102 610 [9] specifies the usage and format of UUIE for call-related end-to-end functionality in GSM-R
systems but no other interworking topics.
The present document addresses the interoperability specification gap between the Network Switching Subsystem and
the Fixed Terminal Subsystem with an interface based on the Internet Protocol (IP) [2], the Session Initiation Protocol
(SIP) [3], the Session Description Protocol (SDP) [6] and the Real-Time Transport Protocol (RTP) [7].
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7 ETSI TS 103 389 V3.2.1 (2019-12)
In addition to the table of contents, the following explanation helps to navigate through and to understand the contents
of the present document:
• Clauses 1 to 3 are predefined by ETSI.
• Clause 4 shows and explains the reference system architecture and identifies the interface(s) for the present
document.
• Clause 5 holds the functional requirements for the interface subject to the present document.
• Clause 6 specifies in detail the signalling interface for all supported functions and services.
• Clause 7 specifies in detail the media interface.
• Clause 8 specifies the additions and changes for a voice recorder interface.
• Annex A explains the mechanism to locate SIP entities at the present interface.
• Annex B contains recommendations on the use and implementation of standardized Quality of Service
mechanisms at the present interface.
• Annex C contains recommendations about the security mechanisms.
• Annex D contains a mapping table of EIRENE [1] to interface features.
• Annex E contains a description of group call control scenarios.
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8 ETSI TS 103 389 V3.2.1 (2019-12)
1 Scope
The present document defines the signalling and media interface between the Network Switching Subsystem and the
Fixed Terminal Subsystem in order to provide a clear set of services needed for GSM-R operations. This includes voice
call service and available call-related supplementary services. In addition, requirements for specific implementation of
the signalling and media interface within either the Network Switching Subsystem or the Fixed Terminal Subsystem are
stated where applicable. The present document addresses the Internet Layer and upwards of the Internet Protocol Suite
IETF RFC 1122 [i.18] on the signalling and media interface.
Any service other than voice call service and call-related supplementary services (such as data services, Short Message
Service, etc.) is out of scope of the present document; additional features may be addressed in future releases.
The present document does not specify any other interface between the Network Switching Subsystem and the Fixed
Terminal Subsystem nor does it cover any internal interfaces of either NSS or FTS. Such interfaces may be addressed in
a future release of the present document.
The present document does not address any specific 3GPP Release or Architecture.
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] UIC P001D010 (Version 15.1): "UIC Project EIRENE System Requirements Specification".
NOTE: Available at http://www.uic.org/IMG/pdf/eirene_srs_15.1.pdf.
[2] IETF RFC 791 (1981): "Internet Protocol".
[3] IETF RFC 3261 (2002): "SIP: Session Initiation Protocol".
[4] IETF RFC 3264 (2002): "An Offer/Answer Model with the Session Description Protocol (SDP)".
[5] IETF RFC 4733 (2006): "RTP Payload for DTMF Digits, Telephony Tones, and Telephony
Signals".
[6] IETF RFC 4566 (2006): "SDP: Session Description Protocol".
[7] IETF RFC 3550 (2003): "RTP: A Transport Protocol for Real-Time Applications".
[8] IETF RFC 3326 (2002): "The Reason Header Field for the Session Initiation Protocol (SIP)".
[9] ETSI TS 102 610 (V1.1.0): "Railways Telecommunications (RT); Global System for Mobile
communications (GSM); Usage of the User to User Information Element for GSM Operation on
Railways".
[10] IETF RFC 5234 (2008): "Augmented BNF for Syntax Specifications: ABNF".
[11] IETF RFC 3262 (2002): "Reliability of Provisional Responses in Session Initiation Protocol
(SIP)".
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9 ETSI TS 103 389 V3.2.1 (2019-12)
[12] IETF RFC 4412 (2006): "Communications Resource Priority for the Session Initiation Protocol
(SIP)".
[13] IETF RFC 3325 (2002): "Private Extensions to the Session Initiation Protocol (SIP) for Asserted
Identity within Trusted Networks".
[14] IETF RFC 5876 (2010): "Updates to Asserted Identity in the Session Initiation Protocol (SIP)".
[15] IETF RFC 3323 (2002): "A Privacy Mechanism for the Session Initiation Protocol (SIP)".
[16] IETF RFC 4028 (2005): "Session Timers in the Session Initiation Protocol (SIP)".
[17] IETF RFC 3311 (2002): "The Session Initiation Protocol (SIP) UPDATE Method".
[18] Void.
[19] Void.
[20] Void.
[21] Void.
[22] Recommendation ITU-T Q.850 (2018): "Usage of cause and location in the Digital Subscriber
Signalling System No. 1 and the Signalling System No. 7 ISDN user part".
[23] Recommendation ITU-T E.164 (2010): "The international public telecommunication numbering
plan".
[24] Recommendation ITU-T Q.955.3 (1993): "Stage 3 description for community of interest
supplementary services using DSS 1: Clause 3-Multi-level precedence and pre-emption (MLPP)".
[25] IETF RFC 3986 (2005): "Uniform Resource Identifier (URI): Generic Syntax".
[26] IETF RFC 768 (1980): "User Datagram Protocol".
[27] Recommendation ITU-T G.711 (1988): "Pulse code modulation (PCM) of voice frequencies".
[28] IETF RFC 2833 (2000): "RTP Payload for DTMF Digits, Telephony Tones and Telephony
Signals".
[29] Void.
[30] IETF RFC 3840 (2004): "Indicating User Agent Capabilities in the Session Initiation Protocol
(SIP)".
[31] IETF RFC 4574 (2006): "The Session Description Protocol (SDP) Label Attribute".
[32] Recommendation ITU-T I.255.3 (1990): "Multi-Level Precedence and Pre-emption service
(MLPP)".
[33] IETF RFC 4579 (2006): "Session Initiation Protocol (SIP) Call Control - Conferencing for User
Agents".
[34] IETF RFC 3891 (2004): "The Session Initiation Protocol (SIP) "Replaces" Header".
[35] IETF RFC 7462 (2015): "URNs for the Alert-Info Header Field of the Session Initiation Protocol
(SIP)".
[36] IETF RFC 4244 (2005): "An Extension to the Session Initiation Protocol (SIP) for Request History
Information".
[37] IETF RFC 4458 (2006): "Session Initiation Protocol (SIP) URIs for Applications such as
Voicemail and Interactive Voice Response (IVR)".
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10 ETSI TS 103 389 V3.2.1 (2019-12)
[38] ETSI TS 129 163 (V12.13.0): "Digital cellular telecommunications system (Phase 2+) (GSM);
Universal Mobile Telecommunications System (UMTS); LTE; Interworking between the IP
Multimedia (IM) Core Network (CN) subsystem and Circuit Switched (CS) networks (3GPP
TS 29.163 version 12.13.0 Release 12)".
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] IETF RFC 7433 (2015): "A Mechanism for Transporting User to User Call Control Information in
SIP".
[i.2] ETSI ETS 300 403-1 (V1.3.2): "Integrated Services Digital Network (ISDN); Digital Subscriber
Signalling System No. one (DSS1) protocol; Signalling network layer for circuit-mode basic call
control; Part 1: Protocol specification [ITU-T Recommendation Q.931 (1993), modified]".
[i.3] IETF RFC 6086 (2011): "Session Initiation Protocol (SIP) INFO Method and Package
Framework".
[i.4] IETF RFC 3428 (2002): "Session Initiation Protocol (SIP) Extension for Instant Messaging".
[i.5] IETF RFC 3515 (2001): "The Session Initiation Protocol (SIP) Refer Method".
[i.6] IETF RFC 3265 (2002): "Session Initiation Protocol (SIP)-Specific Event Notification".
[i.7] IETF RFC 3903 (2004): "Session Initiation Protocol (SIP) Extension for Event State Publication".
[i.8] Void.
[i.9] IETF RFC 3665 (2003): "Session Initiation Protocol (SIP) Basic Call Flow Examples".
[i.10] IETF RFC 3960 (2004): "Early Media and Ringing Tone Generation in the Session Initiation
Protocol (SIP)".
[i.11] ETSI EN 300 925 (V7.0.2): "Digital cellular telecommunications system (Phase 2+) (GSM);
Voice Group Call Service (VGCS) - Stage 1 (GSM 02.68 version 7.0.2 Release 1998)".
[i.12] ETSI EN 300 926 (V8.0.1): "Digital cellular telecommunications system (Phase 2+) (GSM);
Voice Broadcast Service (VBS) - Stage 1 (GSM 02.69 version 8.0.1 Release 1999)".
[i.13] IETF RFC 3263 (2002): "Session Initiation Protocol (SIP): Locating SIP Servers".
[i.14] IETF RFC 1035 (1987): "Domain names - implementation and specification".
[i.15] IETF RFC 2181 (1997): "Clarifications to the DNS Specification".
[i.16] IETF RFC 2663 (1999): "IP Network Address Translator (NAT) Terminology and
Considerations".
[i.17] Void.
[i.18] IETF RFC 1122 (1989): "Requirements for Internet Hosts -- Communication Layers".
[i.19] IETF RFC 3551 (2003): "RTP Profile for Audio and Video Conferences with Minimal Control".
[i.20] IETF RFC 7866 (2016): "Session Recording Protocol".
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[i.21] IETF RFC 5009 (2007): "Private Header (P-Header) Extension to the Session Initiation Protocol
(SIP) for Authorization of Early Media".
[i.22] IETF RFC 2474 (1998): "Definitions of the Differentiated Services Field (DS Field) in the IPv4
and IPv6 Headers".
[i.23] IETF RFC 2475 (1998): "An Architecture for Differentiated Services".
[i.24] IETF RFC 4594 (2006): "Configuration Guidelines for DiffServ Service Classes".
[i.25] IETF RFC 5865 (2010): "A Differentiated Services Code Point (DSCP) for Capacity-Admitted
Traffic".
3 Definition of terms, symbols and abbreviations
3.1 Terms
For the purposes of the present document, the terms given in IETF RFC 3261 [3] and the following apply:
call: SIP Dialog between two Signalling Endpoints
NOTE: Established for the purpose of a voice communication and related data exchange.
client: any network element that sends SIP requests and receives SIP responses
NOTE: Clients may or may not interact directly with a human user. User agent clients and proxies are clients.
Communication Session (CS): session that is the subject of recording
dialog: peer-to-peer SIP relationship between two UAs that persists for some time
NOTE: A dialog is established by SIP messages, such as a 2xx response to an INVITE request. A dialog is
identified by a call identifier, local tag, and a remote tag.
final response: response that terminates a SIP transaction, as opposed to a provisional response that does not
NOTE: All 2xx, 3xx, 4xx, 5xx and 6xx responses are final.
Fixed Terminal Subsystem (FTS): part of the EIRENE [1] system that provides access to this network (and services)
via controller equipment (in general referred to as Fixed Terminals)
Fully Qualified Domain Name (FQDN): domain name that includes all higher level domains relevant to the entity
named
header: component of a SIP message that conveys information about the message
header field: component of the SIP message header
NOTE: A header field can appear as one or more header field rows. Header field rows consist of a header field
name and zero or more header field values. Multiple header field values on a given header field row are
separated by commas. Some header fields can only have a single header field value, and as a result,
always appear as a single header field row.
initiator, calling party, caller: party initiating a session (and dialog) with an INVITE request
NOTE: A caller retains this role from the time it sends the initial INVITE that established a dialog until the
termination of that dialog.
invitee, invited user, called party, callee: party that receives an INVITE request for the purpose of establishing a new
session
NOTE: A callee retains this role from the time it receives the INVITE until the termination of the dialog
established by that INVITE.
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12 ETSI TS 103 389 V3.2.1 (2019-12)
Media Endpoint, RTP Endpoint: entity that terminates RTP stream(s) under the control of a single SIP Endpoint in
the same subsystem
NOTE: This entity may be physically separated from the SIP Endpoint.
method: primary function that a request is meant to invoke on a server
NOTE: The method is carried in the request message itself. Example methods are INVITE and BYE.
Network Switching Subsystem (NSS): part of the PLMN infrastructure that performs all necessary functions in order
to handle the call services to and from the mobile stations as well as to and from fixed terminals
operational priority: different call types have call priorities during railway communications
NOTE 1: This is the definition given in EIRENE SRS [1].
NOTE 2: This behaviour is mentioned as operational priority of a call.
option tag: unique identifiers used to designate new options (extensions) in SIP
NOTE: The composition of option tags is defined in IETF RFC 3261 [3].
provisional response: response used by the server to indicate progress, but that does not terminate a SIP transaction
NOTE: 1xx responses are provisional, other responses are considered final.
proxy, proxy server: intermediary entity that acts as both a server and a client for the purpose of making requests on
behalf of other clients
NOTE: A proxy server primarily plays the role of routing, which means its job is to ensure that a request is sent to
another entity "closer" to the targeted user. Proxies are also useful for enforcing policy (for example,
making sure a user is allowed to make a call). A proxy interprets, and, if necessary, rewrites specific parts
of a request message before forwarding it.
Recording Session (RS): SIP session created between SRC and SRS for the purpose of recording a Communication
Session
request: message sent from a client to a server, for the purpose of invoking a particular operation
response: message sent from a server to a client, for indicating the status of a request sent from the client to the server
server: element that receives requests in order to service them and sends back responses to those request
NOTE: Examples of servers are proxies, user agent servers, redirect servers, and registrars.
session: set of multimedia senders and receivers and the data streams flowing from senders to receivers
NOTE: A callee can be invited several times, by different calls, to the same session. If SDP is used, a session is
defined by the concatenation of the SDP user name, session id, network type, address type, and address
elements in the origin field.
Signalling Endpoint, SIP Endpoint: entity that acts as a SIP User Agent
NOTE: Within the scope of the present document this term refers to NSS and FTS.
Signalling Proxy, SIP Proxy: logical entity to route requests to the user's current location, authenticate and authorize
users for services, implement provider call-routing policies, and provide features to users
(SIP) transaction: single request and any responses to that request, which include zero or more provisional responses
and one or more final responses
NOTE: In the case of a transaction where the request was an INVITE (known as an INVITE transaction), the
transaction also includes the ACK only if the final response was not a 2xx response. If the response was a
2xx, the ACK is not considered part of the transaction.
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13 ETSI TS 103 389 V3.2.1 (2019-12)
tag: parameter which is used in the To and From header fields of SIP messages to identify a dialog
NOTE: The composition of tags is defined in IETF RFC 3261 [3].
User Agent (UA): Internet endpoint
User Agent Client (UAC): logical entity that creates a new request, and then uses the client transaction state machinery
to send it
NOTE: The role of UAC lasts only for the duration of that transaction. In other words, if a piece of software
initiates a request, it acts as a UAC for the duration of that transaction. If it receives a request later, it
assumes the role of a user agent server for the processing of that transaction.
User Agent Server (UAS): logical entity that generates a response to a SIP request
NOTE: The response accepts, rejects, or redirects the request. This role lasts only for the duration of that
transaction. In other words, if a piece of software responds to a request, it acts as a UAS for the duration
of that transaction. If it generates a request later, it assumes the role of a user agent client for the
processing of that transaction.
3.2 Symbols
Void.
3.3 Abbreviations
For the purposes of the present document, the following abbreviations apply:
ACK ACKnowledgement
AF Assured Forwarding
AoCC Advice of Charge (Charging)
AoCI Advice of Charge (Information)
B2BUA Back to Back User Agent
BAIC Barring of All Incoming Calls
BAOC Barring of All Outgoing Calls
BIC-Roam Barring of Incoming Calls when Roaming Outside the Home PLMN Country
BNF Backus Naur Form
BOIC Barring of Outgoing International Calls
BOIC-exHC BOIC except those to Home PLMN Country
CCBS Completion of Calls to Busy Subscribers
CFB Call Forwarding on Mobile Subscriber Busy
CFNRc Call Forwarding on Mobile Subscriber Not Reachable
CFNRy Call Forwarding on No Reply
CFU Call Forwarding Unconditional
CLIP Calling Line Identification Presentation
CLIR Calling Line Identification Restriction
CN Core Node
CoLP Connected Line Identification Presentation
CoLR Connected Line Identification Restriction
CS Communication Session
CUG Closed User Group
CW Call Waiting
DL Down Link
DNS Domain Name Service
DSCP Differentiated Service Code Point
DTMF Dual Tone Multi Frequency
ECT Explicit Call Transfer
EF Expedited Forwarding
EIRENE European Integrated Railway Radio Enhanced Network
eMLPP enhanced Multi-Level Precedence and Pre-emption
FQDN Fully Qualified Domain Name
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14 ETSI TS 103 389 V3.2.1 (2019-12)
FTP File Transfer Protocol
FTS Fixed Terminal Subsystem
GSM-R Global System Mobile-Railways
HOLD Call hold
HTTP Hyper Text Transfer Protocol
IETF Internet Engineering Task Force
IN Intelligent Network
INF INForm
INV INVite
IP Internet Protocol
ITU-T International Telecommunication Union - Telecommunication standardization sector
MLPP Multi-Level Precedence and Pre-emption
MO/PP Mobile Originated/Point-to-Point
MPTY Multi Party Service
MT/PP Mobile Terminated/Point-to-Point
NAPT Network Address Port Translation
NAT Network Address Translation
NSS Network Switching Subsystem
OK OKay
OPT OPTion
OSI Open Systems Interconnection
PABX Private Access Branch eXchange
PCM Pulse Code Modulation
PCMA Pulse Code Modulation - A law
PCM-A Pulse Code Modulation - A law
PCMU Pulse Code Modulation - u-law
PHB Per Hop Behaviour
PLMN Public Land Mobile Network
PRA PRovisional Acknowledgment
PRACK Provisional Response Acknowledgement
PSTN Public Switched Telephone Network
QoS Quality of Service
RFC Request For Comments
RS Recording Session
RTP Real-Time Transport Protocol
SDP Session Description Protocol
SE Session Expires
SIP Session Initiation Protocol
SRC Session Recording Client
SRS Session Recording Server
SRTP Secured Real-time Protocol
SSRC Synchronization SouRCe
TDM Time Division Multiplexing
ToS Type of Service
UA User Agent
UAC User Agent Client
UAS User Agent Server
UDP User Datagram Protocol
UIC Union Internationale des Chemins de Fer, International Union of Railways
UPD UPDate
URI Uniform Resource Identifier
URL Uniform Resource Locator
URN Uniform Resource Name
USSD Unstructured Supplementary Service Data
UUI User-to-User Information
UUIE User to User Information Element
UUS1 User-to-User Signalling 1
VBS Voice Broadcast Service
VGCS Voice Group Call Service
ETSI
15 ETSI TS 103 389 V3.2.1 (2019-12)
4 Reference System Architecture
The system architecture used to identify the interface that is the subject of the present document is a simplification of a
GSM-R system down to a minimum of logical entities relevant to the present document.
Within the context of the present document a GSM-R system is logically divided into a GSM-R Network and a Fixed
Terminal Subsystem. The interface between the Mobile Terminals and the NSS as well as the interface between the
Fixed Terminals and the FTS are explicitly not addressed in the present document. The focus
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