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6.3.2 CN Domain Identifier
ETSI TS 123 003 [1], clause 12.2 applies.
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6.3.3 CN Domain Indicator
ETSI TS 125 413 [4], clause 9.2.1.5 applies. ETSI ETSI TS 102 744-1-2 V1.1.1 (2015-10) 20
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6.3.4 RNC Identifier
ETSI TS 123 003 [1], clause 12.3 applies. A RNC node is uniquely identified by its RNC Identifier (RNC ID). The Global RNC Identifier is composed from the PLMN ID and the RNC ID. <RNC ID> ::= <BITSTRING(12)> <GLOBAL RNC ID> ::= <PLMN ID><RNC ID> = <MCC><MNC><RNC ID> The RNC Identifier <RNC ID> refers to the same networ...
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6.3.5 RNC Addresses
ETSI TS 123 060 [2], clause 14.12 applies.
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6.3.6 Service Area Identifier (SAI)
ETSI TS 123 003 [1], clause 12.4 applies. In UMTS, the Service Area Identifier (SAI) is used to indicate the location of a UE to the CN. It is defined as an area covered by one or more cells belonging to the same Location Area. The Service Area Identifier (SAI) is composed from the Service Area Code (SAC) together with...
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6.3.7 Permanent NAS UE Identity
ETSI TS 125 413 [4], clause 9.2.3.1 and ETSI TS 123 221 [i.2], clause 6.16 refer. "Permanent NAS UE Identity" refers to either IMSI or IMEI. The IMSI is transferred from the CN to the RNC using the RANAP Common ID elementary RANAP procedure. When an Iu connection is established, the CN performs the RANAP Common ID proc...
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6.3.8 Temporary NAS UE Identity
ETSI TS 125 413 [4], clause 9.2.3.2 applies. "Temporary NAS UE Identity" refers to either TMSI or P-TMSI.
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6.3.9 Radio Access Bearer Identity (RAB ID)
ETSI TS 125 413 [4], clause 9.2.1.2 applies. The Radio Access Bearer Identity (RAB ID) uniquely identifies the radio access bearer for a specific CN domain for a particular UE. The RAB ID is unique over one Iu connection. The purpose of the RAB ID is to provide an association between a NAS data stream (UE to CN) and th...
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6.3.10 Iu Signalling Connection Identifier
ETSI TS 125 413 [4], clause 9.2.1.38 applies. The Iu Signalling Connection Identifier is a reference to an Iu signalling connection. It is assigned by the RNC during the RANAP Initial UE Message elementary procedure and the CN is required to store and remember it for the duration of the Iu connection. The Iu Signalling...
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6.3.11 Paging Area Identity
ETSI TS 125 413 [4], clause 9.2.1.21 applies. "Paging Area Identity" refers either to LAI or RAI. In either case, LAI (or RAI) in the satellite context uniquely addresses a satellite beam.
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7 System Information
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7.1 General Principles
The broadcast of System Information (SI) by the RNC and the corresponding reception of SI by the UE is a function of the Bearer Control Layer. The RNC broadcasts System Information on the Global Beam Common Channels as well as on all Primary Shared Access Bearers. The System Information contains information elements re...
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7.2 System Information in Idle and Connected Modes
In the RRC-IDLE mode, when the UE camps on a Primary Shared Access Bearer (PSAB) in a particular spot beam, it receives all valid System Information (SI) for that spot beam on the PSAB common channel. The received SI is the "current system information" (see Figure 7.1). When the UE-Specific Signalling Connection is est...
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7.3 System Information Elements
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7.3.0 General
Information elements of the Bulletin Board (including System Information) are defined as part of the Bearer Control layer interface definition in ETSI TS 102 744-3-1 [i.12]. The information elements can be divided into three categories, based on their scope (i.e. the layer of the protocol stack in which they are releva...
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7.3.1 Access Stratum System Information
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7.3.1.0 General
Elements of the Access Stratum System Information control or affect Adaptation Layer behaviour, specifically the Registration Manager (REGM, see ETSI TS 102 744-3-6 [i.15]). The Access Stratum System Information set is divided into five subsets: NOTE: Details of the following Information Elements and Attribute Value Pa...
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7.3.1.1 Primary Bearer
The Primary Bearer information element identifies the primary shared access bearer for a particular spot beam. Zero or more instances of this element may be present in the AS system information. If one or more instances are present, then this indicates that the current beam (i.e. the beam in which this system informati...
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7.3.1.2 PLMN ID
The PLMN ID information element indicates the availability of a particular Public Land Mobile Network using the satellite radio access network. One or more instances of this element may be present in the AS system information. If the current beam (i.e. the beam in which this system information is being received) is sui...
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7.3.1.3 Access Control
The Access Control information element controls the initial access to the RNC. One instance of this element shall be present in the AS system information only if the current beam is suitable for performing the Registration procedure and access restrictions in the current beam are to be enforced. When present, only thos...
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7.3.1.4 Spot Beam Map
The Spot Beam Map information element is a data structure which identifies one or more spot beams and provides their geographical boundaries. Zero or one instance of this element shall be present in the AS system information. If this element is present then the UE shall use the position-assisted spot beam selection met...
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7.3.1.5 Common Signalling Retry
The Common Signalling Retry information element defines the number of times that the UE may repeat the Registration procedure and the timeout between attempts. One instance of this element shall be present in the AS system information only if the current spot beam is suitable for performing the Registration procedure.
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7.3.1.6 GPS Policy Info
The GPS Policy Info AVP informs the UE of the settings of GPS position-related security policies in the RNC. Zero or more instances of this element shall be present in the AS system information only if the current spot beam is suitable for performing the Registration procedure. If present, the UE may (optionally) use t...
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7.3.2 Non-Access Stratum System Information
Elements of the Non-Access Stratum System Information control or affect MM and GMM behaviour. The information set, called "Core Network System Information" in ETSI TS 124 008 [i.4], clause 10.5.1.12 is divided into three subsets: • Common NAS System Information: Location Area Code (LAC) • CS Domain Specific System Info...
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8 UE Idle Mode Behaviour
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8.1 NAS/AS Functional Divisions
The functional divisions between the UE Non-Access Stratum (NAS) and UE Access Stratum (AS) in RRC-IDLE mode are shown in Table 8.1. The NAS/AS divisions for the satellite network are the same as for UMTS with the exception that "spot beam selection and reselection" replaces the concept of "cell selection and reselecti...
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8.2 Idle Mode Procedures
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8.2.1 Non-Access Stratum
The Non-Access Stratum Idle Mode procedures described in ETSI TS 123 122 [3] apply for the satellite network with appropriate changes in terminology ("spot beam" replaces "cell" wherever used).
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8.2.2 Access Stratum (Usl Stratum)
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8.2.2.0 General
ETSI TS 125 304 [i.10] describes the Access Stratum Idle Mode procedures. The general principles in ETSI TS 125 304 [i.10] apply to the satellite network. However the specific properties of the satellite link compared to the terrestrial link used in UMTS requires different methods for the satellite network. There are t...
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8.2.2.1 PLMN Discovery
PLMN Discovery involves interaction between the Non-Access Stratum and the Adaptation Layer to determine if a particular UMTS network is available. When the UE is initialized, the NAS selects a PLMN according to the priority scheme defined in ETSI TS 123 122 [3]. The NAS then requests the Adaptation Layer to determine ...
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8.2.2.2 PSAB Discovery
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8.2.2.2.1 Method
PSAB Discovery involves interaction between the Adaptation Layer and the Bearer Control layer to locate a Primary Shared Access Bearer (PSAB) which can be used for registration with the RNC. PSAB Discovery is also known as spot beam selection. The Adaptation Layer provides the Bearer Control Layer with a list of one or...
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8.2.2.2.2 Spot Beam Selection Methods
The PSAB Discovery procedure supports spot beam selection by two methods: carrier scan and position-assisted. In the carrier scanning method, described earlier, the Bearer Control layer tunes in turn to all of the candidate PSAB frequencies identified by the Adaptation Layer and locks on to the PSAB with the highest si...
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9 Sleep Mode
Sleep Mode is the satellite replacement for the terrestrial UMTS Discontinuous Reception (DRX) mode. In Sleep Mode the UE reduces power consumption by receiving and processing the forward channel only for small fractions of a regular interval of time. During the period the UE is "awake", the UE may receive Paging messa...
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1 Scope
The present document specifies the mandatory requirements for User Equipment (UE) implementing multiple voice communications via the Voice-over-IP (VoIP) service for the Family SL satellite radio interface.
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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 reference document (including any amendments) applies. Referenced documents which ar...
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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 reference document (including any amendments) applies. NOTE: While any hyperlinks in...
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3 Symbols and abbreviations
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3.1 Symbols
For the purposes of the present document, the symbols given in ETSI TS 102 744-1-4 [3], clause 3 apply.
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3.2 Abbreviations
For the purposes of the present document, the abbreviations given in ETSI TS 102 744-1-4 [3], clause 3 apply. ETSI ETSI TS 102 744-4-1 V1.1.1 (2015-10) 7
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4 General
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4.1 Overview
The Circuit Switched (CS) domain is designed to carry only one active voice call per subscriber identity at any time. For User Equipment (UE) that is shared among users, e.g. on ships or aircraft, it is desirable to overcome this limitation and to offer the ability to handle more than one concurrent voice call. Additio...
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4.2 System architecture
The solution described in the present document is based upon the integration of a VoIP domain containing infrastructure operating in tandem with the Circuit Switched domain and operating via the Packet Switched Domain. A simplified representation of the overall system architecture is represented in Figure 4.2, where th...
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4.3 Access to the VoIP domain
The Mobile Terminal registers with the Radio Access Network and then initiates an Attach procedure towards both the CS and PS domains. During these Attach procedures the MT is authenticated, and Security Mode procedures are initiated to secure the signalling and user-plane between the Radio Access Network and the Mobil...
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5 Mobile PBX Requirements
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5.1 General Requirements
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5.1.1 Number of Concurrent Calls
The Mobile PBX shall support a minimum of four concurrent voice calls (one call placed via the CS domain and an additional three placed through the PS domain). It should support up to nine concurrent voice calls. NOTE: The maximum number of concurrent voice calls supported is determined by the subscribers' service pack...
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5.1.2 Support for Direct Inward Dialling
Additional MSISDNs (AMISDNs) may be allocated to a UE network subscription to provide support for Direct Inward Dialling (DID) from another network to individual telephone handsets connected to the Mobile PBX. These DID numbers shall be provided to the subscriber when they request a multiple voice service package from ...
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5.1.3 Support for Short Code Dialling
The Mobile PBX shall support Short Code Dialling in both service domains. ETSI ETSI TS 102 744-4-1 V1.1.1 (2015-10) 10
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5.1.4 Support for Call Barring
The Mobile PBX should provide means to implement local barring of outgoing calls (whitelist and/or blacklist) to be applied either to individual handsets or to a group of handsets.
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5.1.5 Configuration Requirements
The Mobile PBX shall provide a suitable, user friendly interface for its configuration. As a minimum, the interface shall allow the user to change the following parameters through a suitable user interface: • Access Point Name (APN) string [apn_name] • SIP Domain Name string [sip_domain_name] It is recommended that the...
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5.2 Functional Requirements
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5.2.1 Domain Selection Requirements
When a call is placed from a handset connected to the Mobile PBX, and this handset has been allocated the primary MSISDN (or no MSISDN at all), then the Mobile PBX shall always check whether the CS domain can be used to place a call. Only if the CS domain is not available, then the call shall be placed via the VoIP ser...
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5.2.2 Circuit Switched Call Handling Requirements
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5.2.2.1 CS Domain Supplementary Services Requirements
The Call Forward on Busy (CFB) and Call Waiting (CW) supplementary services settings for TS11 (speech) calls configured in the network shall not be modified as this may cause the multi voice service to malfunction. Preferably, if a Mobile PBX is integrated with a UE, the user shall not be able to modify CFB and CW sett...
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5.2.2.2 Support for Direct Inward Dialling in the CS Domain
The Network will provide the Destination MSISDN number in the Call Control SETUP message in the "Called party BCD Number" Information Element (IEI 0x5E, see ETSI TS 124 008 [5], clause 9.3.23.1, Table 9.70). The Mobile PBX shall use the Called Party BCD Number to direct the incoming call to the appropriate handset asso...
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5.2.2.3 Call Waiting in the CS Domain
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5.2.2.3.0 General
The Call Waiting behaviour specified in this clause is mandatory for Mobile PBX integrated with the UE. For a Mobile PBX external to the UE, the functionality may be limited due to the circuit switched interface between the UE and the Mobile PBX, hence the requirements are desirable but not mandatory for external Mobil...
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5.2.2.3.1 CS Voice Calls
The Call Waiting behaviour shall be as follows: • If the CS domain is busy in a call, then the Mobile PBX shall check the Called Party Number in the CC:Setup message and determine whether the handset associated with the Called Party Number is engaged in a call: - If the Handset is busy in the CS domain, then normal CS ...
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5.2.3 Domain Independent Supplementary Services Settings
Both the CS and SIP domain network elements (MSC, SIP Server) allow the user to configure supplementary services settings, subject to subscription, to be changed from the UE. As it cannot be determined in advance, which domain will be used to place a call, it is desirable for the same settings to be applied to both dom...
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5.3 Protocol Requirements
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5.3.0 General
Several protocol parameters are referenced in the following subclauses. These are shown in italics and surrounded by square brackets, e.g. [sip_domain_name]. Some parameters are explicitly configurable through a user interface (see clause 5.1.5) while others are held inside the UE and need to be retrieved as required e...
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5.3.1 Session Management Requirements
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5.3.1.0 General
In order to allow incoming and outgoing calls via the VoIP service domain, the Mobile PBX requires IP connectivity to the SIP Server and to the Media Gateway in the satellite network.
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5.3.1.1 Primary PDP Context Activation
On start-up, the Mobile PBX shall verify whether the UE is attached to the PS domain and if that is the case, initiate the activation of a primary background PDP context towards an APN dedicated to the provision of IP transport to the VoIP service domain. On successful activation of the PDP Context, the Mobile PBX shal...
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5.3.1.2 Secondary PDP Context Definition
A secondary streaming PDP context linked to the primary background PDP context referred to above, shall be defined but only activated when a call is being placed via the VoIP service domain. The Traffic Flow Template (TFT) for the secondary streaming PDP context shall be defined such that only the RTP voice stream(s) a...
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5.3.1.3 Secondary PDP Context Activation, Modification and Deactivation
The Mobile PBX shall keep track of VoIP calls being set up and cleared: • When the first call is set up in the VoIP service domain, the Mobile PBX shall activate the secondary PDP context with a requested Guaranteed Bitrate as specified in clause 5.3.3.2 below prior to sending a SIP Invite to the SIP server. • Whenever...
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5.3.2 SIP Protocol Requirements
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5.3.2.0 General
The Mobile PBX shall implement a SIP User Agent as specified in SIP version 2.0 (see IETF RFC 3261 [1]). Only UDP transport shall be used for all SIP signalling. The PBX shall send all SIP signalling to the SIP server using UDP destination port 5060 but may select a different UDP to receive SIP messages. If using a por...
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5.3.2.1 SIP Registration
The International Mobile Subscriber Identity (IMSI) stored on the USIM within the UE shall be used as the SIP User Name [imsi]. After successful activation of the primary PDP Context, the Mobile PBX shall send a SIP:REGISTER message to the SIP Server at the IP address resolved from the SIP Domain Name string. If a vali...
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5.3.2.2 SIP Options
The Mobile PBX shall respond to SIP OPTIONS polling from the network.
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5.3.2.3 SIP Call Setup (originating from PBX)
To originate a SIP call from the PBX, the PBX shall send a SIP INVITE message. In addition to the requirements in clause 10 of IETF RFC 3261 [1], Table 5.2 specifies those fields which require specific values to be used. Table 5.2: Field Values for SIP:INVITE Message Field Name Field Value From: <sip:[calling_number]@[...
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5.3.2.4 SIP Call Setup (terminating on PBX)
Incoming SIP INVITEs will carry either the primary or an additional MS-ISDN as the called number in the SIP:To header in the format: <sip:[called_number]@[sip_domain_name]> ETSI ETSI TS 102 744-4-1 V1.1.1 (2015-10) 14 where the [called_number] is presented in National Number Format, i.e. without the satellite network p...
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5.3.2.5 SIP Call Maintenance
The network may send reINVITE at regular intervals on every call to check that the SIP User Agent is still contactable and to check that the call is active in order to ensure accurate billing. The network fails the call if the Mobile PBX SIP User Agent does not reply or replies negatively. The Mobile PBX shall respond ...
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5.3.3 Media Handling Requirements
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5.3.3.1 Codec
The Mobile PBX shall support G.729 or G.729A and may optionally support variant G.729B.
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5.3.3.2 Codec Frame Packetization
The Mobile PBX shall support sending multiple codec frames in a single RTP packet as determined from the SDP ptime attribute sent by the network to the Mobile PBX in the SIP:INVITE, SIP:183 Session Progress and SIP:200 OK messages. The following ptime values shall be supported: 20 (default), 40, and 80 ms. The Mobile P...
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5.3.3.3 DTMF Handling
DTMF Tones to and from the Mobile PBX shall be carried over RTP as telephone-events in accordance with IETF RFC 4733 [2].
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5.3.4 SIP Domain Supplementary Services Requirements
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5.3.4.1 Calling Number Presentation on Inbound Calls
For Inbound calls the Calling Number is carried in the SIP:From header in the following format: <sip:[calling_number]@[sip_domain_name]> The Calling number on Inbound Calls is normally presented in International Format with a leading "+" or "00". If the caller has withheld their number or if the number is not available...
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5.3.4.2 Calling Number Presentation on Outbound Calls
For Outbound calls the Calling Number is carried in the SIP:From header in the format: <sip:[calling_number]@[sip_domain_name]> If Calling Number Presentation on Outbound Calls is selected by the user then the Mobile PBX shall set the [calling_number] parameter in the SIP:From Header. If individual DID numbers are allo...
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5.3.4.3 Supplementary Services Settings related to Calling Number Delivery
The Mobile PBX shall optionally allow the user to block or unblock Calling Number Delivery for the entire PBX. Annex A: VoIP Domain Call Service Codes provides a list of applicable codes used for this purpose in the SIP domain.
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5.3.4.4 Call Waiting and Call Hold in the SIP Domain
Call Waiting and Call Hold is not supported by the Multi-Voice SIP server in the network. However, a Mobile PBX may support these features locally towards connected handsets. ETSI ETSI TS 102 744-4-1 V1.1.1 (2015-10) 16 Annex A (normative): VoIP Domain Call Service Codes Call Service Code Confirmation Busy Call Forward...
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1 Scope
The present document is the baseline for the definition of the Release 2 Security services and Architecture.
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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 102 731 [1] apply.
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3.2 Symbols
For the purposes of the present document, the symbols given in ETSI TS 102 731 [1] apply.
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3.3 Abbreviations
For the purposes of the present document, the abbreviations given in ETSI TS 102 731 [1] apply. ETSI ETSI TS 102 731 V2.0.0 (2022-11) 6 4 Use of references and numbering in Release 2 specifications TC ITS specifications are developed in Releases and each Release needs to be self-consistent. The following rules apply: •...
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5 Normative Provisions
ETSI TS 102 731 [1] shall apply altogether. ETSI ETSI TS 102 731 V2.0.0 (2022-11) 7 History Document history V1.1.1 September 2010 Publication V2.0.0 November 2022 Publication
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1 Scope
The present document defines the GEM platform. GEM is applicable for specifications and standards based on the GEM APIs, content formats, and semantic guarantees. The present document is firstly intended to be used by entities writing terminal specifications and/or standards based on GEM. Secondly it is intended for de...
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2 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 reference document (including any amendments) applies. Referenced documents which ar...
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2.1 Normative references
The following referenced documents are necessary for the application of the present document. [1] ETSI TS 102 727 (V1.1.1): "Digital Video Broadcasting (DVB); Multimedia Home Platform (MHP) Specification 1.2.2". NOTE: Available at http://www.etsi.org/deliver/etsi_ts/102700_102799/102727/. [2] ETSI TS 101 154 (V1.8.1): ...
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2.2 Informative references
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] Application Definition Blu-ray Disc Format - BD-J Baseline Application and Logical Model Definition for BD-ROM, March 2005. NOTE: http://www.blu-raydi...
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3 Definitions and abbreviations
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3.1 Definitions
For the purposes of the present document, the terms and definitions given in TS 102 034 [79] and the following terms and definitions apply: Application Program Interface (API): interface between an application and a particular feature, function or resource of GEM application: functional implementation realized as softw...