TS 102 744-4-1 - V1.1.1 - Satellite Earth Stations and
Contents
1. If a Secondary PDP Context Activation triggered by an incoming SIP INVITE i e Mobile Terminated Call Setup fails then the Mobile PBX shall clear the call that is being set up by sending a SIP CANCEL message to the SIP server If a Secondary PDP Context Modification triggered by an incoming SIP INVITE i e Mobile Terminated Call Setup fails then the Mobile PBX shall clear the call that is being set up by sending a SIP CANCEL message to the SIP server If a Secondary PDP Context Activation or Modification triggered by a Mobile Originated call setup from a handset connected to the Mobile PBX fails then the Mobile PBX shall abandon the call setup and signal the call setup failure to the handset as appropriate for the type of handset e g call failure tone or SIP Error message A failed Secondary PDP Context Modification may be followed by an immediate Network Initiated PDP Context Deactivation On receiving Secondary PDP Context Modification failure followed by a Network Initiated PDP Context Deactivation the Mobile PBX should attempt to re activate a Secondary PDP Context with a suitable Guaranteed Bitrate to carry all existing calls in the VoIP domain i e reverting to the Guaranteed Bitrate operational before the failed Secondary PDP Context Modification was attempted If this re activation fails the Mobile PBX shall clear all existing calls in the VoIP domain ETSI 13 ETSI TS 102 744 4 1 V1 1 1 2015 10 5 3 2 SIP Pr2. APN dedicated to the provision of IP transport to the VoIP service domain On successful activation of the PDP Context the Mobile PBX shall retain the IP address assigned to this PDP context contact_ip for use in SIP signalling ETSI 12 ETSI TS 102 744 4 1 V1 1 1 2015 10 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 are carried on the secondary streaming PDP context The TFT and equivalent uplink TFT may use UDP Port Number ranges or DiffServ Code Point DSCP values in the filter definition When using UDP Port Number ranges then it is recommended to use the destination port number range in the downlink TFT filter definition which covers the port number range that will be used by the Mobile PBX for RTP streams Likewise in the uplink TFT filter definition a source port number range should be defined with the same values Using DSCP values in the TFT filters requires the Mobile PBX to use a distinctive DSCP value for IP packets carrying RTP streams It is strongly recommended for the Mobile PBX to use the same DSCP value for RTP packets as the DSCP value that the Media Gateway is configured to use If RTCP packets are
3. domain to set up a call Only if the CS domain is unavailable i e one voice call is already in progress between the user and another party will any additional calls be set up using the VoIP domain The present document specifies the mandatory requirements for equipment implementing a User Agent in the VoIP service domain This User Agent is essentially providing a Private Branch Exchange PBX in the mobile domain as shown in Figure 4 1 and is therefore referred to as Mobile PBX throughout the present document E L Telephone M i ui User Telephone User Agent Mobile PBX Equipment UE De a n n I a NI a N Telephone Laptop Figure 4 1 Equipment Configuration for support of Multiple Voice service A Mobile PBX can be implemented either as an integral part of a UE or as external hardware that can be connected to an existing UE In general the requirements specified in the present document apply to both implementations unless indicated otherwise 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 the VoIP domain is functionally provided by a VoIP server ETSI 8 ETSI TS 102 744 4 1 V1 1 1 2015 10 M
4. provider prior to shipping the equipment The interface should also allow the user to manually configure the primary and any additional MSISDNs and to associate each MSISDN with a particular telephone handset This interface shall also provide for one telephone handset to be the default handset to receive incoming calls without a DID number 5 2 Functional Requirements 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 service domain When a call is placed from a handset connected to the Mobile PBX and this handset has been allocated one of the additional MSISDNs then the call shall always be placed via the VoIP service domain 5 2 2 Circuit Switched Call Handling Requirements 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 settings Alternatively for an external Mobile PBX the user manual shall carry appropriate warnings that
5. which are or may be or may become essential to the present document Foreword This Technical Specification TS has been produced by ETSI Technical Committee Satellite Earth Stations and Systems SES The present document is part 4 sub part 1 of a multi part deliverable Full details of the entire series can be found in ETSI TS 102 744 1 1 4 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 This multi part deliverable Release 1 defines a satellite radio interface that provides UMTS services to users of mobile terminals via geostationary GEO satellites in the frequency range 1 518 000 MHz to 1 559 000 MHz downlink and 1 626 500 MHz to 1 660 500 MHz and 1 668 000 MHz to 1 675 000 MHz uplink ETSI 6 ETSI TS 102 744 4 1 V1 1 1 2015 10 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 2 References 2 1 Normative references References are either specific identified by date of publication and or edition number or version
6. E TSI TS 102 44 4 1 V1 1 1 2015 10 TEC HNICAL SPECIACATION Satellite Earth Stations and Systems SES Family SL Satellite Radio Interface Release 1 Part 4 Enhanced Services and Applications Sub part 1 Multiple Voice Services 2 ETSI TS 102 744 4 1 V1 1 1 2015 10 Reference DTS SES 00299 4 1 Keywords 3GPP GPRS GSM GSO interface MSS radio satellite TDM TDMA UMTS ETSI 650 Route des Lucioles F 06921 Sophia Antipolis Cedex FRANCE Tel 33 4 92 94 42 00 Fax 33 4 93 65 47 16 Siret N 348 623 562 00017 NAF 742 C Association but non lucratif enregistr e la Sous Pr fecture de Grasse 06 N 7803 88 Important notice The present document can be downloaded from http www etsi org standards search The present document may be made available in electronic versions and or in print The content of any electronic and or print versions of the present document shall not be modified without the prior written authorization of ETSI In case of any existing or perceived difference in contents between such versions and or in print the only prevailing document is the print of the Portable Document Format PDF version kept on a specific network drive within ETSI Secretariat Users of the present document should be aware that the document may be subject to revision or change of status Information on the current status of this and other ETSI documents is available at http portal etsi org tb status stat
7. IP Options The Mobile PBX shall respond to SIP OPTIONS polling from the network 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 From lt sip calling_number sip_domain_name gt or lt sip imsi sip_domain_name gt lt sip dialled_digits sip_domain_name gt No SDP Offer is required in the SIP INVITE for Mobile PBX originated calls The Mobile PBX shall always adhere to the SDP Answer returned by the SIP Server in particular the prime parameter see clause 5 3 3 2 shall be applied to the outbound RTP stream The Mobile PBX shall provide Call Progress Tones towards the calling handset and convert any incoming SIP error messages to appropriate call failure tones or optionally voice announcements 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 lt sip called_number sip_domain_name gt ETSI 14 ETSI TS 102 744 4 1 V1 1 1 2015 10 where the called_number is presented in National Number Format i e without the satellite network provider s country code If individual DID numbers are allocated to specific handsets conne
8. SC VLR VolP Server Figure 4 2 Overall System Architecture for support of Multiple Voice service A new VoIP Domain provided via a VoIP server is introduced into the Family SL ground segment and provides an interface between the existing Circuit Switched domain and the Packet Switched domain for transport of multiple voice calls when the single CS voice circuit towards a mobile terminal is busy The interface with the PSTN is handled by the existing Circuit Switched domain for both mobile terminated and mobile originated calls The existing voice mail system will be used to support per MSISDN voice mail Multiple MSISDN numbers are allocated to the subscriber These numbers may be allocated to individual handsets by configuration of the mobile terminal or the external VoIP PBX Alternatively functionality may be provided in the mobile terminal or VoIP PBX to determine which handset to alert upon Mobile Terminated call initiation utilizing the primary MSISDN 4 3 Access to the VolP 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 Mobile Terminal ETSI 9 ETSI TS 102 744 4 1 V1 1 1 2015 10 MT RAN SGSN GGSN MSC GMSC HLR di location Updates CS Atta
9. all 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 Call Waiting Call Hold procedures should be invoked towards the handset giving the user the choice of accepting or rejecting the call If the Handset is busy in the VoIP domain the call shall automatically be rejected in the CS domain with Cause Code 17 user busy In this case the call will then be diverted to the VoIP domain and the call will be again be presented via the VoIP domain allowing the user to accept or reject the incoming call and or put the ongoing call on hold If the Handset is idle the call shall automatically be rejected in the CS domain with Cause Code 17 user busy In this case the call will then be diverted to the VoIP domain and the call will again be presented via the VoIP domain for the user to accept or reject 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 domains where applicab
10. any changes to the CFB and CW settings may cause malfunction of the Multi Voice service 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 Ox5E 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 associated with the DID number If the Called Party BCD Number IE is absent or if the Called Party BCD Number is not available to the Mobile PBX e g if the Mobile PBX is implemented as an external device connected to a UE then the handling of such calls shall be specific to the manufacturer s implementation 5 2 2 3 Call Waiting in the CS Domain 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 Mobile PBX implementations For user equipment that does not support Call Waiting in the CS domain the following subclauses are not applicable In this case incoming calls will always be routed to the VoIP domain if the CS domain is busy ETSI 11 ETSI TS 102 744 4 1 V1 1 1 2015 10 5 2 2 3 1 CS Voice Calls The C
11. c 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 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 Not applicable 3 symbols and abbreviations 3 1 symbols For the purposes of the present document the symbols given in ETSI TS 102 744 1 4 3 clause 3 apply 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 7 ETSI TS 102 744 4 1 V1 1 1 2015 10 4 General 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 Additional voice calls can be provided by a Voice over IP VoIP service domain using the Packet Switched PS domain as its transport The general principle of operation assumes that the first preference is to use the CS
12. ch Insert Subs Details Routing Area Update Insert Subs Details nn O PS Attach IMS APN IP Address Allocation Primary PDP Context Activatio SIP Registration of IMSI SIP Registration T Figure 4 3 Simplified Presentation of Initial Access Procedure i Refresh Subscriber Data Conditional on successful CS and PS attach procedures the mobile terminal initiates a background Primary PDP context towards the Inmarsat VoIP APN which is authorized by the HLR if the subscriber is granted access to Multiple Voice services Conditional on the Mobile Terminal subscriber being authorized to access VoIP domain infrastructure an IP address is allocated to the mobile terminal and the PDP context activated The Mobile Terminal then performs a DNS lookup to obtain the IP address of the SIP server within the VoIP domain the fully qualified domain name of the SIP server will be configured in the Mobile Terminal The Mobile Terminal then initiates a SIP registration procedure towards the SIP server 9 Mobile PBX Requirements 5 1 General Requirements 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 s
13. cted to the Mobile PBX then the incoming call shall be routed to that handset accordingly otherwise the call shall be routed in accordance with the appropriate user configuration Incoming SIP INVITEs will always carry an SDP offer the Mobile PBX shall accept the first codec in the list as well as the prime parameter specified The Mobile PBX is not required to provide audible ringback tones towards the caller 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 positively to the re INVITE if and only if there is already an active call matching that specified in the re INVITE 5 3 3 Media Handling Requirements 5 3 3 1 Codec The Mobile PBX shall support G 729 or G 729A and may optionally support variant G 729B 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 PBX shall determine the required Guaranteed Uplink and Downlink Bitrates for the secondary Strea
14. e word anonymous The Mobile PBX should present the Calling Line ID to the handset to which the call is directed if the handset supports Calling Line ID ETSI 15 ETSI TS 102 744 4 1 V1 1 1 2015 10 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 lt sip calling_number sip_domain_name gt 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 allocated to the handset connected to the Mobile PBX and it is intended to present the DID number of the handset as the Calling Number then the corresponding individual number in National Number Format shall be used in the calling_number parameter 1 e without the satellite network provider s country code If the calling number is not configured in the Mobile PBX no DID number is allocated to the handset placing the call or if it is not intended to present the DID number allocated to the handset then the Mobile PBX shall use the following format lt sip imsi sip_domain_name gt 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 t
15. ear 14 NS e Codec Fame PAC he Zaini 14 5 3 3 3 BEMETT nO ir in EEEN 14 5 3 4 SIP Domain Supplementary Services Requirements 14 5 3 4 1 Calling Number Presentation on Inbound CallS 14 5 3 4 2 Calling Number Presentation on Outbound Calls 15 5 3 4 3 Supplementary Services Settings related to Calling Number Dellvery in 15 5 3 4 4 Call W ating dnd Call Hold mthe SEP Domanda ola 15 ETSI 4 ETSI TS 102 744 4 1 V1 1 1 2015 10 Annex A normative VoIP Domain Call Service Codes csscccsssssssccccssssssccccccsssccccccsssscccesess 16 History ETSI 5 ETSI TS 102 744 4 1 V1 1 1 2015 10 Intellectual Property Rights IPRs essential or potentially essential to the present document 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 http 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
16. ens 9 5 1 1 ING ber OI Concur nt C alls sieni noiiente ana 9 ye Bz Support tor Direct nward Dil 9 5 1 3 Suppor TOR Sbor Code Dalin see ace 9 5 1 4 Supportior Call Barrmmoe di cca 10 5 1 5 At AA e 10 SZ Funcional Requiem bc ia 10 dz Domam Seleeton R guiremenib lia 10 52 2 Circuit Switched Call Handling Requirements 10 22 1 CS Domain Supplementary Services Requirements 10 S222 Support for Direct Inward Dialling in the CS Domain i 10 S225 Call Waiting 1n the ES DOM as 10 522 30 A A A E A E 10 5223A CS Voice Calls a cidos 11 Id Domain Independent Supplementary Services Settings 11 5 3 Protocol Requiem os 11 5 3 0 Cs 11 5 3 1 Session Management Requiero 11 5 3 1 0 O a A A AS 11 del Primary PDP Come xt Acivalo rele ni 11 TEZ Secondary PDP Context De ini sissen a a a aO 12 5 3 143 Secondary PDP Context Activation Modification and Deactivation 12 a SIP Prorocol Regane me i Sensi Gaacanersnenbobenosausanceusbacsaadee veda snes E E aO e OERE 13 5 3 2 0 GICTICT As ode snenagas neern oE E E O EEE E O 13 AN A sai inesadaninsebietsrecostubobawh owned oanetaiins E EE 13 nye aed STD ile i erence me eer reer rere r ete ies Gren era 13 Za SIP Gall Settip 0nematn amp fron PB rada 13 5 3 2 4 SIP Call Setup lermunati ne ON EPD X rele A aa 13 VIZI A abano ria il ana 14 569 Media Handlins Re quite Medir ciao nO nee LAREDO Rd ap 14 5 3 3 1 Te O
17. ervice package and is enforced by the network 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 the satellite network service provider The Mobile PBX may optionally support telephone handsets that are not associated with a DID number Such handsets would typically be used for outbound calls only 5 1 3 Support for Short Code Dialling The Mobile PBX shall support Short Code Dialling in both service domains ETSI 10 ETSI TS 102 744 4 1 V1 1 1 2015 10 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 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 e Access Point Name APN string apn_name o SIP Domain Name string sip_domain_name It is recommended that these parameters are pre configured by the manufacturer with default strings as advised by the satellite network service
18. he SIP domain 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 16 ETSI TS 102 744 4 1 V1 1 1 2015 10 Annex A normative VolP Domain Call Service Codes Call Service Busy Call Forwarding enable 90 lt number to forward to gt Tone Tone Busy Call Forwarding disable Calling Number Delivery enable Calling Number Delivery disable Calling Number Delivery Blocking single call 67 lt called number gt Calling Number Delivery Blocking override 82 lt called number gt ETSI 17 ETSI TS 102 744 4 1 V1 1 1 2015 10 History Document history ETSI
19. le The Mobile PBX shall optionally provide a feature e g through the user interface which simultaneously applies changes to supplementary services settings to both domains such that the same feature is provided regardless of the serving domain Supplementary Service Codes applicable in the VoIP domain are specified in Annex A VoIP Domain Call Service Codes 5 3 Protocol Requirements 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 either through internal or external interfaces depending on whether the Mobile PBX is integrated within the UE or external to it NOTE An external Mobile PBX would typically obtain such parameters through AT Commands and Responses This is outside the scope of the present document 5 3 1 Session Management Requirements 5 910 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 5 9 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
20. ming PDP Context from the ptime attribute as follows for ptime 20 ms 26 kbit s for ptime 40 ms 18 kbit s for ptime 80 ms 12 kbit s For mobile terminated calls the Mobile PBX shall obtain the requested ptime value from the incoming SIP INVITE message to select the appropriate Bitrate for the subsequent Secondary PDP Context Activation or Modification For mobile originated calls the Mobile PBX shall assume that the last ptime value provided by the SIP server will also apply to the call which is in the process of being set up to select the appropriate Bitrate for the Secondary PDP Context Activation The Mobile PBX shall store the last ptime received from the SIP server in non volatile memory such that the value is retained during a reset reboot or power cycling of the mobile PBX 9 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 5 3 4 SIP Domain Supplementary Services Requirements 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 lt sip calling_number sip_domain_name gt 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 for other reasons then the calling_number parameter will contain th
21. 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 are not found to be publicly available in the expected location might be found at http 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 IETF RFC 3261 2002 SIP Session Initiation Protocol J Rosenberg 2 IETF RFC 4733 2006 RTP Payload for DTMF Digits Telephony Tones and Telephony Signals H Schulzrinne 3 ETSI TS 102 744 1 4 Satellite Earth Stations and Systems SES Family SL Satellite Radio Interface Release 1 Part 1 General Specifications Sub part 4 Applicable External Specifications Symbols and Abbreviations 4 ETSI TS 102 744 1 1 Satellite Earth Stations and Systems SES Family SL Satellite Radio Interface Release 1 Part 1 General Specifications Sub part 1 Services and Architectures 5 ETSI TS 124 008 Digital cellular telecommunications system Phase 2 Universal Mobile Telecommunications System UMTS LTE Mobile radio interface Layer 3 specification Core network protocols Stage 3 3GPP TS 24 008 2 2 Informative references References are either specifi
22. otocol Requirements 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 port number other than 5060 the PBX shall indicate its SIP port in the parameter port_no as specified below otherwise the use of this parameter is optional 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 valid IP address is specified instead of a SIP Domain Name string then the IP address shall be used instead The SIP REGISTER message shall comply with the requirements in clause 10 of IETF RFC 3261 1 with Table 5 1 specifying those fields which require specific values to be used Table 5 1 Field Values for SIP REGISTER Message From I lt sip limsiO sio domain_namel gt To lt sip ims l sip domain name gt User Agent The field value shall identify the manufacturer and version of the Mobile PBX Contact lt sip ims f contact ip port_no gt 36000 Expires 3600 E 5 3 2 2 S
23. sent from a Mobile PBX then the TFT filters shall be configured such that the RTCP packets are carried over the primary background PDP Context in order to optimize streaming bandwidth usage NOTE The network side Media Gateway is configured not to send RTCP packets to the client 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 e 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 e Whenever another call is set up in the VoIP domain up to the maximum number of concurrent calls supported by the Mobile PBX the requested Guaranteed Bitrate shall be modified upwards to accommodate the additional RTP stream over the same secondary PDP Context prior to sending a SIP Invite to the SIP server e Whenever a call is cleared in the VoIP domain then the Mobile PBX shall first determine whether there are any other calls ongoing in the VoIP domain If other calls are still ongoing then the requested Guaranteed Bitrate shall be modified downwards such that the remaining RTP stream s can be carried over the secondary PDP context If the call being cleared is the only call in the VoIP domain then the Mobile PBX shall request the deactivation of the secondary PDP context
24. us asp If you find errors in the present document please send your comment to one of the following services https portal etsi org People CommiteeSupportStaff aspx Copyright Notification No part may be reproduced or utilized in any form or by any means electronic or mechanical including photocopying and microfilm except as authorized by written permission of ETSI The content of the PDF version shall not be modified without the written authorization of ETSI The copyright and the foregoing restriction extend to reproduction in all media European Telecommunications Standards Institute 2015 All rights reserved DECT PLUGTESTS UMTS and the ETSI logo are Trade Marks of ETSI registered for the benefit of its Members 3GPP and LTE are Trade Marks of ETSI registered for the benefit of its Members and of the 3GPP Organizational Partners GSM and the GSM logo are Trade Marks registered and owned by the GSM Association ETSI 3 ETSI TS 102 744 4 1 V1 1 1 2015 10 Contents intellectual Property RI MS pi A 5 NN A A ili O O hi 5 Nas AA A scat sch T 5 Introduction iii 5 l SCOPE ina REO O E Ani ai 6 2 Riel 6 2 1 Normative ille aa 6 22 O Lelli az 6 3 Symbols and DEIA dci coca 6 3 1 SA CO RR ROERO ITA 6 3 2 Abbanoa 6 4 Cicala 7 4 1 DV CVS Ws lecco J 4 2 SYC EDICE a PO PEE OE a a a 7 4 3 Accesso MeV oP domai enean e elia ae ai 8 5 MobilePBX Reguinemenbs succes 9 5 1 General Regae MEN Seana a E a a a A a E A e
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