Internet DRAFT - draft-canal-p909-pint-spirits

draft-canal-p909-pint-spirits



EURESCOM P909 contribution to PINT and SPIRITS                [Page 1]


PINT Working Group                            Valerie Blavette
SPIRITS Working Droup                         Gianni Canal 
Internet Draft                                Uwe Herzog 
                                              Carlo Alberto Licciardi 
                                              Stephane Tuffin
Expires: September 2000


             EURESCOM P909 contribution to PINT and SPIRITS
       Interaction between Internet and PSTN to request services 
                      from one domain to the other

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   Status of this Memo 
   
   This  document  is  an Internet-Draft and is in full conformance with
   all provisions of Section 10 of RFC2026.  
   
   Internet-Drafts are working documents  of  the  Internet  Engineering
   Task Force  (IETF),  its  areas,  and  its working groups.  Note that
   other   groups   may   also   distribute   working    documents    as
   Internet-Drafts.  
   
   Internet-Drafts are draft documents valid for a maximum of six months 
   and  may be updated, replaced, or obsoleted by other documents at any
   time.  It  is  inappropriate  to  use  Internet-Drafts  as  reference
   material or to cite them other than as "work in progress."   
   
   The   list   of   current   Internet-Drafts   can   be   accessed  at
   http://www.ietf.org/ietf/1id-abstracts.txt  
   
   The list of Internet-Draft Shadow  Directories  can  be  accessed  at
   http://www.ietf.org/shadow.html.   
   
1. Abstract 
   
   This  document  is  intended  to  provide  input  to the IETF SPIRITS
   Working Group(SPIRITS WG) and to the IETF PINT  Working  Group  (PINT
   WG)  from  the  viewpoint  of  European  network  operators  that are
   involved in EURESCOM  P909  Project  "Enabling  Technologies  for  IN
   Evolution and IN-Internet Integration". The input is based on current 
   results that were achieved in the project.  As the project title says 
   P909 project  deals with IN-Internet integration issues.  The project
   has defined an architecture for IN-Internet inte gration and  it  has
   selected  and  described  some  IN-Internet  services  which  can  be
   interesting from the business perspective and  challenging  from  the
   technical perspective.  
   Some  of  these  services  are "officially" chartered in IETF: ICW in
   SPIRITS,  Click-to-Dial  (Request  to  Call)  as  well  as   proposed


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EURESCOM P909 contribution to PINT and SPIRITS                [Page 2]


   Conference Service in PINT. However, there are additional IN-Internet 
   convergence  services that P909 studies and that are included in this
   document only as examples.  
   Selected services can help  in  defining  requirements  both  in  the
   context  of  how  services  supported  by  IP network entities can be
   started from IN  (Intelligent  Network)  requests  and  how  Internet
   applications  can  request and enrich PSTN (Public Switched Telephone
   Network) telephony services.  
   In section 2 and 3 of this  Internet  Draft  you  find  some  general
   information on   EURESCOM  and  P909  project.    Section  4  briefly
   introduces  some  of  the  services  P909  has  selected,  section  5
   describes for some services why there is a need to notify entities in 
   the  Internet  from  PSTN/Intelligent  Networks  domain  in  order to
   provide the requested service features.    Section  6  describes  the
   service  requirements  for scenarios where the PSTN telephony service
   is requested from Internet.  
   Sections 7, 8 and 9  respectively  address  security  considerations,
   supply  references, and provide the authors addresses, as required by
   [1].  
   Section 10  acknowledges  individuals  providing  assistance  in  the
   creation of this document, and, finally, Section 11 provides Glossary 
   of terms.  
   
   TABLE OF CONTENTS 
   
   1. Abstract 
   2. EURESCOM 
   3. EURESCOM P909 Project 
   4. Overall Service Description 
   4.1 Internet Call Waiting 
   4.2 Virtual Second Line 
   4.3 Click-to-Dial 
   4.4 Distributed and Enhanced Call Center 
   4.5 Meeting Scheduler 
   4.6 Unified Communication 
   4.7 Virtual Presence 
   4.8 Virtual Private Network 
   5. IN service triggering internet services 
   5.1 Virtual Presence Service 
   5.2 Internet Call Waiting 
   5.3 Meeting Scheduler 
   5.4 Unified Communication 
   5.5 Distributed and Enhanced Call Center 
   6. IN service request from the Internet 
   6.1 Click-to-Dial 
   6.2 Meeting Scheduler 
   7. Security considerations 
   8. References 
   9. Authors addresses 
   10. Acknowledgements 
   11. Glossary 


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2. EURESCOM 
   
   EURESCOM(European  Institute  for  Research  and Strategic Studies in
   Telecommunications) is  an  Institute  for  performing  collaborative
   Projects   on   research  and  strategic  studies  in  all  areas  of
   telecommunications.  It  was  founded  in  1991  and  is  located  in
   Heidelberg,  Germany.  Currently,  there  are 24 Shareholders from 23
   European countries involved in EURESCOM and it is open to any network 
   operator or service provider who wishes to join.  EURESCOM's  overall
   mission  is  to carry out pre-competitive R&D Projects. The aim is to
   support the R&D Project  Shareholders  to  establish  future-oriented
   telecommunication networks  and  services.  In addition, EURESCOM has
   to carry out studies in order to be able  to  recommend  longer  term
   strategic options concerning R&D in services, networks and supporting 
   technologies.  More precisely, EURESCOM's objectives are to: 
   -   stimulate   and  co-ordinate  pre-competitive  and  pre-normative
   collaborative R&D Projects including field trials and pilot projects 
   - enable the development of harmonised strategies by its Shareholders 
   for the planning, operation and provision of future  European  public
   telecommunications networks and services 
   - contribute to Europe-wide service deployment and service usage 
   - contribute to European and world-wide standardisation 
   -  contribute  to  the  work  of  international  bodies  and relevant
   European R&D programmes 
   
3. EURESCOM P909 Project 
   
   Enabling technologies for IN evolution and IN-Internet integration is 
   the focus of P909. The IN-Internet convergence can  be  expressed  in
   terms  of  delivery  of specifications, and identification/evaluation
   and  integration  of  off-the-shelf  products   supported   by   some
   prototypes development.  In particular the project addresses: 
   
   -  Identification  of  IN-Internet  services in a competitive context
   (click-2-dial, unified messaging, ...) 
   
   -  Definition  (or  adaptation)  of  a  service   architecture   that
   facilitates the synergy between IN and Internet services; 
   
   -  Evolution  of  the Network Intelligence to support new IN services
   and integrated IN-Internet services in terms of: 
   
      - evolution of IN architecture (Reference and Business Models) and 
   elements (SCP, SMS, IN-IP, ...); 
      - integration of IN and Internet services (i.e.  to  integrate  IN
   and  Internet  in a simple call completion service with a transparent
   access, independent of caller and called terminal); 
      - Definition of an unified Call Control for  IN-Internet  services
   based on ongoing initiatives (e.g Parlay) 
   


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   -  Definition  of  guidelines  to  PNOs for step-wise introduction of
   middleware solutions.  
   
   - Investigation of protocols (e.g. IETF PINT, IETF SPIRITS, IETF SIP) 
   for Web and  IN  services  convergence  (IN  Services;  customisation
   through the Web, interfacing IN with 3rd party systems, definition of 
   open interfaces between SCP and Web server); 
   
   -  Selecting  and testing products maturity from IN and Internet side
   w.r.t.  the defined architecture; 
   
   - Integration and assembling of existing products (or prototypes)  in
   order to prototype middleware platforms for IN and Internet services; 
   this could lead to use gateway to access new SCPs (i.e. a CORBA SCP), 
   to  integrate  and control technologies and products such as Voice-IP
   gateways, PINT/SPIRITS gateways,  gatekeepers  and  MCUs  (Multimedia
   Conference Units) ITU H323 compliant.  
   
   -  Implementating services on top of IN-Internet middleware platforms
   (the definition of such platforms leads also  to  the  definition  of
   components  that  enable/facilitate  the access to other stakeholders
   and the related co-operation /federation).  
   
   Companies that are involved in P909 project are: Telefonica I+D,  KPN
   Research,  T-Nova DeutscheTelekom Innovationsgesellschaft mbH, France
   Telecom, CSELT/Telecom Italia, Telenor AS, Broadcom Eireann  Research
   and Portugal Telecom.  
   
4. Overall Service Description 
   
   P909 has  selected  a  line of services to study.  These services are
   interesting from  the  business  and  technical  perspective  in  the
   context of IN-Internet convergence.  
   
   Some  of  these  IN-Internet  services  are "officially" chartered or
   proposed in IETF: Click-to-Dial (Request to Call), Conference Service 
   in PINT and ICW in SPIRITS. However, there is a number of  additional
   IP-PSTN  convergence services that P909 studies and that are included
   in this document for example only purpose.  
   
   This section briefly describes the selected services.  
   
4.1 Internet Call Waiting 
   
   Internet call waiting service enables a user engaged  in  a  dial  up
   Internet  session  to  be  alerted when an incoming call has arrived.
   After the Internet user has been alerted, he is given several options 
   for  handling  the  call  e.g.,  forwarding  it,  sending  a  waiting
   announce/tone,  accepting  the call over PSTN suspending the Internet
   session, or accepting the call over IP  keeping  alive  the  Internet
   session.  


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4.2 Virtual Second Line 
   
   This  service  allows  the  subscriber to answer incoming phone calls
   while his/her single  telephone  line  is  busy  due  to  an  ongoing
   Internet session.    A  vocal  gateway  can  be used to transform the
   incoming PSTN call into a Voice over IP flow directed to the terminal 
   interconnected to the Internet.  In  this  way,  the  terminal  could
   manage  the  IP flow carrying the voice along with the other IP flows
   originated by the Web surfing.  
   
4.3 Click-to-Dial (Request-to-Call) 
   
   A user is able to initiate a telephone  call  by  clicking  a  button
   during a  web  session.    The  called address (as well as the caller
   address) is either an IP  address  or  a  phone  line  number.    The
   charging  party  could  be  either the initiator or one of the called
   parties.  
   
4.4 Distributed and Enhanced Call Center  
   
   The integration of IN and Internet would allow to decrease the  costs
   of the  call  centres.   A first opportunity is the implementation of
   network-based   call   centre   solutions   in   order   to   achieve
   distributed/virtual call  centres.    With  this approach, the actual
   necessary infrastructure in each call  centre  could  disappear,  and
   some new paradigms like teleworking could be used.  
   
4.5 Meeting Scheduler  
   
   Meeting  Scheduler  is  a  service that enables an user to have a web
   based user interface to schedule a meeting.  He decides the time  and
   the attendants.    The  timing  information  is  used  by an external
   application that launches the service via the access function.  Every 
   meeting attendant is confirmed, either with  an  e-mail  notification
   and  a  following confirmation process via a web page or with a phone
   call.  
   
4.6 Unified Communication 
   
   The Unified Communication Service allows users to send, retrieve  and
   receive  messages  disregarding the format and the terminal where the
   user is connected.  The user should be able to create and respond  to
   multimedia messages from any terminal and create and send any type of 
   message without  regard to the recipient's mailbox requirements.  The
   user should also be able to reply to messages  and  forward  messages
   with calls, and reply and forward calls with messages.  
   
4.7 Virtual Presence  
   
   "Virtual  Presence"  service  allows  its  subscribers  to be reached


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   anywhere, anyway (by using both asynchronous messages and  real  time
   communication)  and  from  any  terminal independently of the type of
   terminal he is logged to.  
   "Virtual Presence" extends the characteristics of a  Personal  Number
   service to  the  context of Internet/Telecom convergence.  The aim of
   the service is to provide an integrated set of features  that  enable
   for example a subscriber to control the incoming calls according to a 
   set  of personal screening/ routing rules (a sort of "net-secretary")
   and terminal registrations, that he/she could dynamically modify.  
   
4.8 Virtual Private Network  
   
   The Enhanced VPN Service would allow to use most of cases of the VoIP 
   paradigm.  The main characteristics of this service could be: 
   - Integration of several terminals into the  VPN:  fixed  and  mobile
   phones, PC, ...  
   -  Several  communication  modes  (Phone to phone, Phone to PC, PC to
   phone, PC to PC) 
   - Use of  traditional  features  in  VPN  service:  Unified  billing,
   Abbreviated  dialling,  Call  restrictions  (extra/intra  VPN calls),
   etc.  
   The service logic could  manage  information  about  the  connections
   between  the  VPN  positions,  in order to select the cheapest way to
   reach the destination of a call, either in an intra VPN call or in an 
   extra VPN call.  
   
5. IN service triggering to the internet 
   
   Some  IN  Services  could  be  extended  to  notify  events  or  send
   information to  the  Internet domain.  It should enable services like
   Internet Call Waiting, Virtual Presence, etc.  It must be possible to 
   have access to the Internet domain from IN for the purpose of 
   - Invoking Internet services, e.g.  forwarding of fax/SMS as email 
   - Notify call information 
   - Controlling VoIP resources 
   - Access to customer  and  service  information  stored  on  Internet
   Databases/Web Server,  e.g.  downloading share information from a Web
   server for transfer to a GSM phone using SMS.  
   
   The following services can well help in defining requirements in  the
   context  of  how  services  supported  by  IP network entities can be
   started from IN (Intelligent Network) requests.  
   
5.1 Virtual Presence Service 
   
   The VP service may work with a heterogeneous set of terminals:  e.g.,
   fixed  and  mobile  phones,  fax and PC, and new generation terminals
   (PDA, WebPhones,  WAP-based  mobile   phones).      The   association
   subscriber-terminal can be dynamically changed (personal mobility).  
   In  addition  a subscriber can be registered on multiple terminals of
   the same type.  A service subscriber has a single virtual identity in 


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   the system, referenced by a logical name, Personal Alias, (used  from
   the  Internet  context)  and  a  Personal  Number  (used both from IN
   context and Internet one).  More precisely the user can: 
   1. Register/De-register: (s)he can choose the terminals  where  (s)he
   decided to be contacted.  
   2. Configure: (s)he can define the group of terminals of common use.  
   3.  Set  personalization  rules: using a graphical application, (s)he
   can introduce and manage filtering rules to be  applied  to  incoming
   asynchronous or synchronous communication requests.  
   
   Rules   are   modeled   according   to   the   ECA   paradigm,   i.e.
   Event-Conditions-Actions: 
   
   1. An event is an external  stimulus  addressed  to  the  subscriber.
   Conditions  represent constraints on rules scope.  
   
   2.  Actions are operations performed if and only if all conditions of
   the rule are satisfied when the event occurs.  
   Rules are activated by a single event; they are evaluated  against  a
   set of conditions, and trigger a sequence of actions.  
   
   Service features that have an impact on the PSTN-to-IP context are: 
   
   -  Notifying  the subscriber about incoming synchronous communication
   request on his/her PC.  As an  example  considering  Gianni  who  had
   previously   set   some   customised   rules  for  handling  incoming
   communication requests from other users.  
   A user tries to contact Gianni from a phone  by  using  his  personal
   number  (1364555).  The  incoming  call  request  is processed by the
   service.  The service analyses the rules and one of them fires (event 
   and condition matches).  For example the  action  associated  to  the
   rule is to route the call to the Gianni's office PC. Then the service 
   forwards  the  call  to  the PC of the subscriber who is requested to
   accept the call through some standard API or network protocols  (e.g.
   IETF SIP). After she/he clicks on the appr opriate button the call is 
   completed and the two users are put through.  
   
   -  Notifying the subscriber about incoming asynchronous communication
   request (e.g. vocal message) on his/her PC.  
   A user tries to contact Gianni from a phone  by  using  his  personal
   number  (1364555).  The  incoming  call  request  is processed by the
   service.  The service analyses the rules and one of them fires (event 
   and condition matches).  For example the  action  associated  to  the
   rule  is  to  route  the  call to the Gianni's home phone but this is
   busy.  The service then evaluates the rule  associated  to  the  busy
   condition  and  for  example it forwards the call to voice-mailbox of
   the subscriber.  (S)He is then notified about this  voice message.  
   
5.2 Internet Call Waiting  
   
   Internet call waiting (ICW) is a service that enables a user  engaged


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   in a dial up Internet session to be alerted when an incoming call has 
   arrived.   After  the  Internet  user  has  been alerted, he is given
   several options for handling the call (e.g., forwarding it, sending a 
   waiting announce/tone, accepting the call over  PSTN  suspending  the
   Internet  session,  or  accepting  the call over IP keeping alive the
   Internet session).  In parallel the  caller  is  announced  that  the
   callee is  busy  and  (s)he  is asked to hold  the line.  In order to
   enable the service the subscriber has to use a client  software  that
   performs  the  registration  phase by storing the association between
   the PSTN line number  (i.e.  CLI)  and  the  IP  address  of  his/her
   Internet session.  
   Service features that have an impact on the PSTN-to-IP context are: 
   - Handling an incoming call 
   The  subscriber receives a PSTN call directed to his/her phone, which
   is busy, since it is engaged in a dial-up Internet session.   The  IN
   Service  Switching  Point  (SSP) triggers the service logic to handle
   this call event.  The caller is connected to a Intelligent Peripheral 
   in  order  to  send  a  message  to  inform  him/her  to  wait  (call
   queueing).   The  service  retrieves the IP address of the callee and
   then, depending on the user profile, notifies the  incoming  call  to
   the  Internet  user  with caller number or name or other call relater
   information.  The Internet user may choose different options: 
   
   1. Accept call on PC using voice over IP 
   2. Accept call on phone 
      3. Suspend IP session and answer the call on the phone 
   4. Reply the caller with a pre-registered message 
   5. Reject the call 
   
   In the following sections ICW service scenarios are described in more 
   details.  
   
5.2.1 Connection To Internet 
   
   This scenario describes what happens when  an  ICW  subscriber  (e.g.
   Bob) connects to Internet by means of a dial-up connection.  
   
   Preconditions: Bob has subscribed to ICW service 
   Post Conditions:  
   Trigger  Detection  Point (DP) 13 (in ETSI Core INAP terminology DP13
   corresponds to TCalledPartyBusy event) is armed so that when there is 
   an incoming call for Bob and his line is busy the  ICW  is  notified.
   The dial-up connection is established.  
   
   First  Bob launches the Internet connection software, which dials the
   ISP phone number, that is an number which represents an IN service.  
   
   As a result  of  the  pre-arranged  agreement  between  the  Internet
   service    provider    and    the    network    provider    the   DP3
   (AnalyzedInformation) was set.  
   


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   The SSP triggers the service logic by sending an InitialDP message to 
   the IN Service Control Point (SCP).  
   
   Then the SSP, through a gateway, notifies the ICW Service Logic about 
   a network event related  to  an  incoming  call  for  the  ISP  phone
   number.  The service logic is executed within a kind of SLEE (Service 
   Logic  Execution  Environment)  that  provide  API  to  interact with
   network functionality.  
   
   The ICW service logic consequently sets TDP13 using  some  management
   interface on  the SSP.  The ICW service logic then subscribes to call
   event related to Bob's call  leg  disconnection  since  it  needs  to
   disarm the TDP13, when the Internet connection is disconnected.  
   
   The  ICW  Service Logic gives instruction to the network to route the
   call to the  address  of  the  Network  Access  Server  (NAS)  to  be
   connected.  
   Furthermore  it  arms DP7 (OAnswer) in order to be notified about the
   result of the call routing.  
   
   The connection is set up between the SSP and the NAS and consequently 
   the network acknowledges the establishment of the connection.  
   
   Since the ICW Service Logic needs to  monitor  NAS  disconnection  as
   well  as  Bob's  disconnection,  it  subscribed  to  call  event (DP9
   ODisconnect) related to NAS disconnection  (this  subscription  could
   not  be  requested  before  the  successful  notification of the call
   routing to the NAS).  
   
   When the network acknowledges the  successful  establishment  of  the
   call  leg  towards the NAS, the call processing is stopped at the DP7
   (OAnswer) since it has been set as an EDP-R.  
   
   The ICW  service  logic  instructs  the  SCP  to  continue  the  call
   processing (this is mapped onto the INAP operation Continue).  
   
   The  subscriber  is now requested to authenticate through a login and
   password.  This subscriber data  are  sent  to  a  Radius  server  to
   authenticate  him  and to retrieve from a directory server Bob's user
   profile based on its home phone number.  
   
   The association between the IP address of Bob and  Bob's  home  phone
   number is stored in his user profile.  
   
   The   software   in   charge  of  handling  invitations  is  launched
   automatically after the connection to Internet  is  established  e.g.
   IETF SIP UAC.  
   
5.2.2 Incoming Call Notification 
   
   This scenario describes what happens when a user tries to call Bob at 


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   his home phone number.  
   Preconditions: Bob is connected to Internet.  
   Post  Conditions:    Bob  is notified of an incoming call and have to
   chose either to reject the call, to answer the call over a PSTN  line
   or to answer the call over IP.  
   
   An unnamed  user  dial  Bob's  phone  number.  Bob's SSP detects that
   Bob's phone line is busy and according  to  the  previous  arming  of
   TDP13,  an  InitialDP  message  is sent to the SCP in order to notify
   that a call has been triggered.  
   The call is routed to an  Interactive  Voice  Response  (IVR),  which
   implements IN-Special Resource Functions.  
   The  ICW  Service Logic instructs to send a ConnectToResource message
   to the SSP.  The SSP then establish the connection to the IVR.  
   The service logic sends the PlayAnnouncement  operation  to  the  IVR
   thus an announcement is played to the user.  
   In  the  meantime (while the announcement is played or even while the
   connection  to  the  IVR  is  established),  the  ICW  Service  Logic
   retrieves  Bob's  user  profile from the directory server in order to
   get Bob's PC IP address.  
   The ICW service logic sends an invitation (e.g. SIP  INVITE  message)
   to Bob with information about the caller (e.g. Calling Line Identity) 
   asking him either to: 
   - Accept call on PC 
   - Accept call on phone 
   - Suspend IP session and answer the call on the phone 
   - Reply the caller with a pre-registered message 
   - Reject the call 
   Bob's choice is sent back to the ICW Service Logic via a SIP response 
   message.  
   The IVR reports the end of the Announcement to the SSP.  
   
5.2.3 Rejecting The Call 
   
   This scenario describes what happens when Bob is invited to a call by 
   an unnamed user and chooses to reject it.  
   
   Preconditions: Bob is connected to Internet via a dial-up connection, 
   an  incoming  call  has  arrived  for  Bob, Bob chooses to reject the
   call.  Post Conditions: The incoming call is terminated.  The dial-up 
   connection is still active.  
   
   When the Internet user rejects the  incoming  call  the  ICW  service
   logic  requests  to  play  an announcement to the unnamed user saying
   that the callee has rejected the call.  
   The service logic asks to play an announcement.  
   When the announcement is finished the  SSP  notify  the  ICW  Service
   Logic about the end of it.  
   The ICW Service Logic then releases the call initiated by the unnamed 
   user.  
   


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5.2.4 Accepting The Call On The Phone 
   
   This scenario describes what happens when Bob is invited to a call by 
   an unnamed user and chooses to answer on his PSTN line.  
   Preconditions:  Bob  is  connected  to Internet by means of a dial-up
   connection, an incoming call has arrived for Bob, Bob chose to answer 
   the call on his PSTN line.  Post Conditions: The  dial-up  connection
   is terminated and the call is established between Bob and the unnamed 
   user using PSTN to PSTN connection.  
   
   The   ICW   client   software  disconnects  the  dial-up  connection.
   Therefore a disconnect signal is sent to the SSP. This  is  triggered
   by  the  SSP  since an EDP9 (ODisconnect) related to the call between
   Bob and the NAS was previously armed.  
   Consequently the SSP sends an EventReportBCSM operation  to  the  ICW
   Service Logic.  
   The  ICW  Service Logic disarms TDP13 on Bob's phone line by means of
   some management interface.  
   Since a connection to an IVR was still established, the  ICW  Service
   Logic releases this connection.  The SSP disconnect the connection to 
   the IVR and sends to the NAS a message to disconnect it.  
   As the result of the previous arming of an EDP9 (ODisconnect) related 
   to  the  disconnection  of  the  NAS the SSP sends an EventReportBCSM
   operation to the ICW Service Logic.  
   The NAS detects the end of the dial-up connection and  instructs  the
   Radius gateway  to  stop the accounting.  The Radius gateway asks the
   accounting server to stop the accounting for Bob's account.  
   Bob's user profile is updated by deleting the  IP  address  from  the
   scope of the previous dial-up connection.  
   The  ICW  Service  Logic  has  just  released  the  call  to the NAS,
   therefore it tries to route the call to Bob's phone line.  
   The call finally is routed to Bob's phone line.  
   
5.2.5 Accepting The Call On IP 
   
   This scenario shows what happens when Bob during his Internet session 
   chooses to answer an incoming call using VoIP.  
   Preconditions: Bob is connected to Internet,  an  incoming  call  has
   arrived for Bob, Bob chooses to answer the call over IP.  
   Post  Conditions: The call is established between Bob and the unnamed
   user using a VoIP gateway.  
   
   Since a connection to an IVR was still established, the  ICW  Service
   Logic release this connection.  
   The SSP disconnect the connection to the IVR.  
   The  ICW  Service  Logic retrieves Bob's IP address and instructs the
   network to route  the  call  to  the  appropriate  VoIP  gateway  and
   requests to be notified of the outcome of the call routing.  
   The SSP establish the PSTN connection to the VoIP gateway.  
   The ICW Service Logic controls the VoIP gateway to terminate the call 
   to Bob's  IP  address.  Depending on the VoIP gateway used, different


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   control interfaces are possible e.g.  H323, MeGaCo, ...  
   The VoIP gateway terminates the call to Bob's IP address.  
   
5.3 Meeting Scheduler  
   
   The service is a meeting scheduler and initiator.  A user has  a  web
   based user  interface to schedule a meeting.  He decides the time and
   the attendants.  This information is recorded in the service profile, 
   once it is decided it is authorised.  The timing information also  is
   used by an external application that launches the service.  
   The  service  retrieves the attendant's list from the service profile
   and uses the user  profile  to  locate  the  users.    Two  kinds  of
   communications  are  considered: VoIP and POTS meaning that users can
   be located both on a telephone and a PC. The service  establishes  as
   many  calls as needed and puts all of them in the same communication.
   Some of the information can reside within  the  user's  domain  (e.g.
   attendant  list),  being  accessed  from  the  service upon execution
   time.  
   Service features that have an impact on the PSTN-to-IP context are: 
   - Notification of phone attendance confirmation on meeting  manager's
   PC 
   During  the  meeting  confirmation phase the service logic checks for
   every meeting participant the availability of an  E-mail  address  in
   his/her user  profile.  If the participant has no e-mail address, the
   service logic chooses a telephone number and  a  time  at  which  the
   meeting  participant  might  be  available  by  elaborating  the User
   Profile data and uses this information to schedule, via  an  external
   application, an attendance confirmation via phone.  
   The  phone  attendance  confirmation  process  is activated by the an
   external scheduler.  Using the information related  to  a  particular
   meeting attendant,  a  new  call  is  created.    The  service  logic
   instructs the Call Control component to  interact  with  the  Service
   Node  to  create  a  new  Call  leg  and  to  route it to the meeting
   participant's phone number.  When (s)he answers, (s)he  is  requested
   to dial  his/her  PIN  to authenticate him/her.  After the successful
   authentication data check against his/her User Profile information, a 
   message is played informing him/her  of  the  scheduled  meeting  and
   asking her/him  to confirm his/her attendance.  Meeting participant's
   answer (dialled digit) is then collected and depending on it, her/his 
   attendance is confirmed in the meeting profile manager.  The  meeting
   manager is informed about this acceptance by means of a pop up window 
   or by updating a graphical application.  
   - Notification of meeting start on PC for VoIP users.  
   After  the  meeting  is  scheduled the service informs the attendants
   about it for example by Email. Each potential attendant is  requested
   to confirm his/her participation.  
   The service can have 2 different choice: 
   1. It could check the User profile to get the location of the user at 
   the time the meeting is scheduled.  
   2. It could ask the attendant to provide this information directly.  
   


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   In  the  first  case  an external scheduler to start a new conference
   triggers the meeting activation process.  It has previously  obtained
   a  list  of  attendants  from  the  service  profile  containing  the
   identification of those users who have confirmed their attendance  at
   the meeting.    From  the  User  Profile,  it  has obtained a list of
   possible locations for  each user, ordered by user preferences at the 
   time when the conference is  going  to  take  place.    Once  it  has
   obtained  all  the  needed  information related to the attendants, it
   starts the conference and notifies VoIP  attendants  of  the  meeting
   start  by  popping  up  a  window on their PC. They can at this point
   accept or deny to join the conference.  
   
   - Notification to the chair's PC of participants who  left  or  joint
   the conference.  
   During the execution of the service the conference status, in term of 
   participants and  time  they have been logging in, is monitored.  The
   chair  (who  scheduled  the  meeting)  may  be  not  logged  in   the
   conference.   (S)He  is  informed in any case about any change of the
   conference status.  For example if a participant leaves or joint  the
   conference  the  chair  is notified about this on his/her PC and this
   information are stored in  a  file  and  used  for  billing  off-line
   procedures.  
   
   - On-line notification which participant(s) is currently talking.  
   The  service  could  have  the  feature  to  update  a graphical user
   interface,  which  maintains  an  image  of  the  current  conference
   status.   So  that  participants  who  are also connected to Internet
   could receive graphical information  about  who  is  talking.    This
   service  feature  depends  on  the  capability of the resource, which
   provides the multiparty audio bridging feature  to  the  service,  to
   capture and use information about traffic flows for each leg.  
   
5.4 Unified Communication 
   
   The  Unified Communication Service allows users to send, retrieve and
   receive messages disregarding the format and the terminal  where  the
   user is  connected.  The user should be able to create and respond to
   multimedia messages from any terminal and create and send any type of 
   message without regard to the recipient's mailbox requirements.   The
   user  should  also  be able to reply to messages and forward messages
   with calls, and reply and forward calls with messages.  
   Service features that have an impact on the PSTN-to-IP context are: 
   - Interworking between different session e.g.    telephone  call  and
   chat session.  
   The  following  scenario  is  very  interesting  from  the  technical
   perspective since messages  are  exchanged  when  different  sessions
   (i.e. telephone call and chat session) are occurring.  
   In  this scenario two users (John and Mary) are involved in telephone
   call and two other  users  (Anna  and  Mike)  are  involved  in  Chat
   session.    Suddenly,  Mike  knows  that  John's  mother  is  at  the
   hospital.  So, he sends a  message  (i.e.  e-mail  message)  to  John


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   marked as  a High Priority message.  The provider notifies John about
   the message.  John holds the phone call, calls the messaging  service
   and   receives   the  message  (in  a  voice  format,  text-to-speech
   conversion is required).  Sends a reply to John in a voice  form  at,
   the  provider  converts  it  in  a text format and sends an E-mail to
   Mike.  
   Meanwhile John unholds the call to inform Mary he has to hang-up.  
   
5.5 Distributed and Enhanced Call Center 
   
   The Distributed and Enhanced Call Center scenario shows several sides 
   of IN-Internet integration.  It features the speech connection  of  a
   PSTN terminal to a VoIP terminal, where the service logic controlling 
   the call  resides  in  a  3rd party domain.  It also shows how a call
   between a VoIP terminal and a PSTN terminal could be set up initiated 
   from the Internet, through a 3rd party  service.    The  call  center
   agents are  provided  with  a  PC  with  a  Web interface.  The PC is
   connected  to  Internet,  and   it   uses   this   con   nection   to
   register/deregister  availability  of  handling  incoming calls (data
   communication), and to establish the communication  with  the  client
   using VoIP  (voice  communication).   This solution has the advantage
   that the call center can have the agent positions totally distributed 
   and the agent could even be located in his/her own home.   The  agent
   is  also  able  to place outgoing calls by the use of a click-to-dial
   like service to initiate a call (PC to phone).  The  logic  and  data
   for selecting the "best" availa ble agent is allocated in an external 
   domain seen  from  the  IN  point  of  view.   It could reside in the
   company administration domain or this service could be outsourced  to
   another company.    The logic and data for agent selection could also
   be allocated in the IN domain, but this case is not examined  further
   here.  
   
   Service features that have an impact on the PSTN-to-IP context are: 
   
   - Incoming call notification and advanced reply.  
   The  best  available  agent  is notified on his/her PC about incoming
   calls directed to him/her.  He is presented with a menu window, which 
   allows him to choose between the following choice: 
   
   1. Accept the call 
   2. Reject the call 
   3. Forward the call to another agent 
   4. Reply by typing a message that is translated into speech 
   5. Reply by sending back  a  dynamic  Web  page  built  by  composing
   information 
   
   
6. IN service request from the Internet.  
   
   It  should  be  possible to open the access to IN functions during an
   internet session.  It should enable: 


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   - PINT-like services: Click (Request) to Dial ,  Click  (Request)  to
   Fax and Voice access to web content.  
   - Access to customer and service information.  
   -  Access  to  IN  routing  services,  IP telephony GW using IN-based
   address resolution.  
   Services may required specific Intelligent Peripheral resources  with
   media transformation like text-to-fax and text-to-speech.  
   
   The  following services can well help in defining requirements in the
   context of how services supported by IN entities can be started  from
   Internet requests.  
   
6.1 Click-to-Dial (C2D) 
   
   A  user  is  able  to  initiate a telephone call by clicking a button
   during a web session.  
   The called address (as well as the caller address) is  either  an  IP
   address or  a  phone line number.  The charging party could be either
   the initiator or one of the called parties.  
   
   Service features that have an impact on the IP-to-PSTN context are: 
   - Starting a call by selecting an user(s) on a Web page 
   
6.1.1 Click2Dial invocation 
   
   The user invokes the Click2Dial service.  
   The service logic contacts Call Control to setup a call  to  setup  a
   call between the caller and the called user.  
   It  should  be  possible  for  a  user to initiate a call between two
   parties, neither of which is the user.  If the user invoking the call 
   is to be billed, this means that the C2D service  has  to  known  the
   initiator  of  the  call  so  that  the  information may be passed to
   billing system.  
   In the C2D service, especially if the call is placed (and paid  for!)
   by a third party, it is difficult to determine who is the originating 
   party and  who  is  the destination party.  This has an impact on the
   use of the Parlay interface insofar as some order must be  placed  on
   the  parties so that there is an origination and a destination to the
   call.  This document assumes that the originating address is taken as 
   the first address entered (User A) and the destination address is the 
   second address entered (User B ) 
   
   Preconditions 
   The user is logged on to the system and has selected the click-2-dial 
   service for invocation 
   Post Conditions 
   A  call  has  been  established  between  the  origination  and   the
   destination.  
   
   The  user  (this  is  the user to be billed for the service) requests


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   that a call be placed between two users: User A and User B.  
   The C2D service is requested to initiate a call between the indicated 
   parties.  
   The C2D first initiates the routing of the  call  towards  the  first
   party in the call (User A).  
   The  C2D service is notified that the routing of the call towards the
   first party has been successful.  
   Then it initiates the routing of the call towards the second party in 
   the call (User B) and it is notified that the  routing  of  the  call
   towards the called party has been successful.  
   The call should now be active.  
   
6.1.2 Call Setup 
   
   The  following sections describe the call setup from the viewpoint of
   the Call control.  
   Four scenarios have been  identified:  PSTN-PSTN,  PSTN-PC,  PC-PSTN,
   PC-PC.  
   
6.1.2.1 PSTN as originating party terminal 
   
   The  address of the origination is resolved from the User Profile. If
   the  originating  user  is  registered  on  a  PSTN  phone  then  the
   originating call leg is routed towards the PSTN.  
   Preconditions 
   The  calling user is logged on to the system and has selected/invoked
   the click-to-dial service.  
   Post Conditions 
   The call has been set up towards the origination address.  A call leg 
   has been established towards the origination address.  
   
   First the user A's UserProfile is contacted  to  return  the  correct
   terminal  to  which the call should be delivered: in this case a PSTN
   address.  
   An invocation to create a new  Call  is  sent  to  the  call  control
   manager which returs the call session Id.  
   The call manager creates a new call.  
   At  this  point  the  service logic initiates the routing of the call
   towards the first party in the call (User A) and waits the result  of
   this operation.  The result is sent to the C2D service logic.  
   
6.1.2.2 PSTN as destination party terminal (PSTN-PSTN) 
   
   The address of the destination is resolved from the User Profile. The 
   user  is  registered  on  a  PSTN.  The  call  is  routed towards the
   destination.  
   
   Preconditions 
   The call has been routed to the origination.  
   Post Conditions 
   The call has been set up towards the destination.   A  call  leg  has


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   been established   towards  the  destination.    The  call  setup  is
   complete.  
   
   The C2D service logic requests to user B's UserProfile to return  the
   correct terminal to which the call should be delivered.  
   The Address of the terminal to which the call should be delivered -in 
   this case a PSTN address - is returned.  
   The  C2D  service logic initiates the routing of the call towards the
   second party in the call (User B).  
   The result of this operation is sent to the C2D service logic.  
   
6.1.2.3 PC as destination party terminal (PSTN-PC) 
   
   The address of the destination is resolved from the User Profile. The 
   user  is  registered  on  a  PC.  The  call  is  routed  towards  the
   destination.  
   Preconditions 
   A call leg has been established to the origination.  
   Post Conditions 
   The call  has  been  set  up towards the destination.  A call leg has
   been established  towards  the  destination.    The  call  setup   is
   complete.  
   
   The C2D service logic requests to user B's User Profile to return the 
   correct terminal to which the call should be delivered.  
   The Address of the terminal to which the call should be delivered -in 
   this case an IP address- is returned.  
   The service logic then invites User B to the current session by using 
   the  User ID of the user and the address of the terminal at which the
   user is to be invited.  
   SIP may be used as the invitation mechanism.  
   User responses to the invitation.  
   The response to the invitation is sent to the service logic.  
   In case the user  accepted,  the  C2D  service  logic  initiates  the
   routing of the call towards the second party in the call (User B).  
   The result of this operation is sent to the C2D service logic.  
   
6.1.2.4 PC as originating party terminal 
   
   The  address of the origination is resolved from the User Profile. If
   the originating user is registered on a PC then a call leg is  routed
   towards the  Internet.  In  reality  the call is not routed yet.  But
   from the point of view of the service logic, the Call Control behaves 
   as if the call has been routed correctly.  
   Preconditions 
   The calling user is logged on to the system and has  selected/invoked
   the click-to-dial service.  
   Post Conditions 
   The call has been set up towards the origination 
   
   The  C2D service logic requests to user A's UserProfile to return the


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   correct terminal to which the call should be delivered.  
   The Address of the terminal to which the call should be  delivered  -
   in this case an IP address - is returned.  
   The service logic then invites User A to establish a new session.  
   SIP may be used as the invitation mechanism.  
   User responses to the invitation.  
   The response to the invitation is sent to the service logic.  
   In  case  user A accepts the service logic instructs the call manager
   to create a new call.  
   The call manager creates a new call.  
   The C2D service logic initiates the routing of the call  towards  the
   first party in the call (User A).  
   The result of this operation is sent to the C2D service logic.  
   
6.1.2.5 PSTN as destination party terminal (PC-PSTN) 
   
   The address of the destination is resolved from the User Profile. The 
   user  is  registered  on  a  PSTN.  The  call  is  routed towards the
   destination.  
   Preconditions 
   The call has been routed to the origination.  
   Post Conditions 
   The call has been set up towards the destination.   A  call  leg  has
   been established   towards  the  destination.    The  call  setup  is
   complete.  
   
   The C2D service logic requests to user B's UserProfile to return  the
   correct terminal to which the call should be delivered.  
   The Address of the terminal to which the call should be delivered -in 
   this case a PSTN address-is returned.  
   The  C2D  service logic initiates the routing of the call towards the
   second party in the call (User B).  
   The result of this operation is sent to the C2D service logic.  
   The call setup is complete.  
   
6.1.2.6 PC as destination party terminal (PC-PC) 
   
   The address of the destination is resolved from the User Profile. The 
   user  is  registered  on  a  PC.  The  call  is  routed  towards  the
   destination.  
   Preconditions 
   A call leg has been established to the origination.  
   Post Conditions 
   The call  has  been  set  up towards the destination.  A call leg has
   been established  towards  the  destination.    The  call  setup   is
   complete.  
   
   The C2D service logic requests to user B's User Profile to return the 
   correct terminal to which the call should be delivered.  
   The Address of the terminal to which the call should be delivered -in 
   this case an IP address - is returned.  


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   The service logic then invites User B to the current session by using 
   the  User ID of the user and the address of the terminal at which the
   user is to be invited.  
   SIP may be used as the invitation mechanism.  
   User responses to the invitation.  
   The response to the invitation is sent to the service logic.  
   In case the user  accepted,  the  C2D  service  logic  initiates  the
   routing of the call towards the second party in the call (User B).  
   The result of this operation is sent to the C2D service logic.  
   The call setup is complete.  
   
6.2 Meeting Scheduler (MS) 
   
   This  section  provides  a  description  for  a meeting scheduler and
   initiator service.  In it, an user has a web based user interface  to
   schedule a  meeting.    He  will  decide the time and the attendants.
   This information is recorded for the  service  profile,  once  it  is
   decided it  is authorised.  The timing information is also used by an
   external  application  that  launches  the  service  via  the  access
   function.   Every  meeting  attendant  is  confirmed,  either with an
   e-mail notification and a following confirmation process  via  a  web
   page or with a phone call.  Once the service is launched, it uses the 
   attendants list  and  the  service  profile to locate the users.  Two
   kind of communications are considered: VoIP  and  POTS.  The  service
   sets up the communication to the end users using a service node.  The 
   service  node  has  a direct connection to the PSTN and uses the call
   control component, controlling the vocal  gateway  to  establish  the
   connection to  the  VoIP  terminals.  The service establishes as many
   calls as needed and puts all of them in the same co mmunication.  
   Service features that have an impact on the IP-to-PSTN context are: 
   - Starting a meeting by selecting a  user(s)  and  timing  on  a  Web
   page.  
   
6.2.1 Meeting creation 
   
   The  Meeting manager accesses via a Web Browser the provider Web page
   containing the MS applet.  (S)He sets relevant  information  for  the
   meeting to  take place, e.g.  meeting attendants and date/time of the
   meeting - along  with  some  authentication  information  to  confirm
   meeting managerĘs identity.  
   Preconditions 
   The  Meeting  manager must have a Web access to log in the system and
   to set the MS service.  
   Post Conditions 
   Meeting confirmation process is started.  
   
   The  Web  Server  launches  the  Meeting  Scheduler  Servlet,   which
   interacts  with  the Service Logic. The MS Servlet checks in order to
   confirm the meeting manager  authentication  data.    Furthermore  it
   checks  if  the  meeting  participants identifier do really exist and
   whether the terminals specified in the  User  Profile  configurations


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   meet the  service  requirements.    The MS Servlet then, accesses the
   Meeting Profile Manager  to  include  the  new  meeting  information.
   Meeting  Profile  Manager returns a meeting identifier whi ch have be
   used by the meeting participants to confirm their attendance  to  the
   meeting.   This  identifier is also shown to the meeting manager in a
   HTML page confirming the success of the operation.  
   
6.2.2 Meeting confirmation 
   
   Preconditions 
   A  meeting  must  have  been  created  following  the  meeting  start
   process.  
   Post Conditions 
   Participants are informed about the meeting by Email or by phone.  
   
   During  the  meeting  confirmation phase the service logic checks for
   every meeting participant the availability of an  E-mail  address  in
   his/her user  profile.    If an E-mail address exists, the MS service
   logic sends an E-mail to the meeting participants  informing  her/him
   of the  meeting.    If the meeting participant has no e-mail address,
   the service logic chooses a telephone number and a time at which  the
   meeting  participant  might  be  available  by  elaborating  the User
   Profile data and uses this information to sch edule, via an  external
   application, an attendance confirmation via phone.  
   The  sequence  is  repeated  for  every  meeting  participant  in the
   participant list of the meeting.  
   
6.2.3 Web attendance confirmation 
   
   Preconditions 
   The participant received an e-mail including the  meeting  attendance
   confirmation  URL, meeting identifier and related meeting information
   like the participant list.  
   
   Post Conditions 
   The participant confirms his attendance.  
   
   The participant accesses via a Web browser the  Web  page  containing
   the meeting   attendance  confirmation  form.    The  participant  is
   authenticated by means of user name and password  which  are  checked
   against personal data in his/her User Profile. Then (s)he selects the 
   confirmation  choice  and  finally (s)he is shown with a HTML page to
   notify the success of the operation.  
   
6.2.4 Conference activation 
   
   Preconditions 
   The meeting activation process has  obtained  the  list  of  possible
   points of presence for of the participants.  
   Post Conditions 
   The conference is activated.  


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   The  meeting  activation  process  is  activated  by a time-dependent
   scheduler to start a new conference.  It has  previously  obtained  a
   list  of  attendants  from the meeting profile manager containing the
   information related to those users who  confirmed  their  attendance.
   From  the User Profile, it gets a list of possible locations for each
   user, ordered by user preferences at the time when the conference  is
   going to take place.  
   Once  it  has  obtained  all  the  needed  information related to the
   attendants, the service logic creates a  new  call  instance  in  the
   Service Node through interacting with the Call Control.  
   
6.2.5 Call legs set up 
   
   Preconditions 
   A new call has been created but no call leg is attached yet.  
   
   Post Conditions 
   The participants' call legs are created 
   
   Once the conference has been started and a call has been created, the 
   meeting activation process needs to create a call leg for each of the 
   attendants.  
   The  first  point  of presence of the first attendant is selected and
   the Call Control is asked to route the call leg to that address.  
   If the address is a telephone number, the Call Control just instructs 
   the Service Node to route the call leg to it.  If the address  is  an
   IP  address,  the  call leg has to be routed through a vocal gateway.
   Consequently the Call Control instructs the Service Node to route the 
   call leg to one of the phone numbers assigned to  the  vocal  gateway
   and  instructs the gatekeeper to translate that number to the desired
   IP address.  When the call reaches the vocal gateway it requires from 
   the gatekeeper, using RAS proto col, the target IP address.   If  the
   end  user  doesnĘt  answer  the  call,  the next point of presence is
   selected and the call leg is  routed  to  this  new  address.    This
   procedure  is  repeated  until  the  end user answers or all possible
   addresses are  tried.    When  an  user  answers,   music   or   some
   announcement  is  played to him/her until the process to call all the
   attendants is finished.  At that point the conference is ready to  be
   set up.  
   
6.2.6 Conference start 
   
   Preconditions 
   A  call has been created and all the related call legs have also been
   created, but they are not attached yet.  
   Post Conditions 
   The conference has been completely set up and the attendants can talk 
   to each other.  
   
   When all attendants have been contacted and they have answered (or at 


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   least all their different possible addresses  have  been  tried)  the
   conference can be started.  As a first step the music/announcement is 
   stopped.   Then the Service Node is instructed to create a conference
   and to attach the call legs to it.  
   

   7. Security considerations 
   
   
   P909 recognises that security considerations are quite essential  for
   successful deployment  of  the  described  services.   P909 addresses
   security considerations for the example services by focusing  on  the
   following items: 
   
   o  Peer  entity  authentication  is  required to allow an end-user to
   prove its identity to the provider domain in the network.   This  can
   be  accomplished via a Web-based access by means of the user name and
   password or personal number and PIN.  
   o In addition, a secure communications could be used in case personal 
   data  are  transmitted  for  authentication  and  user  prefererences
   elaboration purposes e.g.  by means of the SSL technology.  
   o  End-user profile verification to validate the authorisation of the
   end user to use a service.  
   o Prevention of uncontrolled disclosure of  information  without  the
   permission of its owner.  
   
   Since  the  security  issues  are  still work in progress in P909,the
   material of this  section  is  of  preliminary  nature  and  must  be
   revisited in a forthcoming paper.  
   
8. References 
   
   [1] J. Postel, RFC 1543, "Instruction to RFC Authors". October 1993 
   
   [2] S.Bradner, "The Internet Standards Process", RFC2026  
   
   [3]  M.  Barry  et  al., "What an IN system should be", Deliverable 1
   Eurescom P909, October 1999 
   
   [4] V. Blavette et al., "Architecture and  Scenario  for  Internet-IN
   Convergence", Deliverable 2 EURESCOM P909, February 2000 
   
   [5] I.Faynberg et al., "On a pre-SPIRITS Implementation in the Lucent 
   Technologies          Online          Communications         Center",
   draft-ietf-spirits-lucentocc-00.txt, February 2000 
   
   [6] I.Faynberg  et  al.,  "Toward  Definition  of  the  Protocol  for
   PSTN-initiated  Services  Supported  by  PSTN/Internet Interworking",
   draft-ietf-spirits-protocol-00.txt, March 2000 
   
   [7] L. Slutsman, "A  proposal  for  new  PINT  building  blocks  with


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   applications  to  Conference  Calling",  draft-ietf-pint-conf-01.txt,
   Expires: July 2000 
   
   
9. Authors' Addresses 
   
   Valerie Blavette 
   E-mail: valerie.blavette@cnet.francetelecom.fr 
   Telephone: +33 2 96053221 
   Fax: +33 2 96053784 
   CNET/France Telecom 
   Technopole Anticipa 
   2 avenue Pierre Marzin 
   22307 Lannion Cedex (France) 
   
   Gianni Canal 
   E-mail: canal@cselt.it 
   Telephone: +39 011 2285630 
   Fax: +39 011 2285069 
   CSELT/Telecom Italia, 
   via Reiss Romoli, 274, 
   I-10148 Torino (Italy) 
   
   Uwe Herzog 
   E-mail: Uwe.Herzog@telekom.de 
   Telephone: +49 6151 837880 
   Fax: +49 6151 834590 
   T-Nova Deutsche Telekom Innovationsgesellschaft mbH 
   Technologiezentrum Darmstadt D-64307 Darmstadt (Germany) 
   
   Carlo Alberto Licciardi 
   E-mail: carlo.licciardi@cselt.it 
   Telephone: +39 011 2285705 
   Fax: +39 011 2285069 
   CSELT/Telecom Italia, 
   via Reiss Romoli, 274, 
   I-10148 Torino (Italy) 
   
   Stephane Tuffin 
   E-mail: stephane.tuffin@cnet.francetelecom.fr 
   Telephone: +33 2 96053217 
   Fax: +33 2 96053784 
   CNET/France Telecom 
   Technopole Anticipa 
   2 avenue Pierre Marzin 
   22307 Lannion Cedex (France) 

   10. Acknowledgments 
   
   The authors would  like  to  acknowledge  Roberto  Minerva  and  Alec
   Brusilovsky   for   their  contributions  to  the  creation  of  this


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   document.  
   
11. Glossary 
   
   API			Application Programming Interface
   C2D                  Click-To-Dial (Request to Call)   CALL
   CLI			Calling Line Identity
   CORBA			Common Object Request Broker Architecture
   CTI			Computer Telephony Interface
   DP                   Detection Point
   ECA			Event Condition Action
   EDP-R			Event Detection Point-Request
   ETSI			European Telecomminications Standards 
                        Institute
   EURESCOM			European Institute for Research and Strategic 
                        Studies in  Telecommunications
   GSM			Global System for Mobile communication
   GW				GateWay
   HTML			HyperText Markup Language
   ICW                  Internet Call Waiting
   IETF			Internet Engineering Task Force
   IN                   Intelligent Network
   INAP			Intelligent Network Application Protocol
   IP				Internet Protocol
   IN-IP			IN-Intelligent Peripheral
   ISP			Internet Service Provider
   ITU			International Telecommunication Union
   IVR			Interactive Voice Response
   MGCP                 Media Gateway Control Protocol
   MS				Meeting Scheduler
   MCU			Multipoint Conference Unit
   NAS			Network Access Server
   PC				Personal Computer
   PDA			Personal Digital Assistant
   PIC                  Point-In-Call
   PIN			Personal Identification Number
   POTS			Plain Old Telephone Service
   PSTN                 Public Switched Telephone Network
   RFC			Request For Comments
   SCP                  Service Control Point
   SIP			Session Initiation Protocol
   SLEE			Service Logic Execution Environment
   SMS			Service Management System
   SN                   Service Node
   SSL			Secure Socket Layer
   SSP			Service Switching Point
   TDP			Trigger Detection point
   TINA			Telecommunication Information Network 
                        Architecture
   TLC                  Telecommunication
   URL			Uniform Resource Locator


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   VP				Virtual Presence
   VoIP                 Voice over IP (Internet Protocol)
   VPN			Virtual Private Network
   WG				Working Group
















































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