ISO/IEC 23915:2005

Information technology — Telecommunications and information exchange between systems — Corporate Telecommunication Networks — Signalling Interworking between QSIG and SIP — Call Diversion

ISO/IEC 23915:2005 specifies signalling interworking between "QSIG" and the Session Initiation Protocol (SIP) in support of call diversion within corporate telecommunication networks (CN), also known as enterprise networks. "QSIG" is a signalling protocol that operates between Private Integrated services Network eXchanges (PINX) within a Private Integrated Services Network (PISN). A PISN provides circuit-switched basic services and supplementary services to its users. SIP is an application layer protocol for establishing, terminating and modifying multimedia sessions. It is typically carried over IP. ISO/IEC 23915:2005 specifies signalling interworking for call diversion during the establishment of calls between a PISN employing QSIG and a corporate IP network employing SIP. It covers both the impact on SIP of call diversion in the QSIG network and the impact on QSIG of request retargeting in the SIP network. ISO/IEC 23915:2005 is applicable to any interworking unit that can act as a gateway between a PISN employing QSIG and a corporate IP network employing SIP.

Technologies de l'information — Télécommunications et échange d'information entre systèmes — Réseaux de télécommunications d'entreprise — Interaction de signalisation entre QSIG et SIP — Déviation d'appel

General Information

• Status: Published
• Publication Date: 02-Nov-2005
• ICS: 33.040.35 - Telephone networks
• Technical Committee: ISO/IEC JTC 1/SC 6 - Telecommunications and information exchange between systems
• Drafting Committee: ISO/IEC JTC 1/SC 6 - Telecommunications and information exchange between systems
• Current Stage: 9060 - Close of review
• Start Date: 02-Sep-2027

Standards Content (Sample)

INTERNATIONAL ISO/IEC
STANDARD 23915
First edition
2005-11-01


Information technology —
Telecommunications and information
exchange between systems — Corporate
Telecommunication Networks —
Signalling Interworking between QSIG
and SIP — Call Diversion
Technologies de l'information — Télécommunications et échange
d'information entre systèmes — Réseaux de télécommunications
d'entreprise — Interaction de signalisation entre QSIG et SIP —
Déviation d'appel




Reference number
ISO/IEC 23915:2005(E)
©
ISO/IEC 2005

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ISO/IEC 23915:2005(E)
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©  ISO/IEC 2005
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ISO/IEC 23915:2005(E)
Contents Page
Foreword .v
Introduction.vi
1 Scope.1
2 Normative references.1
3 Terms and definitions .2
3.1 External definitions .2
3.2 Other definitions.3
3.2.1 Call diversion .3
3.2.2 Call forwarding busy (CFB).3
3.2.3 Call forwarding no reply (CFNR).3
3.2.4 Call forwarding unconditional (CFU).3
3.2.5 Corporate telecommunication Network (CN).3
3.2.6 Entity A.3
3.2.7 Entity B.3
3.2.8 Entity C.3
3.2.9 Gateway.3
3.2.10 IP network.3
3.2.11 Leg A.3
3.2.12 Leg B.3
3.2.13 Leg C.4
3.2.14 Private Integrated Services Network (PISN) .4
3.2.15 Private Integrated services Network eXchange (PINX) .4
3.2.16 Rerouting entity.4
3.2.17 User A.4
3.2.18 User B.4
3.2.19 User C.4
4 Abbreviations and acronyms .4
5 Background and architecture for SIP-QSIG interworking.5
6 Call diversion .5
7 Call diversion in QSIG.6
8 Call diversion in SIP.7
9 Diversion interworking.7
9.1 Scenarios for diversion interworking.7
9.2 Mapping of numbers, names and URIs.8
9.3 Derivation of QSIG diversion reasons.8
9.3.1 Scenario A1.9
9.3.2 Scenario B1.9
9.3.3 Scenario C2.9
9.4 Derivation of SIP response codes (scenarios A2 and C1) .9
9.5 Mapping the QSIG diversion counter.10
9.6 Privacy considerations.10
9.7 Interworking for scenario A1.10
9.7.1 Transmitting a SIP INVITE request.10
9.7.2 Receipt of a SIP 1xx or 2xx response.11
9.7.3 Receipt of a SIP 4xx, 5xx or 6xx response.11
9.8 Interworking for scenario A2.11
9.8.1 Receipt of a SIP INVITE request.12
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ISO/IEC 23915:2005(E)
9.8.2 Receipt of a QSIG divertingLegInformation1 invoke APDU . 12
9.8.3 Receipt of a QSIG divertingLegInformation3 invoke APDU . 12
9.8.4 Transmitting a SIP response in which History-Info is allowed . 12
9.9 Interworking for scenario B1. 13
9.9.1 Receipt of a SIP 3xx response. 13
9.9.2 Receipt of a QSIG DISCONNECT or FACILITY message containing a callRerouteing return
result APDU. 14
9.9.3 Receipt of a QSIG FACILITY message containing a callRerouteing return error APDU. 14
9.9.4 Receipt of a QSIG FACILITY message containing a cfnrDivertedLegFailed invoke APDU . 14
9.10 Interworking for scenario B2. 15
9.10.1 Receipt of a QSIG FACILITY message containing a CallRerouteing invoke APDU. 15
9.11 Interworking for scenario C1. 15
9.11.1 Receipt of a QSIG SETUP message containing a divertingLegInformation2 invoke APDU . 15
9.11.2 Transmitting a QSIG CONNECT message. 16
9.12 Interworking for scenario C2. 16
9.12.1 Transmitting a QSIG SETUP message . 16
9.12.2 Receipt of a QSIG message containing a divertingLegInformation3 invoke APDU . 17
9.12.3 Sending History-Info in a response . 17
10 Example message sequences. 17
10.1 Scenario A1. 18
10.1.1 Successful call – history information in 200 response. 18
10.1.2 Successful call – history information in provisional response . 19
10.1.3 Failed call. 20
10.2 Scenario A2. 21
10.2.1 Successful call – CFU or CFB . 21
10.2.2 Successful call – CFNR. 22
10.3 Scenario B1. 23
10.3.1 Successful diversion – CFU or CFB . 23
10.3.2 Successful diversion – CFNR. 24
10.3.3 Failure – callRerouting.err received. 25
10.3.4 Failure – No answer following CFNR. 26
10.4 Scenario B2. 27
10.5 Scenario C1. 28
10.6 Scenario C2. 29
10.7 Scenario A1 followed by B1. 30
10.8 Scenario A2 followed by scenario B2. 31
10.9 Scenario C1 followed by scenario A1. 32
10.10 Scenario C2 followed by scenario A2. 33
10.11 Scenario C1 followed by scenario B1. 34
10.12 Scenario C2 followed by scenario B2. 35
11 Security considerations . 35
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ISO/IEC 23915:2005(E)

Foreword
ISO (the International Organization for Standardization) and IEC (the International Electrotechnical
Commission) form the specialized system for worldwide standardization. National bodies that are members of
ISO or IEC participate in the development of International Standards through technical committees
established by the respective organization to deal with particular fields of technical activity. ISO and IEC
technical committees collaborate in fields of mutual interest. Other international organizations, governmental
and non-governmental, in liaison with ISO and IEC, also take part in the work. In the field of information
technology, ISO and IEC have established a joint technical committee, ISO/IEC JTC 1.
International Standards are drafted in accordance with the rules given in the ISO/IEC Directives, Part 2.
The main task of the joint technical committee is to prepare International Standards. Draft International
Standards adopted by the joint technical committee are circulated to national bodies for voting. Publication as
an International Standard requires approval by at least 75 % of the national bodies casting a vote.
Attention is drawn to the possibility that some of the elements of this document may be the subject of patent
rights. ISO and IEC shall not be held responsible for identifying any or all such patent rights.
ISO/IEC 23915 was prepared by Ecma International (as ECMA-360) and was adopted, under a special “fast-
track procedure”, by Joint Technical Committee ISO/IEC JTC 1, Information technology, in parallel with its
approval by national bodies of ISO and IEC.
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ISO/IEC 23915:2005(E)


Introduction
This International Standard is one of a series of Standards defining the interworking of services and
signalling protocols deployed in corporate telecommunication networks (CNs) (also known as
enterprise networks). The series uses telecommunication concepts as developed by ITU-T and
conforms to the framework of International Standards on Open Systems Interconnection as defined by
ISO/IEC.
This International Standard specifies interworking between the Session Initiation Protocol (SIP) and
QSIG within corporate telecommunication networks (also known as enterprise networks) for calls that
undergo diversion. SIP is an Internet application-layer control (signalling) protocol for creating,
modifying, and terminating sessions with one or more participants. These sessions include, in
particular, telephone calls. QSIG is a signalling protocol for creating, modifying and terminating circuit-
switched calls, in particular telephone calls, within Private Integrated Services Networks (PISNs). QSIG
is specified in a number of Standards and published also as ISO/IEC International Standards.
This International Standard is based upon the practical experience of member companies and the
results of their active and continuous participation in the work of ISO/IEC JTC1, ITU-T, IETF, ETSI and
other international and national standardization bodies. It represents a pragmatic and widely based
consensus.
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INTERNATIONAL STANDARD ISO/IEC 23915:2005(E)

Information technology — Telecommunications and information
exchange between systems — Corporate Telecommunication
Networks — Signalling Interworking between QSIG and SIP —
Call Diversion
1 Scope
This document specifies signalling interworking between "QSIG" and the Session Initiation Protocol (SIP) in

support of call diversion within corporate telecommunication networks (CN), also known as enterprise
networks.
"QSIG" is a signalling protocol that operates between Private Integrated services Network eXchanges (PINX)
within a Private Integrated Services Network (PISN). A PISN provides circuit-switched basic services and
supplementary services to its users. QSIG is specified in Standards, in particular [1] (call control in support of
basic services), [2] (generic functional protocol for the support of supplementary services) and a number of
Standards specifying individual supplementary services. Diversion services are specified in [4] and the QSIG
signalling protocol in support of these services is specified in [5]. In particular, this signalling protocol signals
information about call diversion to the users involved.
SIP is an application layer protocol for establishing, terminating and modifying multimedia sessions. It is
typically carried over IP [8], [10]. Telephone calls are considered as a type of multimedia session where just
audio is exchanged. SIP is defined in [11]. An extension to SIP provides history information [14] that can be
used to signal information about the retargeting of a request, in particular a call establishment request, as it is
routed through a network.
This document specifies signalling interworking for call diversion during the establishment of calls between a
PISN employing QSIG and a corporate IP network employing SIP. It covers both the impact on SIP of call
diversion in the QSIG network and the impact on QSIG of request retargeting in the SIP network. Signalling
interworking for call diversion operates on top of signalling interworking for basic calls, which is specified in [6].
Call diversion interworking between a PISN employing QSIG and a public IP network employing SIP is outside
the scope of this specification. However, the functionality specified in this specification is in principle
applicable to such a scenario when deployed in conjunction with other relevant functionality (e.g., number
translation, security functions, etc.).
This specification is applicable to any interworking unit that can act as a gateway between a PISN employing
QSIG and a corporate IP network employing SIP.
2 Normative references
The following referenced documents are indispensable for the application of this document. For dated
references, only the edition cited applies. For undated references, the latest edition of the referenced
document (including any amendments) applies.
[1] International Standard ISO/IEC 11572 "Information technology — Telecommunications and information
exchange between systems — Private Integrated Services Network — Circuit mode bearer services — Inter-
exchange signalling procedures and protocol" (also published by Ecma as Standard ECMA-143).
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ISO/IEC 23915:2005(E)
[2] International Standard ISO/IEC 11582 "Information technology -- Telecommunications and information
exchange between systems -- Private Integrated Services Network -- Generic functional protocol for the
support of supplementary services -- Inter-exchange signalling procedures and protocol" (also published by
Ecma as Standard ECMA-165).
[3] International Standard ISO/IEC 13868 "Information technology -- Telecommunications and information
exchange between systems -- Private Integrated Services Network -- Inter-exchange signalling protocol --
Name identification supplementary services" (also published by Ecma as Standard ECMA-164).
[4] International Standard ISO/IEC 13872 "Information technology -- Telecommunications and information
exchange between systems -- Private Integrated Services Network -- Specification, functional model and
information flows -- Call Diversion supplementary services" (also published by Ecma as Standard ECMA-173).
[5] International Standard ISO/IEC 13873 "Information technology -- Telecommunications and information
exchange between systems -- Private Integrated Services Network -- Inter-exchange signalling protocol -- Call
Diversion supplementary services" (also published by Ecma as Standard ECMA-174).
[6] International Standard ISO/IEC 17343 "Information technology -- Telecommunications and information
exchange between systems -- Corporate telecommunication networks -- Signalling interworking between
QSIG and SIP -- Basic services" (also published by Ecma as Standard ECMA-339).
[7] Ecma Technical Report TR/86, "Corporate Telecommunication Networks – User Identification in a
SIP/QSIG Environment".
[8] J. Postel, "Internet Protocol", RFC 791.
[9] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119.
[10] S. Deering, R. Hinden, "Internet Protocol, Version 6 (IPv6)", RFC 2460.
[11] J. Rosenberg, H. Schulzrinne, et al., "SIP: Session initiation protocol", RFC 3261.
[12] J. Peterson, "A Privacy Mechanism for the Session Initiation Protocol (SIP)", RFC 3323.
[13] H. Schulzrinne, D. Oran, G. Camarillo, "The Reason Header field for the Session Initiation Protocol (SIP)",
RFC 3326.
[14] M. Barnes "An Extension to the Session Initiation Protocol for Request History Information", draft-ietf-
sipping-history-info-03 (work in progress).
3 Terms and definitions
In this document, the key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD",
"SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" are to be interpreted as described in
RFC 2119 [9] and indicate requirement levels for compliant implementations.
For the purposes of this specification, the following definitions apply.
3.1  External definitions
The definitions in [1] and [11] apply as appropriate.
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ISO/IEC 23915:2005(E)
3.2  Other definitions
3.2.1
Call diversion
the act of retargeting a call during call establishment by changing the user identity that is used as the basis for
routing to the destination.
3.2.2
Call forwarding busy (CFB)
call diversion invoked because the targeted user is busy.
3.2.3
Call forwarding no reply (CFNR)
call diversion invoked because the targeted user fails to reply within a certain time.
3.2.4
Call forwarding unconditional (CFU)
call diversion invoked for reasons other than those leading to CFB or CFNR.
3.2.5
Corporate telecommunication Network (CN)
sets of privately-owned or carrier-provided equipment that are located at geographically dispersed locations
and are interconnected to provide telecommunication services to a defined group of users.
NOTE 1 A CN can comprise a PISN, a private IP network (intranet) or a combination of the two.
NOTE 2 Also known as enterprise network.
3.2.6
Entity A
the entity that provides information about diversion to user A.
3.2.7
Entity B
the entity that invokes diversion for a call targeted at user B.
3.2.8
Entity C
the entity that provides information about diversion to user C.
3.2.9
Gateway
an entity that performs interworking between a PISN using QSIG and an IP network using SIP.
3.2.10
IP network
a network, unless otherwise stated a corporate network, offering connectionless packet-mode services based
on the Internet Protocol (IP) as the network layer protocol.
3.2.11
Leg A
the call segment between entity A and the rerouting entity for a call that undergoes diversion.
3.2.12
Leg B
the call segment between the rerouting entity and entity B for a call that undergoes diversion.
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ISO/IEC 23915:2005(E)
3.2.13
Leg C
the call segment between the rerouting entity and entity C for a call that undergoes diversion.
3.2.14
Private Integrated Services Network (PISN)
a CN or part of a CN that employs circuit-switched technology.
3.2.15
Private Integrated services Network eXchange (PINX)
a PISN nodal entity comprising switching and call handling functions and supporting QSIG signalling in
accordance with [1].
3.2.16
Rerouting entity
the entity that performs call rerouting on request from entity B and that provides information about diversion to
entity A and entity C.
3.2.17
User A
the calling user of a call that undergoes diversion.
3.2.18
User B
the user on behalf of which call diversion is invoked for an incoming call to that user.
3.2.19
User C
the user to which a call is diverted.
4 Abbreviations and acronyms
APDU Application Protocol Data Unit
CFB Call forwarding busy
CFNR Call forwarding no reply
CFU Call forwarding unconditional
IP Internet Protocol
PINX Private Integrated services Network eXchange
PISN Private Integrated Services Network
SIP Session Initiation Protocol
UA User Agent
UAC User Agent Client
UAS User Agent Server
URI Universal Resource Identifier
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ISO/IEC 23915:2005(E)
5 Background and architecture for SIP-QSIG interworking
The background and architecture of [6] applies. In addition, the interworking function in the protocol model
handles interworking for call diversion services. This involves interworking between the QSIG call diversion
protocol specified in [5] and SIP, including the use of SIP request history information as specified in [14].
6 Call diversion
Call diversion, as specified in QSIG and for the purposes of this document, is the act of retargeting a call
during call establishment by changing the user identity that is used as the basis for routing to the destination.
This can be viewed as being a change of destination user, although in some cases two identities can belong
to the same user, e.g., a home number and office number. The three users involved are known as user A (the
calling user A), user B (the called user or diverting user) and user C (the diverted-to user).
Reasons for invoking diversion are various and can depend on factors such as the state of the line serving
user B, the time of day and the type or identity of user A. It could also be as a result of action by user B in
response to the arrival of a call (sometimes known as call deflection). A diversion can occur immediately, i.e.
without alerting user B, or after a period of alerting without reply. With the exception of call deflection,
diversion requirements must be pre-configured into some equipment acting on behalf of user B, e.g., a
telephone, a PINX or a SIP proxy. This could be achieved, for example, by rules-based scripting.
It is often useful or even important that the users involved in a diverted call (user A and user C) are informed
of the diversion. This can be particularly important for automata, e.g., for a call diverted to a voice mail system
it might be important to indicate to the system that the call has been diverted from user B. However, privacy
considerations can sometimes lead to the suppression of this information.
The general model for a call that undergoes diversion is shown in Figure 1. Entity B is the entity that invokes
diversion, based on configuration or, in the case of call deflection, on request fro
...