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Network Working Group J. Elwell
Request for Comments: 4497 Siemens
BCP: 117 F. Derks
Category: Best Current Practice NEC Philips
P. Mourot
O. Rousseau
Alcatel
May 2006
Interworking between the Session Initiation Protocol (SIP) and QSIG
Status of This Memo
This document specifies an Internet Best Current Practices for the
Internet Community, and requests discussion and suggestions for
improvements. Distribution of this memo is unlimited.
Copyright Notice
Copyright (C) The Internet Society (2006).
Abstract
This document specifies interworking between the Session Initiation
Protocol (SIP) and QSIG within corporate telecommunication networks
(also known as enterprise networks). 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 Ecma Standards and published also as ISO/IEC standards.
Elwell, et al. Best Current Practice [Page 1]
RFC 4497 Interworking between SIP and QSIG May 2006
Table of Contents
1. Introduction ....................................................4
2. Terminology .....................................................5
3. Definitions .....................................................5
3.1. External Definitions .......................................5
3.2. Other definitions ..........................................5
3.2.1. Corporate Telecommunication Network (CN) ............5
3.2.2. Gateway .............................................6
3.2.3. IP Network ..........................................6
3.2.4. Media Stream ........................................6
3.2.5. Private Integrated Services Network (PISN) ..........6
3.2.6. Private Integrated Services Network Exchange
(PINX) ..............................................6
4. Acronyms ........................................................6
5. Background and Architecture .....................................7
6. Overview .......................................................10
7. General Requirements ...........................................11
8. Message Mapping Requirements ...................................12
8.1. Message Validation and Handling of Protocol Errors ........12
8.2. Call Establishment from QSIG to SIP .......................14
8.2.1. Call Establishment from QSIG to SIP Using
En Bloc Procedures .................................14
8.2.2. Call Establishment from QSIG to SIP Using
Overlap Procedures .................................16
8.3. Call Establishment from SIP to QSIG .......................20
8.3.1. Receipt of SIP INVITE Request for a New Call .......20
8.3.2. Receipt of QSIG CALL PROCEEDING Message ............21
8.3.3. Receipt of QSIG PROGRESS Message ...................22
8.3.4. Receipt of QSIG ALERTING Message ...................22
8.3.5. Inclusion of SDP Information in a SIP 18x
Provisional Response ...............................23
8.3.6. Receipt of QSIG CONNECT Message ....................24
8.3.7. Receipt of SIP PRACK Request .......................25
8.3.8. Receipt of SIP ACK Request .........................25
8.3.9. Receipt of a SIP INVITE Request for a Call
Already Being ......................................25
8.4. Call Clearing and Call Failure ............................26
8.4.1. Receipt of a QSIG DISCONNECT, RELEASE, or
RELEASE COMPLETE ...................................26
8.4.2. Receipt of a SIP BYE Request .......................29
8.4.3. Receipt of a SIP CANCEL Request ....................29
8.4.4. Receipt of a SIP 4xx-6xx Response to an
INVITE Request .....................................29
8.4.5. Gateway-Initiated Call Clearing ....................32
8.5. Request to Change Media Characteristics ...................32
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9. Number Mapping .................................................32
9.1. Mapping from QSIG to SIP ..................................33
9.1.1. Using Information from the QSIG Called
Party Number Information Element ...................33
9.1.2. Using Information from the QSIG Calling
Party Number Information Element ...................33
9.1.3. Using Information from the QSIG Connected
Number Information Element .........................35
9.2. Mapping from SIP to QSIG ..................................36
9.2.1. Generating the QSIG Called Party Number
Information Element ................................36
9.2.2. Generating the QSIG Calling Party Number
Information Element ................................37
9.2.3. Generating the QSIG Connected Number
Information Element ................................38
10. Requirements for Support of Basic Services ....................39
10.1. Derivation of QSIG Bearer Capability Information
Element ..................................................39
10.2. Derivation of Media Type in SDP ..........................39
11. Security Considerations .......................................40
11.1. General ..................................................40
11.2. Calls from QSIG to Invalid or Restricted Numbers .........40
11.3. Abuse of SIP Response Code ...............................41
11.4. Use of the To Header URI .................................41
11.5. Use of the From Header URI ...............................41
11.6. Abuse of Early Media .....................................42
11.7. Protection from Denial-of-Service Attacks ................42
12. Acknowledgements ..............................................43
13. Normative References ..........................................43
Appendix A. Example Message Sequences .............................45
Elwell, et al. Best Current Practice [Page 3]
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1. Introduction
This document specifies signalling interworking between QSIG and the
Session Initiation Protocol (SIP) in support of basic services within
a corporate telecommunication network (CN) (also known as enterprise
network).
QSIG is a signalling protocol that operates between Private
Integrated Services 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 Ecma Standards; in particular, [2] (call control in support of
basic services), [3] (generic functional protocol for the support of
supplementary services), and a number of standards specifying
individual supplementary services.
NOTE: The name QSIG was derived from the fact that it is used for
signalling at the Q reference point. The Q reference point is a
point of demarcation between two PINXs.
SIP is an application-layer protocol for establishing, terminating,
and modifying multimedia sessions. It is typically carried over IP
[15], [16]. Telephone calls are considered a type of multimedia
session where just audio is exchanged. SIP is defined in [10].
As the support of telephony within corporate networks evolves from
circuit-switched technology to Internet technology, the two
technologies will coexist in many networks for a period, perhaps
several years. Therefore, there is a need to be able to establish,
modify, and terminate sessions involving a participant in the SIP
network and a participant in the QSIG network. Such calls are
supported by gateways that perform interworking between SIP and QSIG.
This document specifies SIP-QSIG signalling interworking for basic
services that provide a bi-directional transfer capability for
speech, DTMF, facsimile, and modem media between a PISN employing
QSIG and a corporate IP network employing SIP. Other aspects of
interworking, e.g., the use of RTP and SDP, will differ according to
the type of media concerned and are outside the scope of this
specification.
Call-related and call-independent signalling in support of
supplementary services is outside the scope of this specification,
but support for certain supplementary services (e.g., call transfer,
call diversion) could be the subject of future work.
Elwell, et al. Best Current Practice [Page 4]
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Interworking between QSIG and SIP permits a call originating at a
user of a PISN to terminate at a user of a corporate IP network, or a
call originating at a user of a corporate IP network to terminate at
a user of a PISN.
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. Terminology
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 [4] and
indicate requirement levels for compliant SIP implementations.
3. Definitions
For the purposes of this specification, the following definitions
apply.
3.1. External Definitions
The definitions in [2] and [10] apply as appropriate.
3.2. Other definitions
3.2.1. 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: A CN can comprise a PISN, a private IP network (intranet), or a
combination of the two.
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3.2.2. Gateway
An entity that performs interworking between a PISN using QSIG and an
IP network using SIP.
3.2.3. 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.
3.2.4. Media Stream
Audio or other user information transmitted in UDP packets, typically
containing RTP, in a single direction between the gateway and a peer
entity participating in a session established using SIP.
NOTE: Normally a SIP session establishes a pair of media streams, one
in each direction.
3.2.5. Private Integrated Services Network (PISN)
A CN or part of a CN that employs circuit-switched technology.
3.2.6. Private Integrated Services Network Exchange (PINX)
A PISN nodal entity comprising switching and call handling functions
and supporting QSIG signalling in accordance with [2].
4. Acronyms
DNS Domain Name Service
IP Internet Protocol
PINX Private Integrated services Network eXchange
PISN Private Integrated Services Network
RTP Real-time Transport Protocol
SCTP Stream Control Transmission Protocol
SDP Session Description Protocol
SIP Session Initiation Protocol
TCP Transmission Control Protocol
TLS Transport Layer Security
TU Transaction User
UA User Agent
UAC User Agent Client
UAS User Agent Server
UDP User Datagram Protocol
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5. Background and Architecture
During the 1980s, corporate voice telecommunications adopted
technology similar in principle to Integrated Services Digital
Networks (ISDN). Digital circuit switches, commonly known as Private
Branch eXchanges (PBX) or more formally as Private Integrated
services Network eXchanges (PINX) have been interconnected by digital
transmission systems to form Private Integrated Services Networks
(PISN). These digital transmission systems carry voice or other
payload in fixed-rate channels, typically 64 Kbit/s, and signalling
in a separate channel. A technique known as common channel
signalling is employed, whereby a single signalling channel
potentially controls a number of payload channels or bearer channels.
A typical arrangement is a point-to-point transmission facility at T1
or E1 rate providing a 64 Kbit/s signalling channel and 23 or 30
bearer channels, respectively. Other arrangements are possible and
have been deployed, including the use of multiple transmission
facilities for a signalling channel and its logically associated
bearer channels. Also, arrangements involving bearer channels at
sub-64 Kbit/s have been deployed, where voice payload requires the
use of codecs that perform compression.
QSIG is the internationally-standardized message-based signalling
protocol for use in networks as described above. It runs in a
signalling channel between two PINXs and controls calls on a number
of logically associated bearer channels between the same two PINXs.
The signalling channel and its logically associated bearer channels
are collectively known as an inter-PINX link. QSIG is independent of
the type of transmission capabilities over which the signalling
channel and bearer channels are provided. QSIG is also independent
of the transport protocol used to transport QSIG messages reliably
over the signalling channel.
QSIG provides a means for establishing and clearing calls that
originate and terminate on different PINXs. A call can be routed
over a single inter-PINX link connecting the originating and
terminating PINX, or over several inter-PINX links in series with
switching at intermediate PINXs known as transit PINXs. A call can
originate or terminate in another network, in which case it enters or
leaves the PISN environment through a gateway PINX. Parties are
identified by numbers, in accordance with either [17] or a private
numbering plan. This basic call capability is specified in [2]. In
addition to basic call capability, QSIG specifies a number of further
capabilities supporting the use of supplementary services in PISNs.
More recently, corporate telecommunications networks have started to
exploit IP in various ways. One way is to migrate part of the
network to IP using SIP. This might, for example, be a new branch
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office with a SIP proxy and SIP endpoints instead of a PINX.
Alternatively, SIP equipment might be used to replace an existing
PINX or PINXs. The new SIP environment needs to interwork with the
QSIG-based PISN in order to support calls originating in one
environment and terminating in the other. Interworking is achieved
through a gateway.
Interworking between QSIG and SIP at gateways can also be used where
a SIP network interconnects different parts of a PISN, thereby
allowing calls between the different parts. A call can enter the SIP
network at one gateway and leave at another. Each gateway would
behave in accordance with this specification.
Another way of connecting two parts of a PISN would be to encapsulate
QSIG signalling in SIP messages for calls between the two parts.
This is outside the scope of this specification but could be the
subject of future work.
This document specifies signalling protocol interworking aspects of a
gateway between a PISN employing QSIG signalling and an IP network
employing SIP signalling. The gateway appears as a PINX to other
PINXs in the PISN. The gateway appears as a SIP endpoint to other
SIP entities in the IP network. The environment is shown in Figure
1.
+------+ IP network PISN
| |
|SIP | +------+
|Proxy | /| |
| | / |PINX |
+---+--+ *-----------+ / | |
| | | +-----+/ +------+
| | | | |
| | | |PINX |
---+-----+-------+--------+ Gateway +--------| |
| | | | | |\
| | | | +-----+ \
| | | | \ +------+
| | | | \| |
+--+---+ +--+---+ *-----------+ |PINX |
|SIP | |SIP | | |
|End- | |End- | +------+
|point | |point |
+------+ +------+
Figure 1: Environment
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In addition to the signalling interworking functionality specified in
this specification, it is assumed that the gateway also includes the
following functionality:
- one or more physical interfaces on the PISN side supporting one or
more inter-PINX links, each link providing one or more constant bit
rate channels for media streams and a reliable layer 2 connection
(e.g., over a fixed rate physical channel) for transporting QSIG
signalling messages; and
- one or more physical interfaces on the IP network side supporting,
through layer 1 and layer 2 protocols, IP as the network layer
protocol and UDP [6] and TCP [5] as transport layer protocols,
these being used for the transport of SIP signalling messages and,
in the case of UDP, also for media streams;
- optionally the support of TLS [7] and/or SCTP [9] as additional
transport layer protocols on the IP network side, these being used
for the transport of SIP signalling messages; and
- a means of transferring media streams in each direction between the
PISN and the IP network, including as a minimum packetization of
media streams sent to the IP network and de-packetization of media
streams received from the IP network.
NOTE: [10] mandates support for both UDP and TCP for the transport of
SIP messages and allows optional support for TLS and/or SCTP for this
same purpose.
The protocol model relevant to signalling interworking functionality
of a gateway is shown in Figure 2.
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+---------------------------------------------------------+
| Interworking function |
| |
+-----------------------+---------+-----------------------+
| | | |
| SIP | | |
| | | |
+-----------------------+ | |
| | | |
| UDP/TCP/TLS/SCTP | | QSIG |
| | | |
+-----------------------+ | |
| | | |
| IP | | |
| | | |
+-----------------------+ +-----------------------+
| IP network | | PISN |
| lower layers | | lower layers |
| | | |
+-----------------------+ +-----------------------+
Figure 2: Protocol model
In Figure 2, the SIP box represents SIP syntax and encoding, the SIP
transport layer, and the SIP transaction layer. The Interworking
function includes SIP Transaction User (TU) functionality.
6. Overview
The gateway maps received QSIG messages, where appropriate, to SIP
messages and vice versa and maintains an association between a QSIG
call and a SIP dialog.
A call from QSIG to SIP is initiated when a QSIG SETUP message
arrives at the gateway. The QSIG SETUP message initiates QSIG call
establishment, and an initial response message (e.g., CALL
PROCEEDING) completes negotiation of the bearer channel to be used
for that call. The gateway then sends a SIP INVITE request, having
translated the QSIG called party number to a URI suitable for
inclusion in the Request-URI. The SIP INVITE request and the
resulting SIP dialog, if successfully established, are associated
with the QSIG call. The SIP 2xx response to the INVITE request is
mapped to a QSIG CONNECT message, signifying answer of the call.
During establishment, media streams established by SIP and SDP are
connected to the bearer channel.
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A call from SIP to QSIG is initiated when a SIP INVITE request
arrives at the gateway. The gateway sends a QSIG SETUP message to
initiate QSIG call establishment, having translated the SIP Request-
URI to a number suitable for use as the QSIG called party number.
The resulting QSIG call is associated with the SIP INVITE request and
with the eventual SIP dialog. Receipt of an initial QSIG response
message completes negotiation of the bearer channel to be used,
allowing media streams established by SIP and SDP to be connected to
that bearer channel. The QSIG CONNECT message is mapped to a SIP 200
OK response to the INVITE request.
Appendix A gives examples of typical message sequences that can
arise.
7. General Requirements
In order to conform to this specification, a gateway SHALL support
QSIG in accordance with [2] as a gateway and SHALL support SIP in
accordance with [10] as a UA. In particular, the gateway SHALL
support SIP syntax and encoding, the SIP transport layer, and the SIP
transaction layer in accordance with [10]. In addition, the gateway
SHALL support SIP TU behaviour for a UA in accordance with [10]
except where stated otherwise in Sections 8, 9, and 10 of this
specification.
NOTE: [10] mandates that a SIP entity support both UDP and TCP as
transport layer protocols for SIP messages. Other transport layer
protocols can also be supported.
The gateway SHALL also support SIP reliable provisional responses in
accordance with [11] as a UA.
NOTE: [11] makes provision for recovering from loss of provisional
responses (other than 100) to INVITE requests when using unreliable
transport services in the IP network. This is important for ensuring
delivery of responses that map to essential QSIG messages.
The gateway SHALL support SDP in accordance with [8] and its use in
accordance with the offer/answer model in [12].
Section 9 also specifies optional use of the Privacy header in
accordance with [13] and the P-Asserted-Identity header in accordance
with [14].
The gateway SHALL support calls from QSIG to SIP and calls from SIP
to QSIG.
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SIP methods not defined in [10] or [11] are outside the scope of this
specification but could be the subject of other specifications for
interworking with QSIG, e.g., for interworking in support of
supplementary services.
As a result of DNS lookup by the gateway in order to determine where
to send a SIP INVITE request, a number of candidate destinations can
be attempted in sequence. The way in which this is handled by the
gateway is outside the scope of this specification. However, any
behaviour specified in this document on receipt of a SIP 4xx or 5xx
final response to an INVITE request SHOULD apply only when there are
no more candidate destinations to try or when overlap signalling
applies in the SIP network (see 8.2.2.2).
8. Message Mapping Requirements
8.1. Message Validation and Handling of Protocol Errors
The gateway SHALL validate received QSIG messages in accordance with
the requirements of [2] and SHALL act in accordance with [2] on
detection of a QSIG protocol error. The requirements of this section
for acting on a received QSIG message apply only to a received QSIG
message that has been successfully validated and that satisfies one
of the following conditions:
-the QSIG message is a SETUP message and indicates a destination in
the IP network and a bearer capability for which the gateway is able
to provide interworking; or
-the QSIG message is a message other than SETUP and contains a call
reference that identifies an existing call for which the gateway is
providing interworking between QSIG and SIP.
The processing of any valid QSIG message that does not satisfy any of
these conditions is outside the scope of this specification. Also,
the processing of any QSIG message relating to call-independent
signalling connections or connectionless transport, as specified in
[3], is outside the scope of this specification.
If segmented QSIG messages are received, the gateway SHALL await
receipt of all segments of a message and SHALL validate and act on
the complete reassembled message.
The gateway SHALL validate received SIP messages (requests and
responses) in accordance with the requirements of [10] and SHALL act
in accordance with [10] on detection of a SIP protocol error.
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Requirements of this section for acting on a received SIP message
apply only to a received message that has been successfully validated
and that satisfies one of the following conditions:
- the SIP message is an INVITE request that contains no tag parameter
in the To header field, does not match an ongoing transaction
(i.e., is not a merged request; see Section 8.2.2.2 of [10]), and
indicates a destination in the PISN for which the gateway is able
to provide interworking; or
- the SIP message is a request that relates to an existing dialog
representing a call for which the gateway is providing interworking
between QSIG and SIP; or
- the SIP message is a CANCEL request that relates to a received
INVITE request for which the gateway is providing interworking with
QSIG but for which the only response sent is informational (1xx),
no dialog having been confirmed; or
- the SIP message is a response to a request sent by the gateway in
accordance with this section.
The processing of any valid SIP message that does not satisfy any of
these conditions is outside the scope of this specification.
NOTE: These rules mean that an error detected in a received message
will not be propagated to the other side of the gateway. However,
there can be an indirect impact on the other side of the gateway,
e.g., the initiation of call clearing procedures.
The gateway SHALL run QSIG protocol timers as specified in [2] and
SHALL act in accordance with [2] if a QSIG protocol timer expires.
Any other action on expiry of a QSIG protocol timer is outside the
scope of this specification, except that if it results in the
clearing of the QSIG call, the gateway SHALL also clear the SIP call
in accordance with Section 8.4.5.
The gateway SHALL run SIP protocol timers as specified in [10] and
SHALL act in accordance with [10] if a SIP protocol timer expires.
Any other action on expiry of a SIP protocol timer is outside the
scope of this specification, except that if it results in the
clearing of the SIP call, the gateway SHALL also clear the QSIG call
in accordance with Section 8.4.5.
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8.2. Call Establishment from QSIG to SIP
8.2.1. Call Establishment from QSIG to SIP Using En Bloc Procedures
The following procedures apply when the gateway receives a QSIG SETUP
message containing a Sending Complete information element or the
gateway receives a QSIG SETUP message and is able to determine that
the number in the Called party number information element is
complete.
NOTE: In the absence of a Sending Complete information element, the
means by which the gateway determines the number to be complete is an
implementation matter. It can involve knowledge of the numbering
plan and/or use of inter-digit timer expiry.
8.2.1.1. Receipt of QSIG SETUP Message
On receipt of a QSIG SETUP message containing a number that the
gateway determines to be complete in the Called party number
information element, or containing a Sending complete information
element and a number that could potentially be complete, the gateway
SHALL map the QSIG SETUP message to a SIP INVITE request. The
gateway SHALL also send a QSIG CALL PROCEEDING message.
The gateway SHALL generate the SIP Request-URI, To, and From fields
in the SIP INVITE request in accordance with Section 9. The gateway
SHALL include in the INVITE request a Supported header containing
option tag 100rel, to indicate support for [11].
The gateway SHALL include SDP offer information in the SIP INVITE
request as described in Section 10. It SHOULD also connect the
incoming media stream to the user information channel of the inter-
PINX link, to allow the caller to hear in-band tones or announcements
and prevent speech clipping on answer. Because of forking, the
gateway may receive more than one media stream, in which case it
SHOULD select one (e.g., the first received). If the gateway is able
to correlate an unselected media stream with a particular early
dialog established using a reliable provisional response, it MAY use
the UPDATE method [19] to stop that stream and then use the UPDATE
method to start that stream again if a 2xx response is received on
that dialog.
On receipt of a QSIG SETUP message containing a Sending complete
information element and a number that the gateway determines to be
incomplete in the Called party number information element, the
gateway SHALL initiate QSIG call clearing procedures using cause
value 28, "invalid number format (address incomplete)".
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If information in the QSIG SETUP message is unsuitable for generating
any of the mandatory fields in a SIP INVITE request (e.g., if a
Request-URI cannot be derived from the QSIG Called party number
information element) or for generating SDP information, the gateway
SHALL NOT issue a SIP INVITE request and SHALL initiate QSIG call
clearing procedures in accordance with [2].
8.2.1.2. Receipt of SIP 100 (Trying) Response to an INVITE Request
A SIP 100 response SHALL NOT trigger any QSIG messages. It only
serves the purpose of suppressing INVITE request retransmissions.
8.2.1.3. Receipt of SIP 18x provisional response to an INVITE request
The gateway SHALL map a received SIP 18x response to an INVITE
request to a QSIG PROGRESS or ALERTING message based on the following
conditions.
- If a SIP 180 response is received and no QSIG ALERTING message has
been sent, the gateway SHALL generate a QSIG ALERTING message. The
gateway MAY supply ring-back tone on the user information channel of
the inter-PINX link, in which case the gateway SHALL include progress
description number 8 in the QSIG ALERTING message. Otherwise the
gateway SHALL NOT include progress description number 8 in the QSIG
ALERTING message unless the gateway is aware that in-band information
(e.g., ring-back tone) is being transmitted.
- If a SIP 181/182/183 response is received, no QSIG ALERTING message
has been sent, and no message containing progress description number
1 has been sent, the gateway SHALL generate a QSIG PROGRESS message
containing progress description number 1.
NOTE: This will ensure that QSIG timer T310 is stopped if running at
the Originating PINX.
In all other scenarios, the gateway SHALL NOT map the SIP 18x
response to a QSIG message.
If the SIP 18x response contains a Require header with option tag
100rel, the gateway SHALL send back a SIP PRACK request in accordance
with [11].
8.2.1.4. Receipt of SIP 2xx Response to an INVITE Request
If the gateway receives a SIP 2xx response as the first SIP 2xx
response to a SIP INVITE request, the gateway SHALL map the SIP 2xx
response to a QSIG CONNECT message. The gateway SHALL also send a
SIP ACK request to acknowledge the 2xx response. The gateway SHALL
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NOT include any SDP information in the SIP ACK request. If the
gateway receives further 2xx responses, it SHALL respond to each in
accordance with [10], SHOULD issue a BYE request for each, and SHALL
NOT generate any further QSIG messages.
Media streams will normally have been established in the IP network
in each direction. If so, the gateway SHALL connect the media
streams to the corresponding user-information channel on the inter-
PINX link if it has not already done so and stop any local ring-back
tone.
If the SIP 2xx response is received in response to the SIP PRACK
request, the gateway SHALL NOT map this message to any QSIG message.
NOTE: A SIP 2xx response to the INVITE request can be received later
on a different dialog as a result of a forking proxy.
8.2.1.5. Receipt of SIP 3xx Response to an INVITE Request
On receipt of a SIP 3xx response to an INVITE request, the gateway
SHALL act in accordance with [10].
NOTE: This will normally result in sending a new SIP INVITE request.
Unless the gateway supports the QSIG Call Diversion Supplementary
Service, no QSIG message SHALL be sent. The definition of Call
Diversion Supplementary Service for QSIG to SIP interworking is
beyond the scope of this specification.
8.2.2. Call Establishment from QSIG to SIP Using Overlap Procedures
SIP uses en bloc signalling, and it is strongly RECOMMENDED to avoid
using overlap signalling in a SIP network. A SIP/QSIG gateway
dealing with overlap signalling SHOULD perform a conversion from
overlap to en bloc signalling method using one or more of the
following mechanisms:
- timers;
- numbering plan information;
- the presence of a Sending complete information element in a
received QSIG INFORMATION message.
If the gateway performs a conversion from overlap to en bloc
signalling in the SIP network, then the procedures defined in Section
8.2.2.1 SHALL apply.
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However, for some applications it might be impossible to avoid using
overlap signalling in the SIP network. In this case, the procedures
defined in Section 8.2.2.2 SHALL apply.
8.2.2.1. En Bloc Signalling in SIP Network
8.2.2.1.1. Receipt of QSIG SETUP Message
On receipt of a QSIG SETUP message containing no Sending complete
information element and a number in the Called party number
information element that the gateway cannot determine to be complete,
the gateway SHALL send back a QSIG SETUP ACKNOWLEDGE message, start
QSIG timer T302, and await further number digits.
8.2.2.1.2. Receipt of QSIG INFORMATION Message
On receipt of each QSIG INFORMATION message containing no Sending
complete information element and containing a number that the gateway
cannot determine to be complete, QSIG timer T302 SHALL be restarted.
When QSIG timer T302 expires or a QSIG INFORMATION message containing
a Sending complete information element is received, the gateway SHALL
send a SIP INVITE request as described in Section 8.2.1.1. The
Request-URI and To fields (see Section 9) SHALL be generated from the
concatenation of information in the Called party number information
element in the received QSIG SETUP and INFORMATION messages. The
gateway SHALL also send a QSIG CALL PROCEEDING message.
8.2.2.1.3. Receipt of SIP Responses to INVITE Requests
SIP responses to INVITE requests SHALL be mapped as described in
8.2.1.
8.2.2.2. Overlap Signalling in SIP Network
The procedures below for using overlap signalling in the SIP network
are in accordance with the principles described in [18] for using
overlap sending when interworking with ISDN User Part (ISUP). In
[18], there is discussion of some potential problems arising from the
use of overlap sending in the SIP network. These potential problems
are applicable also in the context of QSIG-SIP interworking and can
be avoided if overlap sending in the QSIG network is terminated at
the gateway, in accordance with Section 8.2.2.1. The procedures
below should be used only where it is not feasible to use the
procedures of Section 8.2.2.1.
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8.2.2.2.1. Receipt of QSIG SETUP Message
On receipt of a QSIG SETUP message containing no Sending complete
information element and a number in the Called party number
information element that the gateway cannot determine to be complete,
the gateway SHALL send back a QSIG SETUP ACKNOWLEDGE message and
start QSIG timer T302. If the QSIG SETUP message contains the
minimum number of digits required to route the call in the IP
network, the gateway SHALL send a SIP INVITE request as specified in
Section 8.2.1.1. Otherwise, the gateway SHALL wait for more digits
to arrive in QSIG INFORMATION messages.
8.2.2.2.2. Receipt of QSIG INFORMATION Message
On receipt of a QSIG INFORMATION message, the gateway SHALL handle
the QSIG timer T302 in accordance with [2].
NOTE: [2] requires the QSIG timer to be stopped if the INFORMATION
message contains a Sending complete information element or to be
restarted otherwise.
Further behaviour of the gateway SHALL depend on whether or not it
has already sent a SIP INVITE request. If the gateway has not sent a
SIP INVITE request and it now has the minimum number of digits
required to route the call, it SHALL send a SIP INVITE request as
specified in Section 8.2.2.1.2. If the gateway still does not have
the minimum number of digits required, it SHALL wait for more QSIG
INFORMATION messages to arrive.
If the gateway has already sent one or more SIP INVITE requests,
whether or not final responses to those requests have been received,
it SHALL send a new SIP INVITE request in accordance with Section 3.2
of [18]. The updated Request-URI and To fields (see Section 9) SHALL
be generated from the concatenation of information in the Called
party number information element in the received QSIG SETUP and
INFORMATION messages.
NOTE: [18] requires the new request to have the same Call-ID and the
same From header (including tag) as in the previous INVITE request.
[18] recommends that the CSeq header should contain a value higher
than that in the previous INVITE request.
8.2.2.2.3. Receipt of SIP 100 (Trying) Response to an INVITE Request
The requirements of Section 8.2.1.2 SHALL apply.
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8.2.2.2.4. Receipt of SIP 18x Provisional Response to an INVITE Request
The requirements of Section 8.2.1.3 SHALL apply.
8.2.2.2.5. Receipt of SIP 2xx Response to an INVITE Request
The requirements of Section 8.2.1.4 SHALL apply. In addition, the
gateway SHALL send a SIP CANCEL request in accordance with Section
3.4 of [18] to cancel any SIP INVITE transactions for which no final
response has been received.
8.2.2.2.6. Receipt of SIP 3xx Response to an INVITE Request
The requirements of Section 8.2.1.5 SHALL apply.
8.2.2.2.7. Receipt of a SIP 4xx, 5xx, or 6xx Final Response to an
INVITE Request
On receipt of a SIP 4xx, 5xx, or 6xx final response to an INVITE
request, the gateway SHALL send back a SIP ACK request. Unless the
gateway is able to retry the INVITE request to avoid the problem
(e.g., by supplying authentication in the case of a 401 or 407
response), the gateway SHALL also send a QSIG DISCONNECT message
(8.4.4) if no further QSIG INFORMATION messages are expected and
final responses have been received to all transmitted SIP INVITE
requests.
NOTE: Further QSIG INFORMATION messages will not be expected after
QSIG timer T302 has expired or after a Sending complete information
element has been received.
In all other cases, the receipt of a SIP 4xx, 5xx, or 6xx final
response to an INVITE request SHALL NOT trigger the sending of any
QSIG message.
NOTE: If further QSIG INFORMATION messages arrive, these will result
in further SIP INVITE requests being sent, one of which might result
in successful call establishment. For example, initial INVITE
requests might produce 484 (Address Incomplete) or 404 (Not Found)
responses because the Request-URIs derived from incomplete numbers
cannot be routed, yet a subsequent INVITE request with a routable
Request-URI might produce a 2xx final response or a more meaningful
4xx, 5xx, or 6xx final response.
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8.2.2.2.8. Receipt of Multiple SIP Responses to an INVITE Request
Section 3.3 of [18] applies.
8.2.2.2.9. Cancelling Pending SIP INVITE Transactions
As stated in Section 3.4 of [18], when a gateway sends a new SIP
INVITE request containing new digits, it SHOULD NOT send a SIP CANCEL
request to cancel a previous SIP INVITE transaction that has not had
a final response. This SIP CANCEL request could arrive at an egress
gateway before the new SIP INVITE request and trigger premature call
clearing.
NOTE: Previous SIP INVITE transactions can be expected to result in
SIP 4xx class responses, which terminate the transaction. In Section
8.2.2.2.5, there is provision for cancelling any transactions still
in progress after a SIP 2xx response has been received.
8.2.2.2.10. QSIG Timer T302 Expiry
If QSIG timer T302 expires and the gateway has received 4xx, 5xx, or
6xx responses to all transmitted SIP INVITE requests, the gateway
SHALL send a QSIG DISCONNECT message. If T302 expires and the
gateway has not received 4xx, 5xx, or 6xx responses to all
transmitted SIP INVITE requests, the gateway SHALL ignore any further
QSIG INFORMATION messages but SHALL NOT send a QSIG DISCONNECT
message at this stage.
NOTE: A QSIG DISCONNECT request will be sent when all outstanding SIP
INVITE requests have received 4xx, 5xx, or 6xx responses.
8.3. Call Establishment from SIP to QSIG
8.3.1. Receipt of SIP INVITE Request for a New Call
On receipt of a SIP INVITE request for a new call, if a suitable
channel is available on the inter-PINX link, the gateway SHALL
generate a QSIG SETUP message from the received SIP INVITE request.
The gateway SHALL generate the Called party number and Calling party
number information elements in accordance with Section 9 and SHALL
generate the Bearer capability information element in accordance with
Section 10. If the gateway can determine that the number placed in
the Called party number information element is complete, the gateway
MAY include the Sending complete information element.
NOTE: The means by which the gateway determines the number to be
complete is an implementation matter. It can involve knowledge of
the numbering plan and/or use of the inter-digit timer.
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The gateway SHOULD send a SIP 100 (Trying) response.
If information in the SIP INVITE request is unsuitable for generating
any of the mandatory information elements in a QSIG SETUP message
(e.g., if a QSIG Called party number information element cannot be
derived from SIP Request-URI field) or if no suitable channel is
available on the inter-PINX link, the gateway SHALL NOT issue a QSIG
SETUP message and SHALL send a SIP 4xx, 5xx, or 6xx response. If no
suitable channel is available, the gateway should use response code
503 (Service Unavailable).
If the SIP INVITE request does not contain SDP information and does
not contain either a Required header or a Supported header with
option tag 100rel, the gateway SHOULD still proceed as above,
although an implementation can instead send a SIP 488 (Not Acceptable
Here) response, in which case it SHALL NOT issue a QSIG SETUP
message.
NOTE: The absence of SDP offer information in the SIP INVITE request
means that the gateway might need to send SDP offer information in a
provisional response and receive SDP answer information in a SIP
PRACK request (in accordance with [11]) in order to ensure that tones
and announcements from the PISN are transmitted. SDP offer
information cannot be sent in an unreliable provisional response
because SDP answer information would need to be returned in a SIP
PRACK request. The recommendation above still to proceed with call
establishment in this situation reflects the desire to maximise the
chances of a successful call. However, if important in-band
information is likely to be denied in this situation, a gateway can
choose not to proceed.
NOTE: If SDP offer information is present in the INVITE request, the
issuing of a QSIG SETUP message is not dependent on the presence of a
Required header or a Supported header with option tag 100rel.
On receipt of a SIP INVITE request relating to a call that has
already been established from SIP to QSIG, the procedures of 8.3.9
SHALL apply.
8.3.2. Receipt of QSIG CALL PROCEEDING Message
The receipt of a QSIG CALL PROCEEDING message SHALL NOT result in any
SIP message being sent.
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8.3.3. Receipt of QSIG PROGRESS Message
A QSIG PROGRESS message can be received in the event of interworking
on the remote side of the PISN or if the PISN is unable to complete
the call and generates an in-band tone or announcement. In the
latter case, a Cause information element is included in the QSIG
PROGRESS message.
The gateway SHALL map a received QSIG PROGRESS message to a SIP 183
(Session Progress) response to the INVITE request. If the SIP INVITE
request contained either a Require header or a Supported header with
option tag 100rel, the gateway SHALL include in the SIP 183 response
a Require header with option tag 100rel.
NOTE: In accordance with [11], inclusion of option tag 100rel in a
provisional response instructs the UAC to acknowledge the provisional
response by sending a PRACK request. [11] also specifies procedures
for repeating a provisional response with option tag 100rel if no
PRACK is received.
If the QSIG PROGRESS message contained a Progress indicator
information element with Progress description number 1 or 8, the
gateway SHALL connect the media streams to the corresponding user
information channel of the inter-PINX link if it has not already done
so, provided that SDP answer information is included in the
transmitted SIP response to the INVITE request or has already been
sent or received. Inclusion of SDP offer or answer information in
the 183 provisional response SHALL be in accordance with Section
8.3.5.
If the QSIG PROGRESS message is received with a Cause information
element, the gateway SHALL either wait until the tone/announcement is
complete or has been applied for sufficient time before initiating
call clearing, or wait for a SIP CANCEL request. If call clearing is
initiated, the cause value in the QSIG PROGRESS message SHALL be used
to derive the response to the SIP INVITE request in accordance with
Table 1.
8.3.4. Receipt of QSIG ALERTING Message
The gateway SHALL map a QSIG ALERTING message to a SIP 180 (Ringing)
response to the INVITE request. If the SIP INVITE request contained
either a Require header or a Supported header with option tag 100rel,
the gateway SHALL include in the SIP 180 response a Require header
with option tag 100rel.
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NOTE: In accordance with [11], inclusion of option tag 100rel in a
provisional response instructs the UAC to acknowledge the provisional
response by sending a PRACK request. [11] also specifies procedures
for repeating a provisional response with option tag 100rel if no
PRACK is received.
If the QSIG ALERTING message contained a Progress indicator
information element with Progress description number 1 or 8, the
gateway SHALL connect the media streams to the corresponding user
information channel of the inter-PINX link if it has not already done
so, provided that SDP answer information is included in the
transmitted SIP response or has already been sent or received.
Inclusion of SDP offer or answer information in the 180 provisional
response SHALL be in accordance with Section 8.3.5.
8.3.5. Inclusion of SDP Information in a SIP 18x Provisional Response
When sending a SIP 18x provisional response to the INVITE request, if
a QSIG message containing a Progress indicator information element
with progress description number 1 or 8 has been received the gateway
SHALL include SDP information. Otherwise, the gateway MAY include
SDP information. If SDP information is included, it shall be in
accordance with the following rules.
If the SIP INVITE request contained a Required or Supported header
with option tag 100rel, and if SDP offer and answer information has
already been exchanged, no SDP information SHALL be included in the
SIP 18x provisional response.
If the SIP INVITE request contained a Required or Supported header
with option tag 100rel, and if SDP offer information was received in
the SIP INVITE request but no SDP answer information has been sent,
SDP answer information SHALL be included in the SIP 18x provisional
response.
If the SIP INVITE request contained a Required or Supported header
with option tag 100rel, and if no SDP offer information was received
in the SIP INVITE request and no SDP offer information has already
been sent, SDP offer information SHALL be included in the SIP 18x
provisional response.
NOTE: In this case, SDP answer information can be expected in the SIP
PRACK.
If the SIP INVITE request contained neither a Required nor a
Supported header with option tag 100rel, SDP answer information SHALL
be included in the SIP 18x provisional response.
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NOTE: Because the provisional response is unreliable, SDP answer
information needs to be repeated in each provisional response and in
the final SIP 2xx response.
NOTE: If the SIP INVITE request contained no SDP offer information
and neither a Required nor a Supported header with option tag 100rel,
it should have been rejected in accordance with Section 8.3.1.
8.3.6. Receipt of QSIG CONNECT Message
The gateway SHALL map a QSIG CONNECT message to a SIP 200 (OK) final
response for the SIP INVITE request. The gateway SHALL also send a
QSIG CONNECT ACKNOWLEDGE message.
If the SIP INVITE request contained a Required or Supported header
with option tag 100rel, and if SDP offer and answer information has
already been exchanged, no SDP information SHALL be included in the
SIP 200 response.
If the SIP INVITE request contained a Required or Supported header
with option tag 100rel, and if SDP offer information was received in
the SIP INVITE request but no SDP answer information has been sent,
SDP answer information SHALL be included in the SIP 200 response.
If the SIP INVITE request contained a Required or Supported header
with option tag 100rel, and if no SDP offer information was received
in the SIP INVITE request and no SDP offer information has already
been sent, SDP offer information SHALL be included in the SIP 200
response.
NOTE: In this case, SDP answer information can be expected in the SIP
ACK.
If the SIP INVITE request contained neither a Required nor a
Supported header with option tag 100rel, SDP answer information SHALL
be included in the SIP 200 response.
NOTE: Because the provisional response is unreliable, SDP answer
information needs to be repeated in each provisional response and in
the final 2xx response.
NOTE: If the SIP INVITE request contained no SDP offer information
and neither a Required nor a Supported header with option tag 100rel,
it may have been rejected in accordance with Section 8.3.1.
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The gateway SHALL connect the media streams to the corresponding user
information channel of the inter-PINX link if it has not already done
so, provided that SDP answer information is included in the
transmitted SIP response or has already been sent or received.
8.3.7. Receipt of SIP PRACK Request
The receipt of a SIP PRACK request acknowledging a reliable
provisional response SHALL NOT result in any QSIG message being sent.
The gateway SHALL send back a SIP 200 (OK) response to the SIP PRACK
request.
If the SIP PRACK contains SDP answer information and a QSIG message
containing a Progress indicator information element with progress
description number 1 or 8 has been received, the gateway SHALL
connect the media streams to the corresponding user information
channel of the inter-PINX link.
8.3.8. Receipt of SIP ACK Request
The receipt of a SIP ACK request SHALL NOT result in any QSIG message
being sent.
If the SIP ACK contains SDP answer information, the gateway SHALL
connect the media streams to the corresponding user information
channel of the inter-PINX link if it has not already done so.
8.3.9. Receipt of a SIP INVITE Request for a Call Already Being
Established
A gateway can receive a call from SIP using overlap procedures. This
should occur when the UAC for the INVITE request is a gateway from a
network that employs overlap procedures (e.g., an ISUP gateway or
another QSIG gateway) and the gateway has not absorbed overlap.
For a call from SIP using overlap procedures, the gateway will
receive multiple SIP INVITE requests that belong to the same call but
have different Request-URI and To fields. Each SIP INVITE request
belongs to a different dialog.
A SIP INVITE request is considered to be for the purpose of overlap
sending if, compared to a previously received SIP INVITE request, it
has:
- the same Call-ID header;
- the same From header (including the tag);
- no tag in the To header;
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- an updated Request-URI from which can be derived a called party
number with a superset of the digits derived from the previously
received SIP INVITE request;
and if
- the gateway has not yet sent a final response other than 484 to
the previously received SIP INVITE request.
If a gateway receives a SIP INVITE request for the purpose of overlap
sending, it SHALL generate a QSIG INFORMATION message using the call
reference of the existing QSIG call instead of a new QSIG SETUP
message and containing only the additional digits in the Called party
number information element. It SHALL also respond to the SIP INVITE
request received previously with a SIP 484 Address Incomplete
response.
If a gateway receives a SIP INVITE request that meets all of the
conditions for a SIP INVITE request for the purpose of overlap
sending except the condition concerning the Request-URI, the gateway
SHALL respond to the new request with a SIP 485 (Ambiguous) response.
8.4. Call Clearing and Call Failure
8.4.1. Receipt of a QSIG DISCONNECT, RELEASE, or RELEASE COMPLETE
Message
On receipt of QSIG DISCONNECT, RELEASE, or RELEASE COMPLETE message
as the first QSIG call clearing message, gateway behaviour SHALL
depend on the state of call establishment.
1) If the gateway has sent a SIP 200 (OK) response to a SIP INVITE
request and received a SIP ACK request, or if it has received a
SIP 200 (OK) response to a SIP INVITE request and sent a SIP ACK
request, the gateway SHALL send a SIP BYE request to clear the
call.
2) If the gateway has sent a SIP 200 (OK) response to a SIP INVITE
request (indicating that call establishment is complete) but has
not received a SIP ACK request, the gateway SHALL wait until a SIP
ACK is received and then send a SIP BYE request to clear the call.
3) If the gateway has sent a SIP INVITE request and received a SIP
provisional response but not a SIP final response, the gateway
SHALL send a SIP CANCEL request to clear the call.
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NOTE 1: In accordance with [10], if after sending a SIP CANCEL
request a SIP 2xx response is received to the SIP INVITE request,
the gateway will need to send a SIP BYE request.
4) If the gateway has sent a SIP INVITE request but received no SIP
response, the gateway SHALL NOT send a SIP message. If a SIP
final or provisional response is subsequently received, the
gateway SHALL then act in accordance with 1, 2, or 3 above,
respectively.
5) If the gateway has received a SIP INVITE request but not sent a
SIP final response, the gateway SHALL send a SIP final response
chosen according to the cause value in the received QSIG message
as specified in Table 1. SIP response 500 (Server internal error)
SHALL be used as the default for cause values not shown in
Table 1.
NOTE 2: It is not necessarily appropriate to map some QSIG cause
values to SIP messages because these cause values are meaningful only
at the gateway. A good example of this is cause value 44, "Requested
circuit or channel not available", which signifies that the channel
number in the transmitted QSIG SETUP message was not acceptable to
the peer PINX. The appropriate behavior in this case is for the
gateway to send another SETUP message indicating a different channel
number. If this is not possible, the gateway should treat it either
as a congestion situation (no channels available; see Section 8.3.1)
or as a gateway failure situation (in which case the default SIP
response code applies).
In all cases, the gateway SHALL also disconnect media streams, if
established, and allow QSIG and SIP signalling to complete in
accordance with [2] and [10], respectively.
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Table 1: Mapping of QSIG Cause Value to SIP 4xx-6xx responses to an
INVITE request
QSIG Cause value SIP response
----------------------------------------------------------------
1 Unallocated number 404 Not found
2 No route to specified 404 Not found
transit network
3 No route to destination 404 Not found
16 Normal call clearing (NOTE 3)
17 User busy 486 Busy here
18 No user responding 408 Request timeout
19 No answer from the user 480 Temporarily unavailable
20 Subscriber absent 480 Temporarily unavailable
21 Call rejected 603 Decline, if location field
in Cause information element
indicates user. Otherwise:
403 Forbidden
22 Number changed 301 Moved permanently, if
information in diagnostic field
of Cause information element is
suitable for generating a SIP
Contact header. Otherwise:
410 Gone
23 Redirection to new 410 Gone
destination
27 Destination out of order 502 Bad gateway
28 Address incomplete 484 Address incomplete
29 Facility rejected 501 Not implemented
31 Normal, unspecified 480 Temporarily unavailable
34 No circuit/channel 503 Service unavailable
available
38 Network out of order 503 Service unavailable
41 Temporary failure 503 Service unavailable
42 Switching equipment 503 Service unavailable
congestion
47 Resource unavailable, 503 Service unavailable
unspecified
55 Incoming calls barred 403 Forbidden
within CUG
57 Bearer capability not 403 Forbidden
authorized
58 Bearer capability not 503 Service unavailable
presently available
65 Bearer capability not 488 Not acceptable here (NOTE 4)
implemented
69 Requested facility not 501 Not implemented
implemented
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70 Only restricted digital 488 Not acceptable here (NOTE 4)
information available
79 Service or option not 501 Not implemented
implemented, unspecified
87 User not member of CUG 403 Forbidden
88 Incompatible destination 503 Service unavailable
102 Recovery on timer expiry 504 Server time-out
NOTE 3: A QSIG call clearing message containing cause value 16 will
normally result in the sending of a SIP BYE or CANCEL request.
However, if a SIP response is to be sent to the INVITE request, the
default response code should be used.
NOTE 4: The gateway may include a SIP Warning header if diagnostic
information in the QSIG Cause information element allows a suitable
warning code to be selected.
8.4.2. Receipt of a SIP BYE Request
On receipt of a SIP BYE request, the gateway SHALL send a QSIG
DISCONNECT message with cause value 16 (normal call clearing). The
gateway SHALL also disconnect media streams, if established, and
allow QSIG and SIP signalling to complete in accordance with [2] and
[10], respectively.
NOTE: When responding to a SIP BYE request, in accordance with [10],
the gateway is also required to respond to any other outstanding
transactions, e.g., with a SIP 487 (Request Terminated) response.
This applies in particular if the gateway has not yet returned a
final response to the SIP INVITE request.
8.4.3. Receipt of a SIP CANCEL Request
On receipt of a SIP CANCEL request to clear a call for which the
gateway has not sent a SIP final response to the received SIP INVITE
request, the gateway SHALL send a QSIG DISCONNECT message with cause
value 16 (normal call clearing). The gateway SHALL also disconnect
media streams, if established, and allow QSIG and SIP signalling to
complete in accordance with [2] and [10], respectively.
8.4.4. Receipt of a SIP 4xx-6xx Response to an INVITE Request
Except where otherwise specified in the context of overlap sending
(8.2.2.2), on receipt of a SIP final response (4xx-6xx) to a SIP
INVITE request, unless the gateway is able to retry the INVITE
request to avoid the problem (e.g., by supplying authentication in
the case of a 401 or 407 response), the gateway SHALL transmit a QSIG
DISCONNECT message. The cause value in the QSIG DISCONNECT message
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SHALL be derived from the SIP 4xx-6xx response according to Table 2.
Cause value 31 (Normal, unspecified) SHALL be used as the default for
SIP responses not shown in Table 2. The gateway SHALL also
disconnect media streams, if established, and allow QSIG and SIP
signalling to complete in accordance with [2] and [10], respectively.
When generating a QSIG Cause information element, the location field
SHOULD contain the value "user", if generated as a result of a SIP
response code 6xx, or the value "Private network serving the remote
user" in other circumstances.
Table 2: Mapping of SIP 4xx-6xx responses to an INVITE request to
QSIG Cause values
SIP response QSIG Cause value (NOTE 6)
------------------------------------------------------------------
400 Bad request 41 Temporary failure
401 Unauthorized 21 Call rejected (NOTE 5)
402 Payment required 21 Call rejected
403 Forbidden 21 Call rejected
404 Not found 1 Unallocated number
405 Method not allowed 63 Service or option
unavailable, unspecified
406 Not acceptable 79 Service or option not
implemented, unspecified
407 Proxy Authentication required 21 Call rejected (NOTE 5)
408 Request timeout 102 Recovery on timer expiry
410 Gone 22 Number changed
413 Request entity too large 127 Interworking, unspecified
(NOTE 6)
414 Request-URI too long 127 Interworking, unspecified
(NOTE 6)
415 Unsupported media type 79 Service or option not
implemented, unspecified
(NOTE 6)
416 Unsupported URI scheme 127 Interworking, unspecified
(NOTE 6)
420 Bad extension 127 Interworking, unspecified
(NOTE 6)
421 Extension required 127 Interworking, unspecified
(NOTE 6)
423 Interval too brief 127 Interworking, unspecified
(NOTE 6)
480 Temporarily unavailable 18 No user responding
481 Call/transaction does not exist 41 Temporary failure
482 Loop detected 25 Exchange routing error
483 Too many hops 25 Exchange routing error
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484 Address incomplete 28 Invalid number format
(NOTE 6)
485 Ambiguous 1 Unallocated Number
486 Busy here 17 User busy
487 Request terminated (NOTE 7)
488 Not Acceptable Here 65 Bearer capability not
implemented or 31 Normal,
unspecified (NOTE 8)
500 Server internal error 41 Temporary failure
501 Not implemented 79 Service or option not
implemented, unspecified
502 Bad gateway 38 Network out of order
503 Service unavailable 41 Temporary failure
504 Gateway time-out 102 Recovery on timer expiry
505 Version not supported 127 Interworking, unspecified
(NOTE 6)
513 Message too large 127 Interworking, unspecified
(NOTE 6)
600 Busy everywhere 17 User busy
603 Decline 21 Call rejected
604 Does not exist anywhere 1 Unallocated number
606 Not acceptable 65 Bearer capability not
implemented or
31 Normal, unspecified (NOTE 8)
NOTE 5: In some cases, it may be possible for the gateway to provide
credentials to the SIP UAS that is rejecting an INVITE due to
authorization failure. If the gateway can authenticate itself, then
obviously it should do so and proceed with the call. Only if the
gateway cannot authorize itself should the gateway clear the call in
the QSIG network with this cause value.
NOTE 6: For some response codes, the gateway may be able to retry the
INVITE request in order to work around the problem. In particular,
this may be the case with response codes indicating a protocol error.
The gateway SHOULD clear the call in the QSIG network with the
indicated cause value only if retry is not possible or fails.
NOTE 7: The circumstances in which SIP response code 487 can be
expected to arise do not require it to be mapped to a QSIG cause
code, since the QSIG call will normally already be cleared or in the
process of clearing. If QSIG call clearing does, however, need to be
initiated, the default cause value should be used.
NOTE 8: When the Warning header is present in a SIP 606 or 488
message, the warning code should be examined to determine whether it
is reasonable to generate cause value 65. This cause value should be
generated only if there is a chance that a new call attempt with
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different content in the Bearer capability information element will
avoid the problem. In other circumstances, the default cause value
should be used.
8.4.5 Gateway-Initiated Call Clearing
If the gateway initiates clearing of the QSIG call owing to QSIG
timer expiry, QSIG protocol error, or use of the QSIG RESTART message
in accordance with [2], the gateway SHALL also initiate clearing of
the SIP call in accordance with Section 8.4.1. If this involves the
sending of a final response to a SIP INVITE request, the gateway
SHALL use response code 480 (Temporarily Unavailable) if optional
QSIG timer T301 has expired or, otherwise, response code 408 (Request
timeout) or 500 (Server internal error), as appropriate.
If the gateway initiates clearing of the SIP call owing to SIP timer
expiry or SIP protocol error in accordance with [10], the gateway
SHALL also initiate clearing of the QSIG call in accordance with [2]
using cause value 102 (Recovery on timer expiry) or 41 (Temporary
failure), as appropriate.
8.5. Request to Change Media Characteristics
If after a call has been successfully established the gateway
receives a SIP INVITE request to change the media characteristics of
the call in a way that would be incompatible with the bearer
capability in use within the PISN, the gateway SHALL send back a SIP
488 (Not Acceptable Here) response and SHALL NOT change the media
characteristics of the existing call.
9. Number Mapping
In QSIG, users are identified by numbers, as defined in [1]. Numbers
are conveyed within the Called party number, Calling party number,
and Connected number information elements. The Calling party number
and Connected number information elements also contain a presentation
indicator, which can indicate that privacy is required (presentation
restricted), and a screening indicator, which indicates the source
and authentication status of the number.
In SIP, users are identified by Universal Resource Identifiers (URIs)
conveyed within the Request-URI and various headers, including the
From and To headers specified in [10] and optionally the P-Asserted-
Identity header specified in [14]. In addition, privacy is indicated
by the Privacy header specified in [13].
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This clause specifies the mapping between QSIG Called party number,
Calling party number, and Connected number information elements and
corresponding elements in SIP.
A gateway MAY implement the P-Asserted-Identity header in accordance
with [14]. If a gateway implements the P-Asserted-Identity header,
it SHALL also implement the Privacy header in accordance with [13].
If a gateway does not implement the P-Asserted-Identity header, it
MAY implement the Privacy header.
9.1. Mapping from QSIG to SIP
The method used to convert a number to a URI is outside the scope of
this specification. However, the gateway SHOULD take account of the
Numbering Plan (NPI) and Type Of Number (TON) fields in the QSIG
information element concerned when interpreting a number.
Some aspects of mapping depend on whether the gateway is in the same
trust domain (as defined in [14]) as the next hop SIP node (i.e., the
proxy or UA to which the INVITE request is sent or from which INVITE
request is received) to honour requests for identity privacy in the
Privacy header. This will be network-dependent, and it is
RECOMMENDED that gateways supporting the P-Asserted-Identity header
hold a configurable list of next hop nodes that are to be trusted in
this respect.
9.1.1. Using Information from the QSIG Called Party Number Information
Element
When mapping a QSIG SETUP message to a SIP INVITE request, the
gateway SHALL convert the number in the QSIG Called party number
information to a URI and include that URI in the SIP Request-URI and
in the To header.
9.1.2. Using Information from the QSIG Calling Party Number Information
Element
When mapping a QSIG SETUP message to a SIP INVITE request, the
gateway SHALL use the Calling party number information element, if
present, as follows.
If the information element contains a number, the gateway SHALL
attempt to derive a URI from that number. Further behaviour depends
on whether a URI has been derived and the value of the presentation
indication.
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9.1.2.1. No URI derived, and presentation indicator does not have value
"presentation restricted"
In this case (including the case where the Calling party number
information element is absent), the gateway SHALL include a URI
identifying the gateway in the From header. Also, if the gateway
supports the mechanism defined in [14], the gateway SHALL NOT
generate a P-Asserted-Identity header.
9.1.2.2. No URI derived, and presentation indicator has value
"presentation restricted"
In this case, the gateway SHALL generate an anonymous From header.
Also, if the gateway supports the mechanism defined in [14], the
gateway SHALL generate a Privacy header field with parameter
priv-value = "id" and SHALL NOT generate a P-Asserted-Identity
header. The inclusion of additional values of the priv-value
parameter in the Privacy header is outside the scope of this
specification.
9.1.2.3. URI derived, and presentation indicator has value
"presentation restricted"
If the gateway supports the P-Asserted-Identity header and trusts the
next hop proxy to honour the Privacy header, the gateway SHALL
generate a P-Asserted-Identity header containing the derived URI,
SHALL generate a Privacy header with parameter priv-value = "id", and
SHALL generate an anonymous From header. The inclusion of additional
values of the priv-value parameter in the Privacy header is outside
the scope of this specification.
If the gateway does not support the P-Asserted-Identity header or
does not trust the proxy to honour the Privacy header, the gateway
SHALL behave as in Section 9.1.2.2.
9.1.2.4. URI derived, and presentation indicator does not have value
"presentation restricted"
In this case, the gateway SHALL generate a P-Asserted-Identity header
containing the derived URI if the gateway supports this header, SHALL
NOT generate a Privacy header, and SHALL include the derived URI in
the From header. In addition, the gateway MAY use S/MIME, as
described in Section 23 of [10], to sign a copy of the From header
included in a message/sipfrag body of the INVITE request as described
in [20].
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9.1.3. Using Information from the QSIG Connected Number Information
Element
When mapping a QSIG CONNECT message to a SIP 200 (OK) response to an
INVITE request, the gateway SHALL use the Connected number
information element, if present, as follows.
If the information element contains a number, the gateway SHALL
attempt to derive a URI from that number. Further behaviour depends
on whether a URI has been derived and the value of the presentation
indication.
9.1.3.1. No URI derived, and presentation indicator does not have value
"presentation restricted"
In this case (including the case where the Connected number
information element is absent), the gateway SHALL NOT generate a
P-Asserted-Identity header and SHALL NOT generate a Privacy header.
9.1.3.2. No URI derived, and presentation indicator has value
"presentation restricted"
In this case, if the gateway supports the mechanism defined in [14],
the gateway SHALL generate a Privacy header field with parameter
priv-value = "id" and SHALL NOT generate a P-Asserted-Identity
header. The inclusion of additional values of the priv-value
parameter in the Privacy header is outside the scope of this
specification.
9.1.3.3. URI derived, and presentation indicator has value
"presentation restricted"
If the gateway supports the P-Asserted-Identity header and trusts the
next hop proxy to honour the Privacy header, the gateway SHALL
generate a P-Asserted-Identity header containing the derived URI and
SHALL generate a Privacy header with parameter priv-value = "id".
The inclusion of additional values of the priv-value parameter in the
Privacy header is outside the scope of this specification.
If the gateway does not support the P-Asserted-Identity header or
does not trust the proxy to honour the Privacy header, the gateway
SHALL behave as in Section 9.1.3.2.
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9.1.3.4. URI derived, and presentation indicator does not have value
"presentation restricted"
In this case, the gateway SHALL generate a P-Asserted-Identity header
containing the derived URI if the gateway supports this header and
SHALL NOT generate a Privacy header. In addition, the gateway MAY
use S/MIME, as described in Section 23 of [10], to sign a To header
containing the derived URI, the To header being included in a
message/sipfrag body of the INVITE response as described in [20].
NOTE: The To header in the message/sipfrag body may differ from the
to header in the response's headers.
9.2. Mapping from SIP to QSIG
The method used to convert a URI to a number is outside the scope of
this specification. However, NPI and TON fields in the QSIG
information element concerned SHALL be set to appropriate values in
accordance with [1].
Some aspects of mapping depend on whether the gateway trusts the next
hop SIP node (i.e., the proxy or UA to which the INVITE request is
sent or from which INVITE request is received) to provide accurate
information in the P-Asserted-Identity header. This will be
network-dependent, and it is RECOMMENDED that gateways hold a
configurable list of next hop nodes that are to be trusted in this
respect.
Some aspects of mapping depend on whether the gateway is prepared to
use a URI in the From header to derive a number for the Calling party
number information element. The default behaviour SHOULD be not to
use an unsigned or unvalidated From header for this purpose, since in
principle the information comes from an untrusted source (the remote
UA). However, it is recognised that some network administrations may
believe that the benefits to be derived from supplying a calling
party number outweigh any risks of supplying false information.
Therefore, a gateway MAY be configurable to use an unsigned or
unvalidated From header for this purpose.
9.2.1. Generating the QSIG Called Party Number Information Element
When mapping a SIP INVITE request to a QSIG SETUP message, the
gateway SHALL convert the URI in the SIP Request-URI to a number and
include that number in the QSIG Called party number information
element.
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NOTE: The To header should not be used for this purpose. This is
because re-targeting of the request in the SIP network can change the
Request-URI but leave the To header unchanged. It is important that
routing in the QSIG network be based on the final target from the SIP
network.
9.2.2. Generating the QSIG Calling Party Number Information Element
When mapping a SIP INVITE request to a QSIG SETUP message, the
gateway SHALL generate a Calling party number information element as
follows.
If the SIP INVITE request contains an S/MIME signed message/sipfrag
body [20] containing a From header, and if the gateway supports this
capability and can verify the authenticity and trustworthiness of
this information, the gateway SHALL attempt to derive a number from
the URI in that header. If no number is derived from a
message/sipfrag body, if the SIP INVITE request contains a P-
Asserted-Identity header, and if the gateway supports that header and
trusts the information therein, the gateway SHALL attempt to derive a
number from the URI in that header. If a number is derived from one
of these headers, the gateway SHALL include it in the Calling party
number information element and include value "network provided" in
the screening indicator.
If no number is derivable as described above and if the gateway is
prepared to use the unsigned or unvalidated From header, the gateway
SHALL attempt to derive a number from the URI in the From header. If
a number is derived from the From header, the gateway SHALL include
it in the Calling party number information element and include value
"user provided, not screened" in the screening indicator.
If no number is derivable, the gateway SHALL NOT include a number in
the Calling party number information element.
If the SIP INVITE request contains a Privacy header with value "id"
in parameter priv-value and the gateway supports this header, or if
the value in the From header indicates anonymous, the gateway SHALL
include value "presentation restricted" in the presentation
indicator. Based on local policy, the gateway MAY use the presence
of other priv-values to set the presentation indicator to
"presentation restricted". Otherwise the gateway SHALL include value
"presentation allowed" if a number is present or "not available due
to interworking" if no number is present.
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If the resulting Calling party number information element contains no
number and contains value "not available due to interworking" in the
presentation indicator, the gateway MAY omit the information element
from the QSIG SETUP message.
9.2.3. Generating the QSIG Connected Number Information Element
When mapping a SIP 2xx response to an INVITE request to a QSIG
CONNECT message, the gateway SHALL generate a Connected number
information element as follows.
If the SIP 2xx response contains an S/MIME signed message/sipfrag
[20] body containing a To header and the gateway supports this
capability and can verify the authenticity and trustworthiness of
this information, the gateway SHALL attempt to derive a number from
the URI in that header. If no number is derived from a
message/sipfrag body, if the SIP 2xx response contains a
P-Asserted-Identity header, and if the gateway supports that header
and trusts the information therein, the gateway SHALL attempt to
derive a number from the URI in that header. If a number is derived
from one of these headers, the gateway SHALL include it in the
Connected number information element and include value "network
provided" in the screening indicator.
If no number is derivable as described above, the gateway SHOULD NOT
include a number in the Connected number information element.
If the SIP 2xx response contains a Privacy header with value "id" in
parameter priv-value and the gateway supports this header, the
gateway SHALL include value "presentation restricted" in the
presentation indicator. Based on local policy, the gateway MAY use
the presence of other priv-values to set the presentation indicator
to "presentation restricted". Otherwise, the gateway SHALL include
value "presentation allowed" if a number is present or "not available
due to interworking" if no number is present.
If the resulting Connected number information element contains no
number and value "not available due to interworking" in the
presentation indicator, the gateway MAY omit the information element
from the QSIG CONNECT message.
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10. Requirements for Support of Basic Services
This document specifies signalling interworking for basic services
that provide a bi-directional transfer capability for speech,
facsimile, and modem media between the two networks.
10.1. Derivation of QSIG Bearer Capability Information Element
The gateway SHALL generate the Bearer Capability Information Element
in the QSIG SETUP message based on SDP offer information received
along with the SIP INVITE request. If the SIP INVITE request does
not contain SDP offer information or the media type in the SDP offer
information is only 'audio', then the Bearer capability information
element SHALL BE generated according to Table 3. Coding of the
Bearer capability information element for other media types is
outside the scope of this specification.
In addition, the gateway MAY include a Low layer compatibility
information element and/or High layer compatibility information in
the QSIG SETUP message if the gateway is able to derive relevant
information from the SDP offer information. Specific mappings are
outside the scope of this specification.
Table 3: Bearer capability encoding for 'audio' transfer
Field Value
-----------------------------------------------------------------
Coding Standard "CCITT standardized coding" (00)
Information transfer "3,1 kHz audio" (10000)
capability
Transfer mode "circuit mode" (00)
Information transfer rate "64 Kbits/s" (10000)
Multiplier Octet omitted
User information layer 1 Generated by gateway based on
protocol Information of the PISN. Supported
values are
"CCITT recommendation G.711 mu-law"
(00010)
"CCITT recommendation G.711 A-law"
(00011)
10.2. Derivation of Media Type in SDP
The gateway SHALL generate SDP offer information to include in the
SIP INVITE request based on information in the QSIG SETUP message.
The gateway MAY take account of QSIG Low layer compatibility and/or
High layer compatibility information elements, if present in the QSIG
SETUP message, when deriving SDP offer information, in which case
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specific mappings are outside the scope of this specification.
Otherwise, the gateway shall generate SDP offer information based
only on the Bearer capability information element in the QSIG SETUP
message, in which case the media type SHALL be derived according to
Table 4.
Table 4: Media type setting in SDP based on Bearer capability
information element
Information transfer capability in Media type in SDP
Bearer capability information element
---------------------------------------------------------------
"speech" (00000) audio
"3,1 kHz audio" (10000) audio
11. Security Considerations
11.1. General
Normal considerations apply for UA use of SIP security measures,
including digest authentication, TLS, and S/MIME as described in
[10].
The translation of QSIG information elements into SIP headers can
introduce some privacy and security concerns. For example, care
needs to be taken to provide adequate privacy for a user requesting
presentation restriction if the Calling party number information
element is openly mapped to the From header. Procedures for dealing
with this particular situation are specified in Section 9.1.2.
However, since the mapping specified in this document is mainly
concerned with translating information elements into the headers and
fields used to route SIP requests, gateways consequently reveal
(through this translation process) the minimum possible amount of
information.
There are some concerns, however, that arise from the other direction
of mapping, the mapping of SIP headers to QSIG information elements,
which are enumerated in the following paragraphs.
11.2. Calls from QSIG to Invalid or Restricted Numbers
When end users dial numbers in a PISN, their selections populate the
Called party number information element in the QSIG SETUP message.
Similarly, the SIP URI or tel URL and its optional parameters in the
Request-URI of a SIP INVITE request, which can be created directly by
end users of a SIP device, map to that information element at a
gateway. However, in a PISN, policy can prevent the user from
dialing certain (invalid or restricted) numbers. Thus, gateway
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implementers may wish to provide a means for gateway administrators
to apply policies restricting the use of certain SIP URIs or tel
URLs, or SIP URI or tel URL parameters, when authorizing a call from
SIP to QSIG.
11.3. Abuse of SIP Response Code
Some additional risks may result from the mapping of SIP response
codes to QSIG cause values. SIP user agents could conceivably
respond to an INVITE request from a gateway with any arbitrary SIP
response code, and thus they can dictate (within the boundaries of
the mappings supported by the gateway) the Q.850 cause code that will
be sent by the gateway in the resulting QSIG call clearing message.
Generally speaking, the manner in which a call is rejected is
unlikely to provide any avenue for fraud or denial of service (e.g.,
by signalling that a call should not be billed, or that the network
should take critical resources off-line). However, gateway
implementers may wish to make provision for gateway administrators to
modify the response code to cause value mappings to avoid any
undesirable network-specific behaviour resulting from the mappings
recommended in Section 8.4.4.
11.4. Use of the To Header URI
This specification requires the gateway to map the Request-URI rather
than the To header in a SIP INVITE request to the Called party number
information element in a QSIG SETUP message. Although a SIP UA is
expected to put the same URI in the To header and in the Request-URI,
this is not policed by other SIP entities. Therefore, a To header
URI that differs from the Request-URI received at the gateway cannot
be used as a reliable indication that the call has been re-targeted
in the SIP network or as a reliable indication of the original
target. Gateway implementers making use of the To header for mapping
to QSIG elements (e.g., as part of QSIG call diversion signalling)
may wish to make provision for disabling this mapping when deployed
in situations where the reliability of the QSIG elements concerned is
important.
11.5. Use of the From Header URI
The arbitrary population of the From header of requests by SIP user
agents has some well-understood security implications for devices
that rely on the From header as an accurate representation of the
identity of the originator. Any gateway that intends to use an
unsigned or unverified From header to populate the Calling party
number information element of a QSIG SETUP message should
authenticate the originator of the request and make sure that it is
authorized to assert that calling number (or make use of some more
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secure method to ascertain the identity of the caller). Note that
gateways, like all other SIP user agents, MUST support Digest
authentication as described in [10]. Similar considerations apply to
the use of the SIP P-Asserted-Identity header for mapping to the QSIG
Calling party number or Connected number information element, i.e.,
the source of this information should be authenticated. Use of a
signed message/sipfrag body to derive a QSIG Calling party number or
Connected number information element is another secure alternative.
11.6. Abuse of Early Media
There is another class of potential risk that is related to the cut-
through of the backwards media path before the call is answered.
Several practices described in this document involve the connection
of media streams to user information channels on inter-PINX links and
the sending of progress description number 1 or 8 in a backward QSIG
message. This can result in media being cut through end-to-end, and
it is possible for the called user agent then to play arbitrary audio
to the caller for an indefinite period of time before transmitting a
final response (in the form of a 2xx or higher response code) to an
INVITE request. This is useful since it also permits network
entities (particularly legacy networks that are incapable of
transmitting Q.850 cause values) to play tones and announcements to
indicate call failure or call progress, without triggering charging
by transmitting a 2xx response. Also, early cut-through can help
prevent clipping of the initial media when the call is answered.
There are conceivable respects in which this capability could be used
fraudulently by the called user agent for transmitting arbitrary
information without answering the call or before answering the call.
However, in corporate networks, charging is often not an issue, and
for calls arriving at a corporate network from a carrier network, the
carrier network normally takes steps to prevent fraud.
The usefulness of this capability appears to outweigh any risks
involved, which may in practice be no greater than in existing
PISN/ISDN environments. However, gateway implementers may wish to
make provision for gateway administrators to turn off cut-through or
minimise its impact (e.g., by imposing a time limit) when deployed in
situations where problems can arise.
11.7. Protection from Denial-of-Service Attacks
Unlike a traditional PISN phone, a SIP user agent can launch multiple
simultaneous requests in order to reach a particular resource. It
would be trivial for a SIP user agent to launch 100 SIP INVITE
requests at a 100 port gateway, thereby tying up all of its ports. A
malicious user could choose to launch requests to telephone numbers
that are known never to answer, or, where overlap signalling is used,
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to incomplete addresses. This could saturate resources at the
gateway indefinitely, potentially without incurring any charges.
Gateway implementers may therefore wish to provide means of
restricting according to policy the number of simultaneous requests
originating from the same authenticated source, or similar mechanisms
to address this possible denial-of-service attack.
12. Acknowledgements
This document is a product of the authors' activities in Ecma
(www.ecma-international.org) on interoperability of QSIG with IP
networks. An earlier version is published as Standard ECMA-339.
Ecma has made this work available to the IETF as the basis for
publishing an RFC.
The authors wish to acknowledge the assistance of Francois Audet,
Adam Roach, Jean-Francois Rey, Thomas Stach, and members of Ecma
TC32-TG17 in preparing and commenting on this document.
13. Normative References
[1] International Standard ISO/IEC 11571 "Private Integrated
Services Networks (PISN) - Addressing" (also published by Ecma
as Standard ECMA-155).
[2] International Standard ISO/IEC 11572 "Private Integrated
Services Network - Circuit-mode Bearer Services - Inter-Exchange
Signalling Procedures and Protocol" (also published by Ecma as
Standard ECMA-143).
[3] International Standard ISO/IEC 11582 "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).
[4] Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997.
[5] Postel, J., "Transmission Control Protocol", STD 7, RFC 793,
September 1981.
[6] Postel, J., "User Datagram Protocol", STD 6, RFC 768, August
1980.
[7] Dierks, T. and C. Allen, "The TLS Protocol Version 1.0", RFC
2246, January 1999.
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[8] Handley, M. and V. Jacobson, "SDP: Session Description
Protocol", RFC 2327, April 1998.
[9] Stewart, R., Xie, Q., Morneault, K., Sharp, C., Schwarzbauer,
H., Taylor, T., Rytina, I., Kalla, M., Zhang, L., and V. Paxson,
"Stream Control Transmission Protocol", RFC 2960, October 2000.
[10] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A.,
Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP:
Session Initiation Protocol", RFC 3261, June 2002.
[11] Rosenberg, J. and H. Schulzrinne, "Reliability of Provisional
Responses in Session Initiation Protocol (SIP)", RFC 3262, June
2002.
[12] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
Session Description Protocol (SDP)", RFC 3264, June 2002.
[13] Peterson, J., "A Privacy Mechanism for the Session Initiation
Protocol (SIP)", RFC 3323, November 2002.
[14] Jennings, C., Peterson, J., and M. Watson, "Private Extensions
to the Session Initiation Protocol (SIP) for Asserted Identity
within Trusted Networks", RFC 3325, November 2002.
[15] Postel, J., "Internet Protocol", STD 5, RFC 791, September 1981.
[16] Deering, S. and R. Hinden, "Internet Protocol, Version 6 (IPv6)
Specification", RFC 2460, December 1998.
[17] ITU-T Recommendation E.164, "The International Public
Telecommunication Numbering Plan", (1997-05).
[18] Camarillo, G., Roach, A., Peterson, J., and L. Ong, "Mapping of
Integrated Services Digital Network (ISDN) User Part (ISUP)
Overlap Signalling to the Session Initiation Protocol (SIP)",
RFC 3578, August 2003.
[19] Rosenberg, J., "The Session Initiation Protocol (SIP) UPDATE
Method", RFC 3311, October 2002.
[20] Sparks, R., "Internet Media Type message/sipfrag", RFC 3420,
November 2002.
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Appendix A. Example Message Sequences
A.1. Introduction
This appendix shows some typical message sequences that can occur for
an interworking between QSIG and SIP. It is informative.
NOTE: For all message sequence diagrams, there is no message mapping
between QSIG and SIP unless explicitly indicated by dotted lines.
Also, if there are no dotted lines connecting two messages, this
means that these are independent of each other in terms of the time
when they occur.
NOTE: Numbers prefixing SIP method names and response codes in the
diagrams represent sequence numbers. Messages bearing the same
number will have the same value in the CSeq header.
NOTE: In these examples, SIP provisional responses (other than 100)
are shown as being sent reliably, using the PRACK method for
acknowledgement.
A.2. Message Sequences for Call Establishment from QSIG to SIP
Below are typical message sequences for successful call establishment
from QSIG to SIP
Elwell, et al. Best Current Practice [Page 45]
RFC 4497 Interworking between SIP and QSIG May 2006
A.2.1. QSIG to SIP, using en bloc procedures on both QSIG and SIP
+-------------------+
| |
| GATEWAY |
PISN | | IP NETWORK
| +-----+------+------+ |
| | | |
| | | |
| QSIG SETUP | | 1-INVITE |
1|----------------------->|......|----------------------->| 2
| | | |
| | | |
| QSIG CALL PROCEEDING | | 1-100 TRYING |
3|<-----------------------| |<-----------------------+ 4
| | | |
| | | |
| QSIG ALERTING | | 1-180 RINGING |
8|<-----------------------|......|<-----------------------+ 5
| | | |
| | | 2-PRACK |
| | |----------------------->| 6
| | | 2-200 OK |
| | |<-----------------------+ 7
| | | |
| QSIG CONNECT | | 1-200 OK |
11|<-----------------------|......|<-----------------------+ 9
| | | |
| QSIG CONNECT ACK | | 1-ACK |
12|----------------------->| |----------------------->| 10
| | | |
|<======================>| |<======================>|
| AUDIO | | AUDIO |
Figure 3: Typical message sequence for successful call establishment
from QSIG to SIP, using en bloc procedures on both QSIG and SIP
1 The PISN sends a QSIG SETUP message to the gateway to begin a
session with a SIP UA.
2 On receipt of the QSIG SETUP message, the gateway generates a SIP
INVITE request and sends it to an appropriate SIP entity in the IP
network based on the called number.
3 The gateway sends a QSIG CALL PROCEEDING message to the PISN; no
more QSIG INFORMATION messages will be accepted.
4 The IP network sends a SIP 100 (Trying) response to the gateway.
5 The IP network sends a SIP 180 (Ringing) response.
Elwell, et al. Best Current Practice [Page 46]
RFC 4497 Interworking between SIP and QSIG May 2006
6 The gateway may send back a SIP PRACK request to the IP network
based on the inclusion of a Require header or a Supported header
with option tag 100rel in the initial SIP INVITE request.
7 The IP network sends a SIP 200 (OK) response to the gateway to
acknowledge the SIP PRACK request
8 The gateway maps this SIP 180 (Ringing) response to a QSIG
ALERTING message and sends it to the PISN.
9 The IP network sends a SIP 200 (OK) response when the call is
answered.
10 The gateway sends a SIP ACK request to acknowledge the SIP 200
(OK) response.
11 The gateway maps this SIP 200 (OK) response to a QSIG CONNECT
message and sends it to the PISN.
12 The PISN sends a QSIG CONNECT ACKNOWLEDGE message in response to
the QSIG CONNECT message.
Elwell, et al. Best Current Practice [Page 47]
RFC 4497 Interworking between SIP and QSIG May 2006
A.2.2. QSIG to SIP, using overlap receiving on QSIG and en bloc sending
on SIP
+------------------------+
PISN | GATEWAY | IP NETWORK
| |
| QSIG SETUP +--------+-------+-------+ |
1|-------------------------->| | |
| | | |
| QSIG SETUP ACK | | |
2|<--------------------------| | |
| | | |
| QSIG INFORMATION | | |
3|-------------------------->| | |
| | | |
| QSIG INFORMATION | | 1-INVITE |
3a|-------------------------->|.......|----------------------->|4
| QSIG CALL PROCEEDING | | 1-100 TRYING |
5|<--------------------------| |<-----------------------|6
| | | |
| QSIG ALERTING | | 1-180 RINGING |
10|<--------------------------|.......|<-----------------------|7
| | | 2-PRACK |
| | |----------------------->|8
| | | 2-200 OK |
| | |<-----------------------|9
| QSIG CONNECT | | 1-200 OK |
13|<--------------------------|.......|<-----------------------|11
| | | |
| QSIG CONNECT ACK | | 1-ACK |
14|-------------------------->| |----------------------->|12
| AUDIO | | AUDIO |
|<=========================>| |<======================>|
Figure 4: Typical message sequence for successful call establishment
from QSIG to SIP, using overlap receiving on QSIG and en bloc sending
on SIP
1 The PISN sends a QSIG SETUP message to the gateway to begin a
session with a SIP UA. The QSIG SETUP message does not contain a
Sending Complete information element.
2 The gateway sends a QSIG SETUP ACKNOWLEDGE message to the PISN.
More digits are expected.
3 More digits are sent from the PISN within a QSIG INFORMATION
message.
3a More digits are sent from the PISN within a QSIG INFORMATION
message. The QSIG INFORMATION message contains a Sending Complete
information element.
Elwell, et al. Best Current Practice [Page 48]
RFC 4497 Interworking between SIP and QSIG May 2006
4 The Gateway generates a SIP INVITE request and sends it to an
appropriate SIP entity in the IP network, based on the called
number.
5 The gateway sends a QSIG CALL PROCEEDING message to the PISN; no
more QSIG INFORMATION messages will be accepted.
6 The IP network sends a SIP 100 (Trying) response to the gateway.
7 The IP network sends a SIP 180 (Ringing) response.
8 The gateway may send back a SIP PRACK request to the IP network
based on the inclusion of a Require header or a Supported header
with option tag 100rel in the initial SIP INVITE request.
9 The IP network sends a SIP 200 (OK) response to the gateway to
acknowledge the SIP PRACK request.
10 The gateway maps this SIP 180 (Ringing) response to a QSIG
ALERTING message and sends it to the PINX.
11 The IP network sends a SIP 200 (OK) response when the call is
answered.
12 The gateway sends an SIP ACK request to acknowledge the SIP 200
(OK) response.
13 The gateway maps this SIP 200 (OK) response to a QSIG CONNECT
message and sends it to the PINX.
14 The PISN sends a QSIG CONNECT ACKNOWLEDGE message in response to
the QSIG CONNECT message.
Elwell, et al. Best Current Practice [Page 49]
RFC 4497 Interworking between SIP and QSIG May 2006
A.2.3. QSIG to SIP, using overlap procedures on both QSIG and SIP
+----------------------+
PISN | GATEWAY | IP NETWORK
| |
| QSIG SETUP +-------+-------+------+ |
1 |------------------------->| | |
| | | |
| QSIG SETUP ACK | | |
2 |<-------------------------| | |
| | | |
| QSIG INFORMATION | | |
3 |------------------------->| | |
| QSIG INFORMATION | | 1-INVITE |
3 |------------------------->|.......|------------------------>|4
| | | 1-484 |
| | |<------------------------|5
| | | 1-ACK |
| | |------------------------>|6
| QSIG INFORMATION | | 2-INVITE |
7 |------------------------->|.......|------------------------>|4
| | | 2-484 |
| | |<------------------------|5
| | | 2-ACK |
| | |------------------------>|6
| | | |
| QSIG INFORMATION | | |
| Sending Complete IE | | 3-INVITE |
8 |------------------------->|.......|------------------------>|10
| QSIG CALL PROCEEDING | | 3-100 TRYING |
9 |<-------------------------| |<------------------------|11
| | | |
| QSIG ALERTING | | 3-180 RINGING |
15|<-------------------------|.......|<------------------------|12
| | | 4-PRACK |
| | |------------------------>|13
| | | 4-200 OK |
| | |<------------------------|14
| QSIG CONNECT | | 3-200 OK |
18|<-------------------------|.......|<------------------------|16
| | | |
| QSIG CONNECT ACK | | 3-ACK |
19|------------------------->| |------------------------>|17
| AUDIO | | AUDIO |
|<========================>| |<=======================>|
| | | |
Elwell, et al. Best Current Practice [Page 50]
RFC 4497 Interworking between SIP and QSIG May 2006
Figure 5: Typical message sequence for successful call establishment
from QSIG to SIP, using overlap procedures on both QSIG and SIP
1 The PISN sends a QSIG SETUP message to the gateway to begin a
session with a SIP UA. The QSIG SETUP message does not contain a
Sending complete information element.
2 The gateway sends a QSIG SETUP ACKNOWLEDGE message to the PISN.
More digits are expected.
3 More digits are sent from the PISN within a QSIG INFORMATION
message.
4 When the gateway receives the minimum number of digits required to
route the call, it generates a SIP INVITE request and sends it to
an appropriate SIP entity in the IP network based on the called
number
5 Due to an insufficient number of digits, the IP network will
return a SIP 484 (Address Incomplete) response.
6 The SIP 484 (Address Incomplete) response is acknowledged.
7 More digits are received from the PISN in a QSIG INFORMATION
message. A new INVITE is sent with the same Call-ID and From
values but an updated Request-URI.
8 More digits are received from the PISN in a QSIG INFORMATION
message. The QSIG INFORMATION message contains a Sending Complete
information element.
9 The gateway sends a QSIG CALL PROCEEDING message to the PISN; no
more information will be accepted.
10 The gateway sends a new SIP INVITE request with an updated
Request-URI field.
11 The IP network sends a SIP 100 (Trying) response to the gateway.
12 The IP network sends a SIP 180 (Ringing) response.
13 The gateway may send back a SIP PRACK request to the IP network
based on the inclusion of a Require header or a Supported header
with option tag 100rel in the initial SIP INVITE request.
14 The IP network sends a SIP 200 (OK) response to the gateway to
acknowledge the SIP PRACK request.
15 The gateway maps this SIP 180 (Ringing) response to a QSIG
ALERTING message and sends it to the PISN.
16 The IP network sends a SIP 200 (OK) response when the call is
answered.
17 The gateway sends a SIP ACK request to acknowledge the SIP 200
(OK) response.
18 The gateway maps this SIP 200 (OK) response to a QSIG CONNECT
message.
19 The PISN sends a QSIG CONNECT ACKNOWLEDGE message in response to
the QSIG CONNECT message.
Elwell, et al. Best Current Practice [Page 51]
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A.3. Message sequences for call establishment from SIP to QSIG
Below are typical message sequences for successful call establishment
from SIP to QSIG
A.3.1. SIP to QSIG, using en bloc procedures
+----------------------+
IP NETWORK | GATEWAY | PISN
| |
| +-------+-------+------+ |
| | | |
| | | |
| 1-INVITE | | QSIG SETUP |
1 |------------------------->|.......|------------------------>|3
| 1-100 TRYING | | QSIG CALL PROCEEDING |
2 |<-------------------------| |<------------------------|4
| 1-180 RINGING | | QSIG ALERTING |
6 |<-------------------------|.......|<------------------------|5
| | | |
| | | |
| 2-PRACK | | |
7 |------------------------->| | |
| 2-200 OK | | |
8 |<-------------------------| | |
| 1-200 OK | | QSIG CONNECT |
11|<-------------------------|.......|<------------------------|9
| | | |
| 1-ACK | | QSIG CONNECT ACK |
12|------------------------->| |------------------------>|10
| AUDIO | | AUDIO |
|<========================>| |<=======================>|
| | | |
Figure 6: Typical message sequence for successful call establishment
from SIP to QSIG, using en bloc procedures
1 The IP network sends a SIP INVITE request to the gateway.
2 The gateway sends a SIP 100 (Trying) response to the IP network.
3 On receipt of the SIP INVITE request, the gateway sends a QSIG
SETUP message.
4 The PISN sends a QSIG CALL PROCEEDING message to the gateway.
5 A QSIG ALERTING message is returned to indicate that the end user
in the PISN is being alerted.
6 The gateway maps the QSIG ALERTING message to a SIP 180 (Ringing)
response.
Elwell, et al. Best Current Practice [Page 52]
RFC 4497 Interworking between SIP and QSIG May 2006
7 The IP network can send back a SIP PRACK request to the IP network
based on the inclusion of a Require header or a Supported header
with option tag 100rel in the initial SIP INVITE request.
8 The gateway sends a SIP 200 (OK) response to acknowledge the SIP
PRACK request.
9 The PISN sends a QSIG CONNECT message to the gateway when the call
is answered.
10 The gateway sends a QSIG CONNECT ACKNOWLEDGE message to
acknowledge the QSIG CONNECT message.
11 The QSIG CONNECT message is mapped to a SIP 200 (OK) response.
12 The IP network, upon receiving a SIP INVITE final response (200),
will send a SIP ACK request to acknowledge receipt.
Elwell, et al. Best Current Practice [Page 53]
RFC 4497 Interworking between SIP and QSIG May 2006
A.3.2. SIP to QSIG, using overlap receiving on SIP and en bloc sending
on QSIG
+----------------------+
IP NETWORK | GATEWAY | PISN
| |
| 1-INVITE +-------+-------+------+ |
1 |------------------------->| | |
| 1-484 | | |
2 |<-------------------------| | |
| 1-ACK | | |
3 |------------------------->| | |
| 2-INVITE | | |
1 |------------------------->| | |
| 2-484 | | |
2 |<-------------------------| | |
| 2- ACK | | |
3 |------------------------->| | |
| 3-INVITE | | QSIG SETUP |
4 |------------------------->|.......|------------------------>|6
| 3-100 TRYING | | QSIG CALL PROCEEDING |
5 |<-------------------------| |<------------------------|7
| 3-180 RINGING | | QSIG ALERTING |
9 |<-------------------------|.......|<------------------------|8
| | | |
| | | |
| 4-PRACK | | |
10|------------------------->| | |
| 4-200 OK | | |
11|<-------------------------| | |
| 3-200 OK | | QSIG CONNECT |
14|<-------------------------|.......|<------------------------|12
| | | |
| 3-ACK | | QSIG CONNECT ACK |
15|------------------------->| |------------------------>|13
| AUDIO | | AUDIO |
|<========================>| |<=======================>|
| | | |
Figure 7: Typical message sequence for successful call establishment
from SIP to QSIG, using overlap receiving on SIP and en bloc sending
on QSIG
1 The IP network sends a SIP INVITE request to the gateway.
2 Due to an insufficient number of digits, the gateway returns a SIP
484 (Address Incomplete) response.
3 The IP network acknowledges the SIP 484 (Address Incomplete)
response.
Elwell, et al. Best Current Practice [Page 54]
RFC 4497 Interworking between SIP and QSIG May 2006
4 The IP network sends a new SIP INVITE request with the same Call-
ID and updated Request-URI.
5 The gateway now has all the digits required to route the call to
the PISN. The gateway sends back a SIP 100 (Trying) response.
6 The gateway sends a QSIG SETUP message.
7 The PISN sends a QSIG CALL PROCEEDING message to the gateway.
8 A QSIG ALERTING message is returned to indicate that the end user
in the PISN is being alerted.
9 The gateway maps the QSIG ALERTING message to a SIP 180 (Ringing)
response.
10 The IP network can send back a SIP PRACK request to the IP network
based on the inclusion of a Require header or a Supported header
with option tag 100rel in the initial SIP INVITE request.
11 The gateway sends a SIP 200 (OK) response to acknowledge the SIP
PRACK request.
12 The PISN sends a QSIG CONNECT message to the gateway when the call
is answered.
13 The gateway sends a QSIG CONNECT ACKNOWLEDGE message to
acknowledge the CONNECT message.
14 The QSIG CONNECT message is mapped to a SIP 200 (OK) response.
15 The IP network, upon receiving a SIP INVITE final response (200),
will send a SIP ACK request to acknowledge receipt.
Elwell, et al. Best Current Practice [Page 55]
RFC 4497 Interworking between SIP and QSIG May 2006
A.3.3. SIP to QSIG, using overlap procedures on both SIP and QSIG
+----------------------+
IP NETWORK | GATEWAY | PISN
| |
| 1-INVITE +-------+-------+------+ |
1 |------------------------->| | |
| 1-484 | | |
2 |<-------------------------| | |
| 1-ACK | | |
3 |------------------------->| | |
| 2-INVITE | | QSIG SETUP |
4 |------------------------->|.......|------------------------>|6
| 2-100 TRYING | | QSIG SETUP ACK |
5 |<-------------------------| |<------------------------|7
| 3- INVITE | | QSIG INFORMATION |
8 |------------------------->|.......|------------------------>|10
| 3-100 TRYING | | |
9 |<-------------------------| | QSIG CALL PROCEEDING |
| | |<------------------------|11
13| 3-180 RINGING | | QSIG ALERTING |
|<-------------------------|.......|<------------------------|12
| 2-484 | | |
14|<-------------------------| | |
| 2-ACK | | |
15|------------------------->| | |
| 4-PRACK | | |
16|------------------------->| | |
| 4-200 OK | | |
17|<-------------------------| | |
| 3-200 OK | | QSIG CONNECT |
20|<-------------------------|.......|<------------------------|18
| | | |
| 3-ACK | | QSIG CONNECT ACK |
21|------------------------->| |------------------------>|19
| AUDIO | | AUDIO |
|<========================>| |<=======================>|
| | | |
Figure 8: Typical message sequence for successful call establishment
from SIP to QSIG, using overlap procedures on both SIP and QSIG
1 The IP network sends a SIP INVITE request to the gateway.
2 Due to an insufficient number of digits, the gateway returns a SIP
484 (Address Incomplete) response.
3 The IP network acknowledges the SIP 484 (Address Incomplete)
response.
Elwell, et al. Best Current Practice [Page 56]
RFC 4497 Interworking between SIP and QSIG May 2006
4 The IP network sends a new SIP INVITE request with the same
Call-ID and updated Request-URI.
5 The gateway now has all the digits required to route the call to
the PISN. The gateway sends back a SIP 100 (Trying) response to
the IP network.
6 The gateway sends a QSIG SETUP message.
7 The PISN needs more digits to route the call and sends a QSIG
SETUP ACKNOWLEDGE message to the gateway.
8 The IP network sends a new SIP INVITE request with the same
Call-ID and From values and updated Request-URI.
9 The gateway sends back a SIP 100 (Trying) response to the IP
network.
10 The gateway maps the new SIP INVITE request to a QSIG INFORMATION
message.
11 The PISN has all the digits required and sends back a QSIG CALL
PROCEEDING message to the gateway.
12 A QSIG ALERTING message is returned to indicate that the end user
in the PISN is being alerted.
13 The gateway maps the QSIG ALERTING message to a SIP 180 (Ringing)
response.
14 The gateway sends a SIP 484 (Address Incomplete) response for the
previous SIP INVITE request.
15 The IP network acknowledges the SIP 484 (Address Incomplete)
response.
16 The IP network can send back a SIP PRACK request to the IP network
based on the inclusion of a Require header or a Supported header
with option tag 100rel in the initial SIP INVITE request.
17 The gateway sends a SIP 200 (OK) response to acknowledge the SIP
PRACK request.
18 The PISN sends a QSIG CONNECT message to the gateway when the call
is answered.
19 The gateway sends a QSIG CONNECT ACKNOWLEDGE message to
acknowledge the QSIG CONNECT message.
20 The QSIG CONNECT message is mapped to a SIP 200 (OK) response.
21 The IP network, upon receiving a SIP INVITE final response (200),
will send a SIP ACK request to acknowledge receipt.
Elwell, et al. Best Current Practice [Page 57]
RFC 4497 Interworking between SIP and QSIG May 2006
A.4. Message Sequence for Call Clearing from QSIG to SIP
Below are typical message sequences for Call Clearing from QSIG to
SIP
A.4.1. QSIG to SIP, subsequent to call establishment
+-------------------+
| |
| GATEWAY |
PISN | | IP NETWORK
| +-----+------+------+ |
| | | |
| | | |
| QSIG DISCONNECT | | 2- BYE |
1|----------------------->|......|----------------------->|4
| QSIG RELEASE | | 2-200 OK |
2|<-----------------------| |<-----------------------|5
| QSIG RELEASE COMP | | |
3|----------------------->| | |
| | | |
| | | |
| | | |
Figure 9: Typical message sequence for call clearing from QSIG to
SIP, subsequent to call establishment
1 The PISN sends a QSIG DISCONNECT message to the gateway.
2 The gateway sends back a QSIG RELEASE message to the PISN in
response to the QSIG DISCONNECT message.
3 The PISN sends a QSIG RELEASE COMPLETE message in response. All
PISN resources are now released.
4 The gateway maps the QSIG DISCONNECT message to a SIP BYE request.
5 The IP network sends back a SIP 200 (OK) response to the SIP BYE
request. All IP resources are now released.
Elwell, et al. Best Current Practice [Page 58]
RFC 4497 Interworking between SIP and QSIG May 2006
A.4.2. QSIG to SIP, during establishment of a call from SIP to QSIG
+-------------------+
| |
| GATEWAY |
PISN | | IP NETWORK
| +-----+------+------+ |
| | | |
| | | |
| QSIG DISCONNECT | | 1- 4XX / 6XX |
1|----------------------->|......|---------------------->|4
| QSIG RELEASE | | 1- ACK |
2|<-----------------------| |<----------------------|5
| QSIG RELEASE COMP | | |
3|----------------------->| | |
| | | |
| | | |
Figure 10: Typical message sequence for call clearing from QSIG to
SIP, during establishment of a call from SIP to QSIG (gateway has
not sent a final response to the SIP INVITE request)
1 The PISN sends a QSIG DISCONNECT message to the gateway
2 The gateway sends back a QSIG RELEASE message to the PISN in
response to the QSIG DISCONNECT message
3 The PISN sends a QSIG RELEASE COMPLETE message in response. All
PISN resources are now released.
4 The gateway maps the QSIG DISCONNECT message to a SIP 4xx-6xx
response
5 The IP network sends back a SIP ACK request in response to the SIP
4xx-6xx response. All IP resources are now released
Elwell, et al. Best Current Practice [Page 59]
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A.4.3. QSIG to SIP, during establishment of a call from QSIG to SIP
+-------------------+
| |
| GATEWAY |
PISN | | IP NETWORK
| +-----+------+------+ |
| | | |
| | | |
| QSIG DISCONNECT | | 1- CANCEL |
1|----------------------->|......|----------------------->|4
| QSIG RELEASE | |1-487 Request Terminated|
2|<-----------------------| |<-----------------------|5
| QSIG RELEASE COMP | | |
3|----------------------->| | 1- ACK |
| | |----------------------->|6
| | | |
| | | 1- 200 OK |
| | |<-----------------------|7
| | | |
Figure 11: Typical message sequence for call clearing from QSIG to
SIP, during establishment of a call from QSIG to SIP (gateway has
received a provisional response to the SIP INVITE request but not a
final response)
1 The PISN sends a QSIG DISCONNECT message to the gateway.
2 The gateway sends back a QSIG RELEASE message to the PISN in
response to the QSIG DISCONNECT message.
3 The PISN sends a QSIG RELEASE COMPLETE message in response. All
PISN resources are now released.
4 The gateway maps the QSIG DISCONNECT message to a SIP CANCEL
request (subject to receipt of a provisional response, but not of
a final response).
5 The IP network sends back a SIP 487 (Request Terminated) response
to the SIP INVITE request.
6 The gateway, on receiving a SIP final response (487) to the SIP
INVITE request, sends back a SIP ACK request to acknowledge
receipt.
7 The IP network sends back a SIP 200 (OK) response to the SIP
CANCEL request. All IP resources are now released.
Elwell, et al. Best Current Practice [Page 60]
RFC 4497 Interworking between SIP and QSIG May 2006
A.5. Message Sequence for Call Clearing from SIP to QSIG
Below are typical message sequences for Call Clearing from SIP to
QSIG
A.5.1. SIP to QSIG, subsequent to call establishment
+-------------------+
| |
| GATEWAY |
IP NETWORK | | PISN
| +-----+------+------+ |
| | | |
| | | |
| 2- BYE | | QSIG DISCONNECT |
1|----------------------->|......|----------------------->|3
| | | QSIG RELEASE |
| | |<-----------------------|4
| 2-200 OK | | QSIG RELEASE COMP |
2|<-----------------------| |----------------------->|5
| | | |
| | | |
Figure 12: Typical message sequence for call clearing from SIP to
QSIG, subsequent to call establishment
1 The IP network sends a SIP BYE request to the gateway.
2 The gateway sends back a SIP 200 (OK) response to the SIP BYE
request. All IP resources are now released.
3 The gateway maps the SIP BYE request to a QSIG DISCONNECT message.
4 The PISN sends back a QSIG RELEASE message to the gateway in
response to the QSIG DISCONNECT message.
5 The gateway sends a QSIG RELEASE COMPLETE message in response.
All PISN resources are now released.
Elwell, et al. Best Current Practice [Page 61]
RFC 4497 Interworking between SIP and QSIG May 2006
A.5.2. SIP to QSIG, during establishment of a call from QSIG to SIP
+-------------------+
| |
| GATEWAY |
IP NETWORK | | PISN
| +-----+------+------+ |
| | | |
| | | |
| 1- 4XX / 6XX | | QSIG DISCONNECT |
1|----------------------->|......|----------------------->|3
| | | QSIG RELEASE |
| | |<-----------------------|4
| 1- ACK | | QSIG RELEASE COMP |
2|<-----------------------| |----------------------->|5
| | | |
| | | |
| | | |
Figure 13: Typical message sequence for call clearing from SIP to
QSIG, during establishment of a call from QSIG to SIP (gateway has
not previously received a final response to the SIP INVITE request)
1 The IP network sends a SIP 4xx-6xx response to the gateway.
2 The gateway sends back a SIP ACK request in response to the SIP
4xx-6xx response. All IP resources are now released.
3 The gateway maps the SIP 4xx-6xx response to a QSIG DISCONNECT
message.
4 The PISN sends back a QSIG RELEASE message to the gateway in
response to the QSIG DISCONNECT message.
5 The gateway sends a QSIG RELEASE COMPLETE message in response.
All PISN resources are now released.
Elwell, et al. Best Current Practice [Page 62]
RFC 4497 Interworking between SIP and QSIG May 2006
A.5.3. SIP to QSIG, during establishment of a call from SIP to QSIG
+-------------------+
| |
| GATEWAY |
IP NETWORK | | PISN
| +-----+------+------+ |
| | | |
| | | |
| 1- CANCEL | | QSIG DISCONNECT |
1|----------------------->|......|----------------------->|4
| | | QSIG RELEASE |
| | |<-----------------------|5
|1-487 Request Terminated| | QSIG RELEASE COMP |
2|<-----------------------| |----------------------->|6
| | | |
| 1- ACK | | |
3|----------------------->| | |
| | | |
| 1- 200 OK | | |
4|<-----------------------| | |
Figure 14: Typical message sequence for call clearing from SIP to
QSIG, during establishment of a call from SIP to QSIG (gateway has
sent a provisional response to the SIP INVITE request but not a final
response)
1 The IP network sends a SIP CANCEL request to the gateway.
2 The gateway sends back a SIP 487 (Request Terminated) response to
the SIP INVITE request.
3 The IP network, on receiving a SIP final response (487) to the SIP
INVITE request, sends back a SIP ACK request to acknowledge
receipt.
4 The gateway sends back a SIP 200 (OK) response to the SIP CANCEL
request. All IP resources are now released.
5 The gateway maps the SIP 4xx-6xx response to a QSIG DISCONNECT
message.
6 The PISN sends back a QSIG RELEASE message to the gateway in
response to the QSIG DISCONNECT message.
7 The gateway sends a QSIG RELEASE COMPLETE message in response.
All PISN resources are now released.
Elwell, et al. Best Current Practice [Page 63]
RFC 4497 Interworking between SIP and QSIG May 2006
Authors' Addresses
John Elwell
Siemens plc
Technology Drive
Beeston
Nottingham, UK, NG9 1LA
EMail: john.elwell@siemens.com
Frank Derks
NEC Philips Unified Solutions
Anton Philipsweg 1
1223 KZ Hilversum
The Netherlands
EMail: frank.derks@nec-philips.com
Olivier Rousseau
Alcatel Business Systems
32,Avenue Kleber
92700 Colombes
France
EMail: Olivier.Rousseau@alcatel.fr
Patrick Mourot
Alcatel Business Systems
1,Rue Dr A. Schweitzer
67400 Illkirch
France
EMail: Patrick.Mourot@alcatel.fr
Elwell, et al. Best Current Practice [Page 64]
RFC 4497 Interworking between SIP and QSIG May 2006
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Elwell, et al. Best Current Practice [Page 65]