Internet Engineering Task Force J. Elwell
Internet Draft Siemens
F. Derks
Philips
P. Mourot/O. Rousseau
draft-elwell-sipping-qsig2sip-03.txt Alcatel
Expires: April 2003 October 2002
Interworking between SIP and QSIG
Status of this Memo
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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.
As the support of telephony within corporate networks evolves from
circuit-switched technology to Internet technology, the two
technologies will co-exist 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
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network and a participant in the QSIG network. Such calls are
supported by gateways that perform interworking between SIP and QSIG.
This document is a product of the authors' activities in ECMA
(www.ecma.ch) on interoperability of QSIG with IP networks.
1 Introduction....................................................4
2 Terminology.....................................................5
3 Definitions.....................................................5
3.1 External definitions..........................................5
3.2 Other definitions.............................................5
3.2.1 Gateway.....................................................5
3.2.2 IP network..................................................5
3.2.3 Media stream................................................5
4 Acronyms........................................................6
5 Background and architecture.....................................6
6 Overview........................................................9
7 General requirements...........................................10
8 Message mapping requirements...................................11
8.1 Message validation and handling of protocol errors...........11
8.2 Call establishment from QSIG to SIP..........................12
8.2.1 Call establishment from QSIG to SIP using enbloc procedures12
8.2.1.1 Receipt of QSIG SETUP message............................12
8.2.1.2 Receipt of SIP 100 (Trying) response.....................13
8.2.1.3 Receipt of SIP 18x provisional response..................13
8.2.1.4 Receipt of SIP 2xx response..............................14
8.2.1.5 Receipt of SIP 3xx response..............................15
8.2.2 Call establishment from QSIG to SIP using overlap procedures15
8.2.2.1 Enbloc signalling in SIP network.........................15
8.2.2.1.1 Receipt of QSIG SETUP message..........................15
8.2.2.1.2 Receipt of QSIG INFORMATION message....................16
8.2.2.1.3 Receipt of SIP responses...............................16
8.2.2.2 Overlap signalling in SIP network........................16
8.2.2.2.1 Receipt of QSIG SETUP message..........................16
8.2.2.2.2 Receipt of QSIG INFORMATION message....................16
8.2.2.2.3 Receipt of SIP 100 (Trying) response...................17
8.2.2.2.4 Receipt of SIP 18x provisional response................17
8.2.2.2.5 Receipt of SIP 2xx response............................17
8.2.2.2.6 Receipt of SIP 3xx response............................17
8.2.2.2.7 Receipt of a SIP 484 (Address Incomplete) response.....17
8.2.2.2.8 Receipt of a SIP 4xx (except 484), 5xx or 6xx response.18
8.2.2.2.9 Receipt of multiple SIP responses......................18
8.2.2.2.10 Cancelling pending SIP INVITE transactions............18
8.2.2.2.11 SIP INVITE requests reaching multiple gateways........19
8.2.2.2.12 QSIG timer T302 expiry................................19
8.3 Call Establishment from SIP to QSIG..........................19
8.3.1 Receipt of SIP INVITE request for a new call...............19
8.3.2 Receipt of QSIG CALL PROCEEDING message....................20
8.3.3 Receipt of QSIG PROGRESS message...........................20
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8.3.4 Receipt of QSIG ALERTING message...........................21
8.3.5 Inclusion of SDP information in a SIP 18x provisional response
.................................................................22
8.3.6 Receipt of QSIG CONNECT message............................22
8.3.7 Receipt of SIP PRACK request...............................23
8.3.8 Receipt of SIP ACK request.................................24
8.3.9 Receipt of a SIP INVITE request for a call already being
established......................................................24
8.4 Call clearing and call failure...............................24
8.4.1 Receipt of a QSIG DISCONNECT, RELEASE or RELEASE COMPLETE
message..........................................................24
8.4.2 Receipt of a SIP BYE request...............................27
8.4.3 Receipt of a SIP CANCEL request............................27
8.4.4 Receipt of a SIP 4xx - 6xx response........................27
8.4.5 Gateway-initiated call clearing............................29
8.5 Request to change media characteristics......................30
9 Number mapping.................................................30
9.1 Mapping from QSIG to SIP.....................................30
9.1.1 Using information from the QSIG Called party number information
element..........................................................31
9.1.2 Using information from the QSIG Calling party number
information element..............................................31
9.1.2.1 No URI derived and presentation indicator does not have value
"presentation restricted"........................................31
9.1.2.2 No URI derived and presentation indicator has value
"presentation restricted"........................................31
9.1.2.3 URI derived and presentation indicator has value
"presentation restricted"........................................31
9.1.2.4 URI derived and presentation indicator does not have value
"presentation restricted"........................................32
9.1.3 Using information from the QSIG Connected number information
element..........................................................32
9.1.3.1 No URI derived and presentation indicator does not have value
"presentation restricted"........................................32
9.1.3.2 No URI derived and presentation indicator has value
"presentation restricted"........................................32
9.1.3.3 URI derived and presentation indicator has value
"presentation restricted"........................................32
9.1.3.4 URI derived and presentation indicator does not have value
"presentation restricted"........................................33
9.2 Mapping from SIP to QSIG.....................................33
9.2.1 Generating the QSIG Called party number information element33
9.2.2 Generating the QSIG Calling party number information element34
9.2.3 Generating the QSIG Connected number information element...34
10 Requirements for support of basic services....................35
10.1 Derivation of QSIG Bearer capability information element....35
10.2 Derivation of media type in SDP.............................36
11 Security considerations.......................................36
12 Author's Addresses............................................39
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13 Normative References..........................................39
Annex A û Example message sequences..............................41
Annex B û Change log.............................................54
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).
"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 ECMA-143 (call control in support of
basic services), ECMA-165 (generic functional protocol for the
support of supplementary services) and a number of Standards
specifying individual supplementary services.
SIP is an application layer protocol for establishing, terminating
and modifying multimedia sessions. It is typically carried over IP
[16], [17]. Telephone calls are considered as a type of multimedia
session where just audio is exchanged. SIP is defined in RFC 3261
[10].
This document specifies 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. 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.
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.).
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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 [2] and
indicate requirement levels for compliant SIP implementations.
3 Definitions
For the purposes of this specification, the following definitions
apply.
3.1 External definitions
This specification uses the following terms defined in other
documents:
-Call (ECMA-307)
-Corporate telecommunication network (CN) (ECMA-307)
-Private Integrated Services Network (PISN) (ECMA-307)
-Private Integrated services Network eXchange (PINX) (ECMA-133)
Additionally the definitions in ECMA-143 and RFC 3261 apply as
appropriate.
3.2 Other definitions
3.2.1 Gateway
An entity that performs interworking between a PISN using QSIG and an
IP network using SIP.
3.2.2 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.
3.2.3 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.
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NOTE. Normally a SIP session establishes a pair of media streams, one
in each direction.
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
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 24 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
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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 ITU-T recommendation E.164 [18] or
a private numbering plan. This basic call capability is specified in
ECMA-143. 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 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.
Another way of migrating is to use a SIP network to interconnect two
parts of a PISN and encapsulate QSIG signalling in SIP messages for
calls between the two parts of the PISN. 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 +--------| |
| | | | | |\
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| | | | +-----+ \
| | | | \ +------+
| | | | \| |
+--+---+ +--+---+ *-----------+ |PINX |
|SIP | |SIP | | |
|End- | |End- | +------+
|point | |point |
+------+ +------+
Figure 1 û Environment
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 information 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 (RFC 768) and TCP (RFC 761) as transport layer
protocols, these being used for the transport of SIP signalling
messages and, in the case of UDP, also for media information;
-optionally the support of TLS (RFC 2246) and/or SCTP (RFC 2960) 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 information in each direction between
the PISN and the IP network, including as a minimum packetization of
media information sent to the IP network and de-packetization of
media information received from the IP network.
NOTE. RFC 3261 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.
+---------------------------------------------------------+
| Inter-working function |
| |
+-----------------------+---------+-----------------------+
| | | |
| SIP | | |
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| | | |
+-----------------------+ | |
| | | |
| UDP/TCP/TLS/SCTP | | QSIG |
| | | |
+-----------------------+ | |
| | | |
| IP | | |
| | | |
+-----------------------+ +-----------------------+
| IP network | | PISN |
| lower layers | | lower layers |
| | | |
+-----------------------+ +-----------------------+
Figure 2 û 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 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 200 OK
response 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.
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.
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Annex 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 ECMA-143 as a gateway and SHALL support SIP
in accordance with RFC 3261 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 RFC 3261. In addition, the
gateway SHALL support SIP TU behaviour for a UA in accordance with
RFC 3261 except where stated otherwise in this specification.
NOTE 1. RFC 3261 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 RFC 3262 as a UA.
NOTE 2. RFC 3262 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 RFC 2327 and its use
in accordance with the offer / answer model in RFC 3264.
Section 9 also specifies optional use of the Privacy header in
accordance with RFC AAAA and the P-Asserted-Identity header in
accordance with RFC BBBB.
The gateway SHALL support calls from QSIG to SIP and calls from SIP
to QSIG.
SIP methods not defined in RFC 3261, RFC 3262, RFC AAAA or RFC BBBB
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 look-up 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 final
response SHOULD apply only when there are no more candidate
destinations to try.
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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 ECMA-143 and SHALL act in accordance with
ECMA-143 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.
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 RFC 3261 and SHALL
act in accordance with RFC 3261 on detection of a SIP protocol error.
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 8.2.2.2 of RFC 3261) 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
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-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 ECMA-143
and SHALL act in accordance with ECMA-143 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 8.4.5.
The gateway SHALL run SIP protocol timers as specified in RFC 3261
and SHALL act in accordance with RFC 3261 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 8.4.5.
8.2 Call establishment from QSIG to SIP
8.2.1 Call establishment from QSIG to SIP using enbloc 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. 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 the gateway cannot determine to 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.
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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 RFC 3262.
The gateway SHALL include SDP information in the SIP INVITE request
as described in section 10.
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 ôinvalid number format (address incomplete)ö.
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 ECMA-143.
8.2.1.2 Receipt of SIP 100 (Trying) response
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
The gateway SHALL map a received SIP 18x response 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
QSIG ALERTING message SHALL contain a Progress indicator information
element containing progress description number 8. If a media stream
has been established towards the gateway, the gateway SHALL connect
the media stream to the corresponding user information channel of the
inter-PINX link. If no media stream has been established towards the
gateway, the gateway SHALL supply ring-back tone on the user
information channel of the inter-PINX link.
-If a SIP 181/182/183 response is received, no QSIG ALERTING message
has been sent, no QSIG PROGRESS message containing progress
description number 8 has been sent and a media stream has been
established towards the gateway, the gateway SHALL generate a QSIG
PROGRESS message. The QSIG PROGRESS message SHALL contain progress
description number 8 in a Progress indicator information element. The
gateway SHALL also connect the media streams to the corresponding
user information channel of the inter-PINX link.
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-If a SIP 181/182/183 response is received, no QSIG ALERTING message
has been sent, no QSIG PROGRESS message containing progress
description number 1 or 8 has been sent and no media stream has been
established towards the gateway, the gateway SHALL generate a QSIG
PROGRESS message. The QSIG PROGRESS message SHALL contain progress
description number 1 in a Progress indicator information element.
NOTE 1. This will ensure that QSIG timer T310 is stopped if running
at the Originating PINX.
NOTE 2. Media streams are established as a result of receiving SDP
answer information in a reliable provisional response and can be
modified by means of the SIP UPDATE method (RFC 3311, [15]). If a
media stream is established towards the gateway, connecting the media
stream to the corresponding user information channel on the inter-
PINX link will allow the caller to hear in-band tones or
announcements.
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 RFC 3262.
8.2.1.4 Receipt of SIP 2xx response
If the gateway receives a SIP 200 (OK) response as the first SIP 200
response to a SIP INVITE request, the gateway SHALL map the SIP 200
(OK) response to a QSIG CONNECT message. The gateway SHALL also send
a SIP ACK request to acknowledge the 200 (OK) response. The gateway
SHALL NOT include any SDP information in the SIP ACK request. If the
gateway receives further 200 (OK) responses, it SHALL respond to each
in accordance with RFC 3261 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 200 (OK) response is received in response to the SIP PRACK
request, the gateway SHALL NOT map this message to any QSIG message.
If the gateway receives a SIP 2xx response other than 200 (OK), the
gateway SHALL send a SIP ACK request.
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NOTE. A SIP 200 (OK) response can be received later as a result of a
forking proxy.
8.2.1.5 Receipt of SIP 3xx response
On receipt of a SIP 3xx response, the gateway SHALL act in accordance
with RFC 3261.
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 8.2.2.1
SHALL apply.
However, for some applications it might be impossible to avoid using
overlap signalling in the SIP network. In this case the procedures
defined in 8.2.2.2 SHALL apply.
8.2.2.1 Enbloc 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.
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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 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
SIP responses SHALL be mapped as described in 8.2.1.
8.2.2.2 Overlap signalling in SIP network
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 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 ECMA-143.
NOTE 1. ECMA-143 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 8.2.2.1.2. If the gateway still does not have the
minimum number of digits required than it SHALL wait for more QSIG
INFORMATION messages to arrive.
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If the gateway has already sent one or more SIP INVITE requests, and
whether or not final responses to those requests have been received,
it SHALL send a new SIP INVITE request with the new digits. The new
SIP INVITE request SHALL have the same Call-ID as the first SIP
INVITE request sent but SHALL have updated Request-URI and To fields.
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. The CSeq header SHOULD contain a value higher than that in
the previous SIP INVITE request.
NOTE 2. The first SIP INVITE request and all subsequent SIP INVITE
requests sent in this way belong to the same call but to different
dialogs.
8.2.2.2.3 Receipt of SIP 100 (Trying) response
The requirements of 8.2.1.2 SHALL apply.
8.2.2.2.4 Receipt of SIP 18x provisional response
The requirements of 8.2.1.3 SHALL apply.
8.2.2.2.5 Receipt of SIP 2xx response
The requirements of 8.2.1.4 SHALL apply. In addition the gateway
SHALL send a SIP CANCEL request, either immediately or after a short
delay, to cancel any SIP INVITE transactions for which no final
response has been received.
NOTE. Delaying the sending of a SIP CANCEL request allows time for
final responses to be received on any outstanding transactions,
thereby avoiding unnecessary signalling.
8.2.2.2.6 Receipt of SIP 3xx response
The requirements of 8.2.1.5 SHALL apply.
8.2.2.2.7 Receipt of a SIP 484 (Address Incomplete) response
The SIP 484 response indicates that more digits are required to
complete the call. On receipt of a SIP 484 response the gateway SHALL
send back a SIP ACK request. 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.
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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 484 response SHALL NOT
trigger the sending of any QSIG message.
8.2.2.2.8 Receipt of a SIP 4xx (except 484), 5xx or 6xx response
If a SIP 4xx (except 484), 5xx or 6xx final response arrives for a
pending SIP INVITE transaction, the gateway SHALL send a SIP ACK
request. If this occurs when no further QSIG INFORMATION messages are
expected and final responses have been received to all transmitted
SIP INVITE requests, the gateway SHALL send a QSIG DISCONNECT message
(8.4.4). Otherwise the gateway MAY send a QSIG DISCONNECT message.
NOTE. The gateway can take account of the SIP response code and other
information to assess whether to wait for further responses before
initiating clearing.
8.2.2.2.9 Receipt of multiple SIP responses
The responses to all the SIP INVITE requests sent except for the last
one are typically SIP 4xx responses (e.g. 484 (Address Incomplete))
that terminate the SIP INVITE transaction.
However, the gateway can receive a SIP 183 (Session Progress)
response with a media description. The media stream will typically
contain a message such as "àWe are trying to connect youà ". The
issue of receiving different SIP 183 (Session Progress) responses
with media descriptions for different SIP INVITE transactions is a
gateway concern. The gateway SHOULD decide which media stream (if
any) is to be played to the user.
NOTE. The gateway can receive multiple SIP 183 responses with media
description not only as a result of sending multiple INVITE requests
due to overlap sending but also as a result of a forking proxy.
8.2.2.2.10 Cancelling pending SIP INVITE transactions
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
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8.2.2.2.5 there is provision for cancelling any transactions still in
progress after a SIP 2xx response has been received.
8.2.2.2.11 SIP INVITE requests reaching multiple gateways
Each SIP INVITE request sent by a gateway represents a new
transaction and hence can be routed differently. For instance, the
first SIP INVITE request might be routed to a particular egress
gateway and a subsequent SIP INVITE request to another gateway. The
result is that both gateways initiate call establishment in the
remote network. Since one of the call establishments has an
incomplete destination number, it can be expected to fail, having
already consumed resources in the remote network.
To avoid this problem it is RECOMMENDED that all the SIP INVITE
requests should follow the same path as the first one. This would
however restrict the number of services the SIP network can provide.
It would not be possible to route a subsequent SIP INVITE request to
an application server just because the previous one was routed in a
different way.
This issue should be taken into consideration before using overlap
signalling in SIP. If initiating multiple call establishments in the
remote network is not acceptable in a particular application, overlap
signalling SHOULD NOT be used.
8.2.2.2.12 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, and 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
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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 1. 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.
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 SHALL NOT issue a QSIG SETUP message
and SHALL send a SIP 488 (Not Acceptable Here) response.
NOTE 2. 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 RFC 3262) 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.
NOTE 3. 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.
8.3.3 Receipt of QSIG PROGRESS message
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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. 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 RFC 3262, inclusion of option tag 100rel in
a provisional response instructs the UAC to acknowledge the
provisional response by sending a PRACK request. RFC 3262 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 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 183 provisional response SHALL be
in accordance with 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. 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.
NOTE. In accordance with RFC 3262, inclusion of option tag 100rel in
a provisional response instructs the UAC to acknowledge the
provisional response by sending a PRACK request. RFC 3262 also
specifies procedures for repeating a provisional response with option
tag 100rel if no PRACK is received.
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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 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 8.3.5.
8.3.5 Inclusion of SDP information in a SIP 18x provisional response
When sending a SIP 18x provisional response, the gateway SHALL
include SDP information 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 1. 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.
NOTE 2. 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 3. 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 8.3.1.
8.3.6 Receipt of QSIG CONNECT message
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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 1. 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 2. Because the provisional response is unreliable, SDP answer
information needs to be repeated in each provisional response and in
the final 2xx response.
NOTE 3. 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 8.3.1.
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 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
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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
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.
If a gateway receives a SIP INVITE request with the same Call-ID as
an existing call for which the QSIG state is overlap sending and with
updated Request-URI and To fields from which a called party number
with a superset of digits can be derived, it SHALL generate a QSIG
INFORMATION message using the call reference of the existing QSIG
call instead of a new QSIG SETUP message. 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 with the same Call-ID as
an existing SIP INVITE request for which the gateway has not yet sent
a final response and failing to meet the other conditions above
concerning overlap sending, the gateway SHALL clear the call by
sending back a SIP 485 (Ambiguous) response and a QSIG DISCONNECT
message with Cause Value 16 (Normal call clearing).
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 and received a SIP
ACK request or has received a SIP 200 (OK) response and sent a SIP
ACK request, the gateway SHALL send a SIP BYE request to clear the
call.
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2)If the gateway has sent a SIP 200 (OK) response (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.
NOTE 1. In accordance with RFC 3261, 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 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 ECMA-143 and RFC-3261 respectively.
Table 1 û Mapping of QSIG Cause Value to SIP 4xx-6xx responses
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
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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
implemented 4)
69 Requested facility not 501 Not implemented
implemented
70 Only restricted digital 488 Not acceptable here (NOTE
information available 4)
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.
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However, if a SIP response is to be sent, 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 ECMA-143
and RFC-3261 respectively.
NOTE. When responding to a SIP BYE request, in accordance with
RFC 3261 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 ECMA-143 and RFC-3261 respectively.
8.4.4 Receipt of a SIP 4xx - 6xx response
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, the gateway SHALL transmit a QSIG DISCONNECT message.
The cause value in the QSIG DISCONNECT message 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 ECMA-143 and RFC-3261 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 û Mapping of SIP 4xx-6xx responses to QSIG Cause values
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SIP response QSIG Cause value
400 Bad request 41 Temporary failure
401 Unauthorized 21 Call rejected (NOTE 1)
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 1)
408 Request timeout 102 Recovery on timer expiry
410 Gone 22 Number changed
413 Request entity too large 127 Interworking, unspecified
(NOTE 2)
414 Request-URI too long 127 Interworking, unspecified
(NOTE 2)
415 Unsupported media type 79 Service or option not
implemented, unspecified (NOTE
2)
416 Unsupported URI scheme 127 Interworking, unspecified
(NOTE 2)
420 Bad extension 127 Interworking, unspecified
(NOTE 2)
421 Extension required 127 Interworking, unspecified
(NOTE 2)
423 Interval too brief 127 Interworking, unspecified
(NOTE 2)
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
484 Address incomplete 28 Invalid number format (NOTE
2)
485 Ambiguous 1 Unallocated Number
486 Busy here 17 User busy
487 Request terminated (NOTE 3)
488 Not Acceptable Here 65 Bearer capability not
implemented or 31 Normal,
unspecified(NOTE 4)
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 2)
513 Message too large 127 Interworking, unspecified
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Interworking between SIP and QSIG October 2002
(NOTE 2)
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 4)
NOTE 1. 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 2. If at all possible, the gateway should respond to these
protocol errors by remedying unacceptable behavior and attempting to
re-originate the session. Only if this proves impossible should the
gateway clear the call in the QSIG network with this cause value.
NOTE 3. 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 4. 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
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 ECMA-143, the gateway SHALL also initiate clearing
of the SIP call in accordance with 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 RFC 3261, the gateway
SHALL also initiate clearing of the QSIG call in accordance with
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Interworking between SIP and QSIG October 2002
ECMA-143 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
503 (Service unavailable) 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 ECMA-155
[19]. 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 that 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 RFC 3261 and the P-Asserted-Identity
header specified in RFC BBBB [14]. In addition, privacy is indicated
by the Privacy header specified in RFC AAAA [13].
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 RFC BBBB [14]. If a gateway implements the P-Asserted-Identity
header it SHALL also implement the Privacy header in accordance with
RFC AAAA [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 trusts the
adjacent proxy (i.e., the proxy 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-
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Interworking between SIP and QSIG October 2002
dependent and it is RECOMMENDED that gateways supporting the
P-Asserted-Identity header be configurable to either trust or not
trust the proxy 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.
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 NOT generate a
P-Asserted-Identity header, SHALL NOT generate a Privacy header and
SHALL include a URI identifying the gateway in the From header.
9.1.2.2 No URI derived and presentation indicator has value
"presentation restricted"
In this case the gateway SHALL NOT generate a P-Asserted-Identity
header, SHALL generate a Privacy header with parameter priv-value =
"id" if the gateway supports this header, 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.
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
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
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Interworking between SIP and QSIG October 2002
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 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.
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 the gateway SHALL NOT generate a P-Asserted-Identity
header and SHALL generate a Privacy header with parameter priv-value
= "id" if the gateway supports this 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
proxy to honour the Privacy header, the gateway SHALL generate a
Elwell et alia Expires - April 2003 [Page 32]
Interworking between SIP and QSIG October 2002
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 9.1.3.2.
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.
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 ECMA-155.
Some aspects of mapping depend on whether the gateway trusts the
adjacent proxy (i.e., the proxy 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 be configurable to
either trust or not trust the proxy 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 the 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 consider 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 the 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.
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
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Interworking between SIP and QSIG October 2002
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 a P-Asserted-Identity header and
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 the P-Asserted-Identity header,
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 from a P-Asserted-Identity header
(including the case where there is no P-Asserted-Identity header) and
if the gateway is prepared to use the 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, the
gateway SHALL include value "presentation restricted" in the
presentation indicator. 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 Calling party 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 SETUP message.
9.2.3 Generating the QSIG Connected number information element
When mapping a SIP 200 (OK) response to an INVITE request to a QSIG
CONNECT message, the gateway SHALL generate a Connected number
information element as follows.
If the SIP 200 (OK) response contains a P-Asserted-Identity header
and the gateway supports that header and trusts the information
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therein, the gateway SHALL attempt to derive a number from the URI in
that header. If a number is derived from the P-Asserted-Identity
header, 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 (including the case where there is no
P-Asserted-Identity header), the gateway SHALL NOT include a number
in the Connected number information element.
If the SIP 200 (OK) 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. 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.
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 information received along
with the SIP INVITE request. If the SIP INVITE request does not
contain SDP information or the media type in the SDP information is
only æaudioÆ 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.
Table 3 û Bearer capability encoding for æaudioÆ 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
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protocol information of the PISN. Supported
values are
"CCITT recommendation G.711 @-law"
(00010)
"CCITT recommendation G.711 A-law"
(00011)
10.2 Derivation of media type in SDP
The gateway SHALL generate SDP information to include in the SIP
INVITE request based on the Bearer capability information element
received in the QSIG SETUP message. The media type included in the
SDP information SHALL be according to table 4.
Table 4 û 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
"unrestricted digital information" (01000) data
11 Security considerations
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 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.
In most respects, the information that is translated from QSIG to SIP
has no special security requirements. In order for translated
information elements to be used to route requests, they should be
legible to intermediaries; end-to-end confidentiality of this data
would be unnecessary and most likely detrimental. There are also
numerous circumstances under which intermediaries can legitimately
overwrite the values that have been provided by translation, and
hence integrity over these headers is similarly not desirable.
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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. 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 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.
Some additional risks may result from the SIP response code to QSIG
cause value mapping. 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 8.4.4.
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 retargeted 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.
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 the From
header to populate the Calling party number information element of a
QSIG SETUP message should authenticate the originator of the request
Elwell et alia Expires - April 2003 [Page 37]
Interworking between SIP and QSIG October 2002
and make sure that it is authorized to assert that calling number (or
make use of some more 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.
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). 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 to 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.
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,
to incomplete addresses. This could saturate resources at the gateway
indefinitely, potentially without incurring any charges. Gateways
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.
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12 Author's Addresses
John Elwell
Siemens Communications
Technology Drive
Beeston
Nottingham, UK, NG9 1LA
email: john.elwell@siemens.com
Frank Derks
Philips Business Communications
P.O. Box 32
1200 JD, Hilversum
The Netherlands
email: frank.derks@philips.com
Olivier Rousseau
Alcatel Business Systems
32,Avenue Kleber
92700 Colombes
France
email: olivier.rousseau@col.bsf.alcatel.fr
Patrick Mourot
Alcatel Business Systems
1,Rue Dr A. Schweitzer
67400 Illkirch
France
email: patrick.mourot@sxb.bsf.alcatel.fr
13 Normative References
[1] ECMA-133 "Private Integrated Services Network (PISN û Reference
configuration for PISN exchanges (PINX)" (International Standard
ISO/IEC 11579-1)
[2] ECMA-143 "Private Integrated Services Network - Circuit-mode
Bearer Services - Inter-Exchange Signalling Procedures and Protocol"
(International Standard ISO/IEC 11572)
[3] ECMA-165 "Private Integrated Services Network - Generic
Functional Protocol for the Support of Supplementary Services -
Inter-Exchange Signalling Procedures and Protocol" (International
Standard ISO/IEC 11582)
[4] ECMA-307 "Corporate Telecommunication Networks - Signalling
Elwell et alia Expires - April 2003 [Page 39]
Interworking between SIP and QSIG October 2002
Interworking between QSIG and H.323 - Generic Functional Protocol for
the Support of Supplementary Services" (International Standard
ISO/IEC 21409)
[5] J. Postel, "Transmission Control Protocol", RFC 793.
[6] J. Postel, "User Datagram Protocol", RFC 768.
[7] T. Dierks, C.Allen, "The TLS protocol version 1.0", RFC 2246.
[8] M. Handley, V. Jacobson, "SDP: Session Description Protocol", RFC
2327.
[9] R. Stewart et al., "Stream Control Transmission Protocol" RFC
2960.
[10] J. Rosenberg, H. Schulzrinne, et al., "SIP: Session initiation
protocol", RFC 3261.
[11] J. Rosenberg, H. Schulzrinne, "Reliability of Provisional
Responses in SIP", RFC 3262.
[12] J. Rosenberg, H. Schulzrinne, "An Offer/Answer Model with SDP",
RFC 3264.
[13] J. Peterson, ôA Privacy Mechanism for the Session Initiation
Protocol (SIP)ö, RFC AAAA
[14] C. Jennings, J. Peterson, M. Watson, ôPrivate Extensions to the
Session Initiation Protocol (SIP) for Asserted Identity within
Trusted Networksö, RFC BBBB
[15] J. Rosenberg, "The Session Initiation Protocol UPDATE Method",
RFC 3311.
[16] J. Postel, "Internet Protocol", RFC 791.
[17] S. Deering, R. Hinden, "Internet Protocol, Version 6 (IPv6)ö,
RFC 2460.
[18] ITU-T Recommendation E.164, ôThe International Public
Telecommunication Numbering Planö, (1997-05).
[19] ECMA-155 "Private Integrated Services Networks (PISN) û
Addressingö (International Standard ISO/IEC 11571)
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Annex A (informative) û Example message sequences
A.1 Introduction
This annex shows some typical message sequences that can occur for an
interworking between QSIG and SIP.
NOTE 1. 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 2. 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 3. 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
.-------------------.
| |
| 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
| | | |
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| QSIG CONNECT | | 1-200 OK |
11|<-----------------------|......|<-----------------------+ 9
| | | |
| QSIG CONNECT ACK | | 1-ACK |
12|----------------------->| |----------------------->| 10
| | | |
|<======================>| |<======================>|
| AUDIO | | AUDIO |
Figure 3 û Typical message sequence for successful call establishment
from QSIG to SIP using enbloc 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.
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.
+------------------------+
PISN | GATEWAY | IP NETWORK
| |
| QSIG SETUP +--------+-------+-------+ |
1|-------------------------->| | |
| | | |
| QSIG SETUP ACK | | |
2|<--------------------------| | |
| | | |
| QSIG INFORMATION | | |
3|-------------------------->| | |
| | | |
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Interworking between SIP and QSIG October 2002
| 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 û Typical message sequence for successful call establishment
from QSIG to SIP using overlap receiving on QSIG and enbloc 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
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.
Elwell et alia Expires - April 2003 [Page 43]
Interworking between SIP and QSIG October 2002
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.
+----------------------+
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 alia Expires - April 2003 [Page 44]
Interworking between SIP and QSIG October 2002
|<========================>| |<=======================>|
| | | |
Figure 5 û 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 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.
A.3 Message sequences for call establishment from SIP to QSIG
Elwell et alia Expires - April 2003 [Page 45]
Interworking between SIP and QSIG October 2002
Below are typical message sequences for successful call establishment
from SIP to QSIG
+----------------------+
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 û Typical message sequence for successful call establishment
from SIP to QSIG using enbloc 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
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
Elwell et alia Expires - April 2003 [Page 46]
Interworking between SIP and QSIG October 2002
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
+----------------------+
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|------------------------->| | |
| 5-200 OK | | |
11|<-------------------------| | |
| 3-200 OK | | QSIG CONNECT |
14|<-------------------------|.......|<------------------------|12
| | | |
| 3-ACK | | QSIG CONNECT ACK |
15|------------------------->| |------------------------>|13
| AUDIO | | AUDIO |
|<========================>| |<=======================>|
| | | |
Figure 7 û Typical message sequence for successful call establishment
from SIP to QSIG using overlap receiving on SIP and enbloc 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.
Elwell et alia Expires - April 2003 [Page 47]
Interworking between SIP and QSIG October 2002
3 The IP network acknowledge the SIP 484 (Address Incomplete)
response.
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
+----------------------+
IP NETWORK | GATEWAY | PISN
| |
| 1-INVITE +-------+-------+------+ |
1 |------------------------->| | |
| 1-484 | | |
2 |<-------------------------| | |
| 1-ACK | | |
3 |------------------------->| | |
| 2-INVITE | | QSIG SETUP |
4 |------------------------->|.......|------------------------>|6
| 2-100 TRYING | | QSIG SETUP AC |
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|------------------------->| | |
Elwell et alia Expires - April 2003 [Page 48]
Interworking between SIP and QSIG October 2002
| 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 û 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 acknowledge the SIP 484 (Address Incomplete)
response.
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 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
Elwell et alia Expires - April 2003 [Page 49]
Interworking between SIP and QSIG October 2002
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
A.4 Message sequence for call clearing from QSIG to SIP
Below are typical message sequences for Call Clearing from QSIG to
SIP
.-------------------.
| |
| GATEWAY |
PISN | | IP
NETWORK
| `-----+------+------' |
| | | |
| | | |
| QSIG DISCONNECT | | 2- BYE |
1|----------------------->|......|-----------------------
>|4
| QSIG RELEASE | | 2-200 OK |
2|<-----------------------| |<-----------------------
|5
| QSIG RELEASE COMP | | |
3|----------------------->| | |
| | | |
| | | |
| | | |
Figure 9 û 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 alia Expires - April 2003 [Page 50]
Interworking between SIP and QSIG October 2002
| GATEWAY |
PISN | | IP NETWORK
| `-----+------+------' |
| | | |
| | | |
| QSIG DISCONNECT | | 1- 4XX / 6XX |
1|----------------------->|......|---------------------->|4
| QSIG RELEASE | | 1- ACK |
2|<-----------------------| |<----------------------|5
| QSIG RELEASE COMP | | |
3|----------------------->| | |
| | | |
| | | |
Figure 10 û 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
.-------------------.
| |
| 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
| | | |
Elwell et alia Expires - April 2003 [Page 51]
Interworking between SIP and QSIG October 2002
Figure 11 û 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 a provisional response but no final response
having been received)
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
A.5 Message sequence for call clearing from SIP to QSIG
Below are typical message sequences for Call Clearing from SIP to
QSIG
.-------------------.
| |
| GATEWAY |
IP NETWORK | | PISN
| `-----+------+------' |
| | | |
| | | |
| 2- BYE | | QSIG DISCONNECT |
1|----------------------->|......|----------------------->|3
| | | QSIG RELEASE |
| | |<-----------------------|4
| 2-200 OK | | QSIG RELEASE COMP |
2|<-----------------------| |----------------------->|5
| | | |
| | | |
Figure 12 û 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
Elwell et alia Expires - April 2003 [Page 52]
Interworking between SIP and QSIG October 2002
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.
.-------------------.
| |
| GATEWAY |
IP NETWORK | | PISN
| `-----+------+------' |
| | | |
| | | |
| 1- 4XX / 6XX | | QSIG DISCONNECT |
1|----------------------->|......|----------------------->|3
| | | QSIG RELEASE |
| | |<-----------------------|4
| 1- ACK | | QSIG RELEASE COMP |
2|<-----------------------| |----------------------->|5
| | | |
| | | |
| | | |
Figure 13 û 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.
.-------------------.
| |
| GATEWAY |
IP NETWORK | | PISN
| `-----+------+------' |
| | | |
| | | |
| 1- CANCEL | | QSIG DISCONNECT |
1|----------------------->|......|----------------------->|4
| | | QSIG RELEASE |
| | |<-----------------------|5
|1-487 Request Terminated| | QSIG RELEASE COMP |
2|<-----------------------| |----------------------->|6
Elwell et alia Expires - April 2003 [Page 53]
Interworking between SIP and QSIG October 2002
| | | |
| 1- ACK | | |
3|----------------------->| | |
| | | |
| 1- 200 OK | | |
4|<-----------------------| | |
Figure 14 û 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.
Annex B (temporary) û Change log
Compared with draft-elwell-sipping-qsig2sip-02 the following changes
have been made:
- addition of statement concerning QSIG segmented messages in section
8.1;
- clarifications concerning backward media establishment and local
ringback tone in section 8.2.1;
- clarifications to overlap sending in 8.2.2.2;
- additions concerning availability of channel on inter-PINX link in
8.3.1;
- addition of note on handling certain QSIG cause values locally
rather than mapping to SIP response codes in 8.4.1;
- addition of statement concerning generation of location field in
QSIG Cause information element in 8.4.4;
- additions covering QSIG RESTART message and QSIG timer T301 expiry
in 8.4.5;
- addition of procedures for use of P-Asserted-Identity and Privacy
headers in section 9;
- addition of security considerations text in section 11, based on
corresponding text in ISUP draft;
Elwell et alia Expires - April 2003 [Page 54]
Interworking between SIP and QSIG October 2002
- corrections to figures 11 and 14.
Elwell et alia Expires - April 2003 [Page 55]