Request for Comments: 4497 Siemens
BCP: 117 F. Derks
Category: Best Current Practice NEC Philips
P. Mourot
O. Rousseau
Alcatel
May 2006
Interworking between the Session Initiation Protocol (SIP) and QSIG
Status of This Memo
This document specifies an Internet Best Current Practices for the
Internet Community, and requests discussion and suggestions for
improvements. Distribution of this memo is unlimited.
Copyright Notice
Copyright (C) The Internet Society (2006).
Abstract
This document specifies interworking between the Session Initiation
Protocol (SIP) and QSIG within corporate telecommunication networks
(also known as enterprise networks). SIP is an Internet
application-layer control (signalling) protocol for creating,
modifying, and terminating sessions with one or more participants.
These sessions include, in particular, telephone calls. QSIG is a
signalling protocol for creating, modifying, and terminating
circuit-switched calls (in particular, telephone calls) within
Private Integrated Services Networks (PISNs). QSIG is specified in a
number of Ecma Standards and published also as ISO/IEC standards.
Table of Contents
1. Introduction ....................................................4
2. Terminology .....................................................5
3. Definitions .....................................................5
3.1. External Definitions .......................................5
3.2. Other definitions ..........................................5
3.2.1. Corporate Telecommunication Network (CN) ............5
3.2.2. Gateway .............................................6
3.2.3. IP Network ..........................................6
3.2.4. Media Stream ........................................6
3.2.5. Private Integrated Services Network (PISN) ..........6
3.2.6. Private Integrated Services Network Exchange
(PINX) ..............................................6
4. Acronyms ........................................................6
5. Background and Architecture .....................................7
6. Overview .......................................................10
7. General Requirements ...........................................11
8. Message Mapping Requirements ...................................12
8.1. Message Validation and Handling of Protocol Errors ........12
8.2. Call Establishment from QSIG to SIP .......................14
8.2.1. Call Establishment from QSIG to SIP Using
En Bloc Procedures .................................14
8.2.2. Call Establishment from QSIG to SIP Using
Overlap Procedures .................................16
8.3. Call Establishment from SIP to QSIG .......................20
8.3.1. Receipt of SIP INVITE Request for a New Call .......20
8.3.2. Receipt of QSIG CALL PROCEEDING Message ............21
8.3.3. Receipt of QSIG PROGRESS Message ...................22
8.3.4. Receipt of QSIG ALERTING Message ...................22
8.3.5. Inclusion of SDP Information in a SIP 18x
Provisional Response ...............................23
8.3.6. Receipt of QSIG CONNECT Message ....................24
8.3.7. Receipt of SIP PRACK Request .......................25
8.3.8. Receipt of SIP ACK Request .........................25
8.3.9. Receipt of a SIP INVITE Request for a Call
Already Being ......................................25
8.4. Call Clearing and Call Failure ............................26
8.4.1. Receipt of a QSIG DISCONNECT, RELEASE, or
RELEASE COMPLETE ...................................26
8.4.2. Receipt of a SIP BYE Request .......................29
8.4.3. Receipt of a SIP CANCEL Request ....................29
8.4.4. Receipt of a SIP 4xx-6xx Response to an
INVITE Request .....................................29
8.4.5. Gateway-Initiated Call Clearing ....................32
8.5. Request to Change Media Characteristics ...................32
9. Number Mapping .................................................32
9.1. Mapping from QSIG to SIP ..................................33
9.1.1. Using Information from the QSIG Called
Party Number Information Element ...................33
9.1.2. Using Information from the QSIG Calling
Party Number Information Element ...................33
9.1.3. Using Information from the QSIG Connected
Number Information Element .........................35
9.2. Mapping from SIP to QSIG ..................................36
9.2.1. Generating the QSIG Called Party Number
Information Element ................................36
9.2.2. Generating the QSIG Calling Party Number
Information Element ................................37
9.2.3. Generating the QSIG Connected Number
Information Element ................................38
10. Requirements for Support of Basic Services ....................39
10.1. Derivation of QSIG Bearer Capability Information
Element ..................................................39
10.2. Derivation of Media Type in SDP ..........................39
11. Security Considerations .......................................40
11.1. General ..................................................40
11.2. Calls from QSIG to Invalid or Restricted Numbers .........40
11.3. Abuse of SIP Response Code ...............................41
11.4. Use of the To Header URI .................................41
11.5. Use of the From Header URI ...............................41
11.6. Abuse of Early Media .....................................42
11.7. Protection from Denial-of-Service Attacks ................42
12. Acknowledgements ..............................................43
13. Normative References ..........................................43
Appendix A. Example Message Sequences .............................45
1. Introduction
This document specifies signalling interworking between QSIG and the
Session Initiation Protocol (SIP) in support of basic services within
a corporate telecommunication network (CN) (also known as enterprise
network).
QSIG is a signalling protocol that operates between Private
Integrated Services eXchanges (PINX) within a Private Integrated
Services Network (PISN). A PISN provides circuit-switched basic
services and supplementary services to its users. QSIG is specified
in Ecma Standards; in particular, [2] (call control in support of
basic services), [3] (generic functional protocol for the support of
supplementary services), and a number of standards specifying
individual supplementary services.
NOTE: The name QSIG was derived from the fact that it is used for
signalling at the Q reference point. The Q reference point is a
point of demarcation between two PINXs.
SIP is an application-layer protocol for establishing, terminating,
and modifying multimedia sessions. It is typically carried over IP
[15], [16]. Telephone calls are considered a type of multimedia
session where just audio is exchanged. SIP is defined in [10].
As the support of telephony within corporate networks evolves from
circuit-switched technology to Internet technology, the two
technologies will coexist in many networks for a period, perhaps
several years. Therefore, there is a need to be able to establish,
modify, and terminate sessions involving a participant in the SIP
network and a participant in the QSIG network. Such calls are
supported by gateways that perform interworking between SIP and QSIG.
This document specifies SIP-QSIG signalling interworking for basic
services that provide a bi-directional transfer capability for
speech, DTMF, facsimile, and modem media between a PISN employing
QSIG and a corporate IP network employing SIP. Other aspects of
interworking, e.g., the use of RTP and SDP, will differ according to
the type of media concerned and are outside the scope of this
specification.
Call-related and call-independent signalling in support of
supplementary services is outside the scope of this specification,
but support for certain supplementary services (e.g., call transfer,
call diversion) could be the subject of future work.
Interworking between QSIG and SIP permits a call originating at a
user of a PISN to terminate at a user of a corporate IP network, or a
call originating at a user of a corporate IP network to terminate at
a user of a PISN.
Interworking between a PISN employing QSIG and a public IP network
employing SIP is outside the scope of this specification. However,
the functionality specified in this specification is in principle
applicable to such a scenario when deployed in conjunction with other
relevant functionality (e.g., number translation, security functions,
etc.).
This specification is applicable to any interworking unit that can
act as a gateway between a PISN employing QSIG and a corporate IP
network employing SIP.
2. Terminology
In this document, the key words "MUST", "MUST NOT", "REQUIRED",
"SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY",
and "OPTIONAL" are to be interpreted as described in RFC 2119 [4] and
indicate requirement levels for compliant SIP implementations.
3. Definitions
For the purposes of this specification, the following definitions
apply.
3.1. External Definitions
The definitions in [2] and [10] apply as appropriate.
3.2. Other definitions
3.2.1. Corporate Telecommunication Network (CN)
Sets of privately-owned or carrier-provided equipment that are
located at geographically dispersed locations and are interconnected
to provide telecommunication services to a defined group of users.
NOTE: A CN can comprise a PISN, a private IP network (intranet), or a
combination of the two.
3.2.2. Gateway
An entity that performs interworking between a PISN using QSIG and an
IP network using SIP.
3.2.3. IP Network
A network (unless otherwise stated, a corporate network) offering
connectionless packet-mode services based on the Internet Protocol
(IP) as the network-layer protocol.
3.2.4. Media Stream
Audio or other user information transmitted in UDP packets, typically
containing RTP, in a single direction between the gateway and a peer
entity participating in a session established using SIP.
NOTE: Normally a SIP session establishes a pair of media streams, one
in each direction.
3.2.5. Private Integrated Services Network (PISN)
A CN or part of a CN that employs circuit-switched technology.
3.2.6. Private Integrated Services Network Exchange (PINX)
A PISN nodal entity comprising switching and call handling functions
and supporting QSIG signalling in accordance with [2].
4. Acronyms
DNS Domain Name Service
IP Internet Protocol
PINX Private Integrated services Network eXchange
PISN Private Integrated Services Network
RTP Real-time Transport Protocol
SCTP Stream Control Transmission Protocol
SDP Session Description Protocol
SIP Session Initiation Protocol
TCP Transmission Control Protocol
TLS Transport Layer Security
TU Transaction User
UA User Agent
UAC User Agent Client
UAS User Agent Server
UDP User Datagram Protocol
5. Background and Architecture
During the 1980s, corporate voice telecommunications adopted
technology similar in principle to Integrated Services Digital
Networks (ISDN). Digital circuit switches, commonly known as Private
Branch eXchanges (PBX) or more formally as Private Integrated
services Network eXchanges (PINX) have been interconnected by digital
transmission systems to form Private Integrated Services Networks
(PISN). These digital transmission systems carry voice or other
payload in fixed-rate channels, typically 64 Kbit/s, and signalling
in a separate channel. A technique known as common channel
signalling is employed, whereby a single signalling channel
potentially controls a number of payload channels or bearer channels.
A typical arrangement is a point-to-point transmission facility at T1
or E1 rate providing a 64 Kbit/s signalling channel and 23 or 30
bearer channels, respectively. Other arrangements are possible and
have been deployed, including the use of multiple transmission
facilities for a signalling channel and its logically associated
bearer channels. Also, arrangements involving bearer channels at
sub-64 Kbit/s have been deployed, where voice payload requires the
use of codecs that perform compression.
QSIG is the internationally-standardized message-based signalling
protocol for use in networks as described above. It runs in a
signalling channel between two PINXs and controls calls on a number
of logically associated bearer channels between the same two PINXs.
The signalling channel and its logically associated bearer channels
are collectively known as an inter-PINX link. QSIG is independent of
the type of transmission capabilities over which the signalling
channel and bearer channels are provided. QSIG is also independent
of the transport protocol used to transport QSIG messages reliably
over the signalling channel.
QSIG provides a means for establishing and clearing calls that
originate and terminate on different PINXs. A call can be routed
over a single inter-PINX link connecting the originating and
terminating PINX, or over several inter-PINX links in series with
switching at intermediate PINXs known as transit PINXs. A call can
originate or terminate in another network, in which case it enters or
leaves the PISN environment through a gateway PINX. Parties are
identified by numbers, in accordance with either [17] or a private
numbering plan. This basic call capability is specified in [2]. In
addition to basic call capability, QSIG specifies a number of further
capabilities supporting the use of supplementary services in PISNs.
More recently, corporate telecommunications networks have started to
exploit IP in various ways. One way is to migrate part of the
network to IP using SIP. This might, for example, be a new branch
office with a SIP proxy and SIP endpoints instead of a PINX.
Alternatively, SIP equipment might be used to replace an existing
PINX or PINXs. The new SIP environment needs to interwork with the
QSIG-based PISN in order to support calls originating in one
environment and terminating in the other. Interworking is achieved
through a gateway.
Interworking between QSIG and SIP at gateways can also be used where
a SIP network interconnects different parts of a PISN, thereby
allowing calls between the different parts. A call can enter the SIP
network at one gateway and leave at another. Each gateway would
behave in accordance with this specification.
Another way of connecting two parts of a PISN would be to encapsulate
QSIG signalling in SIP messages for calls between the two parts.
This is outside the scope of this specification but could be the
subject of future work.
This document specifies signalling protocol interworking aspects of a
gateway between a PISN employing QSIG signalling and an IP network
employing SIP signalling. The gateway appears as a PINX to other
PINXs in the PISN. The gateway appears as a SIP endpoint to other
SIP entities in the IP network. The environment is shown in Figure
1.
+------+ IP network PISN
| |
|SIP | +------+
|Proxy | /| |
| | / |PINX |
+---+--+ *-----------+ / | |
| | | +-----+/ +------+
| | | | |
| | | |PINX |
---+-----+-------+--------+ Gateway +--------| |
| | | | | |\
| | | | +-----+ \
| | | | \ +------+
| | | | \| |
+--+---+ +--+---+ *-----------+ |PINX |
|SIP | |SIP | | |
|End- | |End- | +------+
|point | |point |
+------+ +------+
Figure 1: Environment
In addition to the signalling interworking functionality specified in
this specification, it is assumed that the gateway also includes the
following functionality:
- one or more physical interfaces on the PISN side supporting one or
more inter-PINX links, each link providing one or more constant bit
rate channels for media streams and a reliable layer 2 connection
(e.g., over a fixed rate physical channel) for transporting QSIG
signalling messages; and
- one or more physical interfaces on the IP network side supporting,
through layer 1 and layer 2 protocols, IP as the network layer
protocol and UDP [6] and TCP [5] as transport layer protocols,
these being used for the transport of SIP signalling messages and,
in the case of UDP, also for media streams;
- optionally the support of TLS [7] and/or SCTP [9] as additional
transport layer protocols on the IP network side, these being used
for the transport of SIP signalling messages; and
- a means of transferring media streams in each direction between the
PISN and the IP network, including as a minimum packetization of
media streams sent to the IP network and de-packetization of media
streams received from the IP network.
NOTE: [10] mandates support for both UDP and TCP for the transport of
SIP messages and allows optional support for TLS and/or SCTP for this
same purpose.
The protocol model relevant to signalling interworking functionality
of a gateway is shown in Figure 2.
+---------------------------------------------------------+
| Interworking function |
| |
+-----------------------+---------+-----------------------+
| | | |
| SIP | | |
| | | |
+-----------------------+ | |
| | | |
| UDP/TCP/TLS/SCTP | | QSIG |
| | | |
+-----------------------+ | |
| | | |
| IP | | |
| | | |
+-----------------------+ +-----------------------+
| IP network | | PISN |
| lower layers | | lower layers |
| | | |
+-----------------------+ +-----------------------+
Figure 2: Protocol model
In Figure 2, the SIP box represents SIP syntax and encoding, the SIP
transport layer, and the SIP transaction layer. The Interworking
function includes SIP Transaction User (TU) functionality.
6. Overview
The gateway maps received QSIG messages, where appropriate, to SIP
messages and vice versa and maintains an association between a QSIG
call and a SIP dialog.
A call from QSIG to SIP is initiated when a QSIG SETUP message
arrives at the gateway. The QSIG SETUP message initiates QSIG call
establishment, and an initial response message (e.g., CALL
PROCEEDING) completes negotiation of the bearer channel to be used
for that call. The gateway then sends a SIP INVITE request, having
translated the QSIG called party number to a URI suitable for