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Contents
BOOKMARK1::Aboutthisinformation productAbout this information product
BOOKMARK2::PurposePurposexvii
BOOKMARK3::ReasonforreissueReason for reissuexviii
BOOKMARK4::SafetylabelsSafety labelsxviii
BOOKMARK5::IntendedaudienceIntended audiencexviii
BOOKMARK6::Howtouse thisinformationproductHow to use this information productxix
The 5ESS®Switch Session Initiation Protocol (SIP) - OA&M Manual,
5E16.2 and Later, 235-200-118, information product enables users,
planners, maintenance personnel, engineers, installers, administrators,
and provisioners to perform the necessary tasks required to support
configuration, installation, monitoring, and repair of SIP signaling for
packet trunking (including packet groups, SIP, SCTP, UDP, IP, QLPS
connectivity, processor groups, and signaling-related hardware).
OA&M procedures related to the OIU-IP bearer for SIP-signaled calls
are covered in Optical Interface Unit - Internet Protocol (OIU-IP)
Interface Specification [5E16.2 and Later Software Releases],
235-900-316.
This information product should be used as the source of complete
details on this protocol to clarify the implementation on the switch
and the interpretation of technical reference (TR) requirements. These
®
details describe the 5ESS
switch offerings in terms of the support of
Session Initiation Protocol (SIP). SIP is an evolving platform in which
new features will be introduced continuously for new revenue
opportunities, for improved operational efficiencies, and for support of
specific applications. Beginning with the 5E16.2 software release, the
5ESS®switch supports the protocols and services defined by Telcordia
Technologies, Inc.
This information product is expected to change as requirements and
standards evolve. Therefore, Lucent Technologies reserves the right to
change or delete any portion of the document, or to add information
in future issues.
Reason for reissue
This information product is being reissued to support feature
99-5E-8645, “DRM IP Trunking”
About this information product
®
Safety labels
Typical safety labels are contained in this information product. The
safety labels include warnings, cautions, and dangers and are
accompanied by icons that indicate the type and level of safety hazard
involved.
CAUTION
Caution indicates the presence of a hazard that can or will
cause minor injury or property damage.
WARNING
Warning indicates the presence of a hazard that can cause
death or severe injury.
DANGER
Danger indicates the presence of a hazard that will cause
death or severe injury.
The 5ESS®Switch Session Initiation Protocol for Packet Trunking OA&M Manual describes the architecture, engineering, provisioning,
®
and maintenance of SIP on the 5ESS
switch. It is published as a
guide for the users, planners, maintenance personnel, engineers,
installers, administrators, and provisioners to perform their SIP-related
Lucent Technologies235-200-118
Issue 3.02B, March 2007
,
5E16.2 and Later Software Releases
tasks. Responsible personnel should have a working knowledge of
telephony, switching, routing, and networking technologies.
About this information product
How to use this
information product
Each chapter in this information product groups related information
about the SIP application.
ChapterContent
1. OverviewOverview of SIP from an industry perspective
and Lucent Technologies’ perspective.
2. ArchitectureOverview of the SIP architecture, starting at
the Network view, then narrowing down to the
System view, and finally, focusing on the
hardware view.
3. Call FlowOverview of SIP call flows through the
network (office-to-office), and through the
®
5ESS
4. Engineering
Considerations
Things to keep in mind when engineering SIP
into your switch and network (for example,
capacity, configuration, constraints, IP
addressing schemes, network management,
performance monitoring, and OS impact).
5. ProvisioningProcedures required for provisioning all
aspects of the SIP platform on your switch and
in your network.
switch hardware.
6. DeprovisioningProcedures required to remove the SIP
®
switch.
7. Maintenance
Considerations
application from the 5ESS
Overview of troubleshooting and maintenance
specific to SIP.
8. GlossaryList of acronyms used within this IP and their
expansions.
9. IndexIndex of subjects covered within this IP.
Conventions used
No special or unusual conventions are used in this information
product.
Systems supported
This information product supports software releases 5E16.2, Feature
Release 3, and later.
•Translations & Dynamic Data Reference, 235-600-228
•Dynamic Data Reference, 235-600-237
•Translations & Dynamic Data Domain Description, 235-600-248
•Audits Manual, 235-600-400
•Asserts Manual, 235-600-500
•Input Messages Manuals, 235-600-700
•Output Messages Manuals, 235-600-750
To comment on this information product, go to the Online Comment
Form (http://www.lucent-info.com/comments/enus/) or email your
comments to the Comments Hotline (comments@lucent.com).
The Lucent Technologies Learning Organization in Indianapolis,
Indiana, distributes this information product. It can be ordered by
phone, fax, mail, or on line. Specific ordering information is listed
below. When ordering, refer to information product number
Session Initiation Protocol (SIP) is a method of establishing,
maintaining, and terminating internet sessions. These sessions
interactively exchange real-time multimedia data (voice, text, and
video) between multiple participants. SIP is based on a client-server
model with intelligent end-points. End servers respond to session
initiation requests and locate the called parties. SIP is an application
layer protocol that can run on top of different transport protocols.
The SIP for Packet Trunking - NAR (99-5E-8382) feature adds a SIP
®
packet trunking interface to the 5ESS
®
5ESS
switch to operate as a gateway for interworking between the
switch. This feature allows the
time division multiplexing (TDM)-based public switched telephone
network (PSTN) trunks and the next-generation internet protocol (IP)
network virtual trunks. It interworks current circuit signaling
protocols, such as Signaling System 7 (SS7) integrated services user
part (ISUP), multifrequency (MF), or dual tone multifrequency
(DTMF), with SIP trunk signaling over IP networks.
The SIP signaling protocol for telephony is the packet signaling
protocol used to establish calls using the Optical Interface Unit Internet Protocol (OIU-IP).
The SIP for Packet Trunking feature contains:
•two new PSU2 protocol handlers (SIP-PH and the GQPH) for
setting up the SIP signaling connection,
•a new lower layer transport protocol, Stream Control
Transmission Protocol (SCTP), and
•integrated operations, administration, maintenance and
®
provisioning (OAM&P) using the conventional 5ESS
switch
interfaces.
Automatic ACM Timer - is a modification to the 99-5E-8382 feature
being implemented in software release 5E16.2 FR6. The Automatic
address complete message (ACM) timer is a new field on RC/V 5.82.
The timer value ranges from 4-14 seconds, with 14 seconds as the
default. This timer is started at the originating packet switch (OPS) by
the SIP terminal process when an INVITE message is sent to a
terminating packet switch (TPS). If the timer expires, the SIP
Terminal Process generates a default ISUP ACM that is sent to the
ISUP originating terminal process (OTP). The ISUP OTP will send an
ACM message out to the ISUP network. This prevents the call from
being terminated by the ISUP office because the TPS response was
too slow.
The features that follow enhance the capabilities provided by the SIP
for Packet Trunking feature (99-5E-8382). Feature descriptions for
these features can be found in the Feature Descriptions, 235-190-400
document.
UDP Transport Layer for SIP
The UDP Transport Layer for SIP (99-5E-8581) feature allows the
™
5E-XC
allowing the 5E-XC
to support SIP with a UDP transport layer instead of SCTP,
™
to connect to SIP network elements that do not
support SCTP.
Support for SIP without Preconditions
The Support for SIP without Preconditions (99-5E-8582) feature
allows SIP calls to be established without the precondition procedures
that were initially required by the SIP for Packet Trunking feature. For
additional information about SIP signaling procedures, with or without
preconditions, see the Session Initiation Protocol (SIP) - InterfaceSpecification, 235-900-344 document.
SIP Support for Line to Packet Trunk Calls
The SIP Support for Line to Packet Trunk Calls (99-5E-8583) feature
allows analog and ISDN line originations and terminations to be
routed to and from SIP packet groups, in addition to the circuit trunk
originations and terminations initially supported by the SIP for Packet
Trunking feature.
SIP without Encapsulated ISUP
The SIP without Encapsulated ISUP (99-5E-8587) feature allows the
™
5E-XC
to interwork with elements that do not generate and process
ISUP messages and to generate appropriate interworking messages to
™
pass through the network. The 5E-XC
supports SIP without
Encapsulated ISUP on a per packet group basis.
SIP Enhancements to Support PSTN Gateway Phase 1
The SIP Enhancements to Support PSTN Gateway Phase 1
(99-5E-8658) feature provides a series of enhancements that allow
phone calls to be setup between PSTN subscribers and IP subscribers
that terminate to a Telephone Application Server (TAS). The PSTN
Gateway sits between the IP network (using SIP signaling) and PSTN
network (using ISUP signaling). The PSTN Gateway performs the
interworking between SIP and ISUP protocols.
DRM IP Trunking
The DRM IP Trunking (99-5E-8645) feature extends SIP-T signaling
and OIU-IP packet trunking to the DRM.
Feature descriptions for the features listed above can be found in the
Feature Descriptions, 235-190-400 document.
Benefits
SIP is an emerging IP-based protocol that is critical for deploying
converged and next generation real-time voice, data and video
communication services. SIP for Packet Trunking gives the 5ESS
®
switch direct access to more efficient IP transport networks. It allows
for the utilization of more cost-effective revenue-generating services.
It also can be used to expand trunking capacity or to replace existing
circuit trunks.
SIP for Packet Trunking:
•reuses the embedded network equipment to migrate to an all IP
multimedia network,
•provides high quality service with 99.999% hardware and
software reliability,
®
•reuses existing integrated 5ESS
switch OAM&P interfaces,
•supports inter-operability with other vendors equipment through
development based on industry standards,
•reuses existing switch hardware and software infrastructure,
•interfaces with the TDM-based PSTN trunks or next generation
IP network virtual trunks, and
•can use SCTP to provide additional network security.
Differences
The existing TDM network contains dedicated circuits between two
end-points. IP calls have no fixed connections. Calls are dynamically
routed through the packet network based on available bandwidth. The
connection information is exchanged using SIP signaling.
The migration from an expansive SS7-based signaling transfer point
(STP) network to an IP-based signaling network will be ongoing. As
SIP evolves, more capabilities will be defined and added.
The features that follow are secured features listed with the associated
secured feature IDs (SFIDs). Recent Change view 8.22 is used to
activate these features. Refer to “Feature Activation (RC/V 8.22)”
(5-13) for the procedure.
•SIP for Packet Trunking (SFID 684)
•SIP without Encapsulated ISUP (SFID 769)
The following list provides the availability for SIP related features:
•SIP for Packet Trunking (feature 99-5E-8382) is available with
the 5E16.2 FR3 software release or later.
•UDP Transport Layer for SIP (feature 99-5E-8581) and Support
for SIP without Preconditions (feature 99-5E-8582) are available
with 5E16.2 FR5 software release or later.
•SIP without Encapsulated ISUP (feature 99-5E-8587) and SIP
Enhancements to Support PSTN Gateway Phase 1 (feature
99-5E-8658) are available with the 5E16.2 FR6 software release
or later.
Deployment
Background knowledge
•The DRM IP Trunking (99-5E-8645) feature is available with the
5E16.2 FR 9 software release or later.
This platform is provided on a per office basis.
When SIP is used in media gateway controllers for call establishment,
there are many cases where it must bridge the PSTN network with IP
networks. To do this, SIP must be extended to transport all of the
PSTN signaling information transparently. By extending SIP
messaging and adding PSTN signaling encapsulation functionality, the
SIP protocol can be used for media gateway to media gateway
communication.
Interfacing with a IP network requires an understanding of the
following IP concepts and equipment:
•routers and Layer 2 (L2) switches,
•Ethernet,
•internet protocol and internet control message protocol (ICMP),
•transport protocols like SCTP and user datagram protocol (UDP),
and
•SIP standards and protocols.
Hardware Dependencies
Software Dependencies
The Optical Interface Unit - Internet Protocol (OIU-IP) hardware
(feature 99-5E-8308) needs to be installed and active.
Other hardware required to support this feature includes two types of
protocol handlers (PHs); the PHE2 to support the SIP PH
functionality, and the PH33 to support the General Quad-link Package
Switch (QLPS) PH (GQPH), as well as the following:
•SM2000 switching modules,
•QLPS,
•Packet Switch Unit - Model 2 (PSU2), and
•Data Fanout - Model 2 (DF2).
Note: This QLPS and PH33 are not required on the DRM/VCDX.
Refer to Chapter 4, “Engineering Considerations” for a list of the
required hardware.
The following software components are needed:
•new SIP signaling software supported by SCTP or UDP for
transport,
•software to support the optical facility interface (OFI) circuit
pack hardware delivered with the Optical Interface Unit for NAR
Market feature (99-5E-7140),
•software to support the optical facility interface internet protocol
(OFI-IP) circuit pack hardware delivered with the Optical
Interface Unit - Internet Protocol feature (99-5E-8308), and
•operational support system (OSS) enhancements.
Incompatibilities
Feature Interactions
There are no incompatibilities with other features.
SIP for Packet trunking defines the signaling protocol to control the
bearer transport delivered with the Optical Interface Unit - Internet
Protocol (OIU-IP) feature (99-5E-8308).
SIP for Packet Trunking that uses UDP Transport Layer for SIP
(99-5E-8581) requires Support for SIP without Preconditions
(99-5E-8582). Both features are dependent on the base SIP feature
(99-5E-8382).
The SIP without Encapsulated ISUP feature (99-5E-8587) is
dependent on the base SIP feature (99-5E-8382). The SIP without
Encapsulated ISUP feature is transport independent.
The SIP Enhancements to Support PSTN Gateway Phase 1
(99-5E-8658) is dependent on the base SIP feature (99-5E-8382).
The DRM IP Trunking (99-5E-8645) is dependent on the base SIP
feature (99-5E-8382).
Inter-operability
One of the main objectives of SIP is to provide call signaling and call
control independent of the IP network technology.
Inter-operability with IP equipment
The IP router or Layer 2 (L2) switch to which the 5ESS®switch
protocol handlers (PHs) interface can be from any vendor, provided
that it meets the requirements of the Session Initiation Protocol (SIP)
- Interface Specification, 235-900-344 document.
®
The IP network components and the 5ESS
switch can be configured
many different ways. The different configurations are described in “IP
Router & Layer 2 Switch Interoperability” (2-15).
Inter-operability with other Switching Systems
SIP for Packet Trunking is based on standards for the protocol layers
including SIP, SCTP, and/or UDP, IP, and ICMP, and the software is
intended to operate with other vendors. However, this feature works
®
best when used with other 5ESS
switches. If another vendor’s switch
adheres to the standards in the same way, the products should be
compatible. No assumptions are made about inter-operability with
other vendor’s equipment.
The Session Initiation Protocol (SIP) - Interface Specification,
235-900-344 document explains the interface implementation in
greater detail.
Customer Security
It is the customer’s responsibility to ensure the security of the IP
backbone network. It is assumed that the IP layer security is
SIP for Packet Trunking is based on standards by the Internet
™
Engineering Task Force (IETF) and Telecordia
Technologies.
SIP standards continue to be defined and extended. Some standards
are in draft form and, in some cases, incomplete or not well defined.
In some instances, standards could not be followed simply because no
standard exists.
IETF RFCTitle
RFC 3261SIP: Session Initiation Protocol
RFC 2960Stream Control Transmission Protocol
RFC 768User Datagram Protocol
RFC 791Internet Protocol
RFC 792Internet Control Message Protocol
RFC 1122Requirements for Internet Hosts -- Communication
Layers
RFC 1332The PPP Internet Protocol Control Protocol (IPCP)
RFC 1661The Point-to-Point Protocol (PPP)
RFC 1662PPP in HDLC-like Framing
RFC 2474Definition of the Differentiated Services Field (DS
Field) in the IPv4 and IPv6 Headers
RFC 2615PPP over SONET/SDH
RFC 1878Variable Length Subnet Table for IPv4
RFC 1918Address Allocation for Private Internets
RFC 917Internet Subnets
RFC 1519Classless Inter-Domain Routing (CIDR): an Address
Assignment and Aggregation Strategy
RFC 2616Hypertext Transfer Protocol -- HTTP/1.1
RFC 2665Definitions of Managed Objects for the Ethernet-like
Interface Types
RFC 2011SNMPv2 Management Information Base for the Internet
Protocol using SMIv2
RFC 2013SNMPv2 Management Information Base for the User
Datagram Protocol using SMIv2
RFC 3372Session Initiation Protocol for Telephones (SIP-T):