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October 27, 2009

SIP-I & SIP-T


Introduction and Overview

Adam Roach
Principal Engineer

This document is for informational purposes only, and Tekelec reserves the right to change any aspect of the products, features or
functionality described in this document without notice. Please contact Tekelec for additional information and updates. Tekelec. For What's Next.
Speakers

Mike Sies (Moderator)


Global Marketing
Tekelec

Adam Roach
Principal Engineer
Tekelec

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About Tekelec
More than one BILLION mobile, mobile Internet and telephone
subscribers around the world make calls, send text messages, make
mobile payments, and enjoy social networking because of Tekelec
Tekelec’s solutions are in 8 of 10 of the world’s largest wireless
operators and in 6 of 10 of the world’s largest wireline operators

ƒ Global STP leader with 40% market share


ƒ Number Portability leadership with over 91
customers in 32 countries
ƒ Comprehensive SIGTRAN experience
ƒ SIP experts – Chair 3 IETF SIP Working Groups
ƒ Over 150 IAS customers
ƒ Unique end-to-end network visibility with
integrated signaling and performance
management solutions
ƒ Leading next-gen messaging provider: SMS
routing, offload, security and advertising
ƒ 49 new customers in the last ten quarters
worldwide

Tekelec Confidential
Agenda

› What are SIP-I and SIP-T?


› Architectures for mixed-protocol networks
› How ISUP parameters are carried through SIP networks
› How ISUP parameters are mapped to and from SIP header
field values
› Challenges specific to mixed-protocol networks (overlapped
dialing, call forking)

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Survey Question

What is your current level


of knowledge about
these technologies?

1.No or very little SIP


knowledge
2.Some SIP, but don’t
know about SIP-I or
SIP-T
3.Already have some
knowledge about SIP-I
and/or SIP-T

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What is SIP-T?

› IETF-defined specification for mixed-


protocol SIP/ISUP networks
› Defined in RFC 3372, RFC 3398,
RFC 3578, and RFC 3204
› Developed roughly in parallel with last major revision of SIP
(mid-2002)

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What is SIP-I?

› ITU-defined specification for transiting ISUP


and/or BICC between gateways using SIP
networks
› Defined as part of ITU-T Q.1912.5
› Developed in 2004, re-using many of the components
developed as part of the IETF SIP-T effort

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How Does SIP-I Relate to SIP-T?

› Very similar in purpose, almost compatible


› Difference in styles, reflecting different approaches between
IETF and ITU standardization:
ƒ SIP-I specifies rigid architectures, profiles, and services; cannot
make use of new SIP features as they are developed
ƒ SIP-T focuses on the tools for mapping core protocol constructs,
leaves feature mapping at the discretion of equipment
manufacturers
› SIP-I includes mapping for BICC and ISUP; SIP-T considers
only ISUP
› SIP-T allows calls to originate and terminate at SIP endpoints,
not just gateways

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Supported Use Cases

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Survey Question

What level of interoperability


have you experienced with
SIP-I and/or SIP-T products?

1.No problems
2.Some minor issues, but
things basically work
3.Major issues, but still
deploying
4.Interop is enough of an
issue that we’re delaying or
canceling deployments
5.No experience yet

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Basic Mixed Protocol Architecture

SIP Application Servers

SIP
SIP
PSTN
PSTN PSTN
PSTN
Network
Network

PSTN Gateway PSTN Gateway

PSTN Phone PSTN Phone

SIP-T Only
SIP Phone SIP Phone SIP Phone

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Overview of Operation: PSTN to PSTN

Performs service-
SIP Message related SIP
With Original Modifies attached
message
ISUP Message ISUP message
Modifications
Attached to match changes
SIP Application Servers made to SIP
ISUP message
Message

SIP
SIP
PSTN
PSTN PSTN
PSTN
Network
Network

PSTN Gateway PSTN Gateway

PSTN Phone Modified PSTN Phone


ISUP
Message
Maps from ISUP
parameters to
SIP Header

SIP Phone SIP Phone SIP Phone

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Overview of Operation: PSTN to SIP

Performs service-
SIP Message related SIP
With Original message
ISUP Message Modifications
Attached
SIP Application Servers
ISUP
Message

SIP
SIP
PSTN
PSTN PSTN
PSTN
Network
Network

PSTN Gateway PSTN Gateway

PSTN Phone PSTN Phone

Maps from ISUP


parameters to
SIP Header Ignores
attached ISUP
SIP Phone SIP Phone SIP Phone Message

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Overview of Operation: SIP to PSTN

Performs service-
related SIP
message Creates new ISUP
Modifications message, based on
SIP Application Servers content of SIP
message

SIP
SIP
PSTN
PSTN PSTN
PSTN
Network
Network

PSTN Gateway PSTN Gateway

PSTN Phone ISUP PSTN Phone


Message
SIP Message
(No ISUP
Attachment)
SIP Phone SIP Phone SIP Phone

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Survey Question

What have you deployed or


do you plan to deploy?

1.Neither SIP-I nor SIP-T


2.SIP-I only
3.SIP-T only
4.Both SIP-I and SIP-T
5.One or both, but not
certain which

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Anatomy of a SIP Message

INVITE sip:2145550444@tekelec.com SIP/2.0


Via: SIP/2.0/UDP 172.17.1.247:2078;branch=z9hG4bK-6vi6sa58smfx;rport
To: "Robert Sparks" <sip:2145550444@tekelec.com>
Header From: "Adam Roach" <sip:2145550500@tekelec.com>;tag=4at3wehz8c
Fields Call-ID: 3c58339ed1f6-lvfoul2ixa8h
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:2145550500@172.17.1.247:2078>
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 168
Header
v=0 Body
o=- 1411917766 1411917766 IN IP4 172.17.1.247
c=IN IP4 172.17.1.247
t=0 0
m=audio 61586 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000

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Anatomy of an ISUP Message

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ISUP Message Tunneling (Ingress)

INVITE sip:2145550444@tekelec.com SIP/2.0


Via: SIP/2.0/UDP 172.17.1.247:2078;branch=z9hG4bK-6vi6sa58smfx;rport
To: "Robert Sparks" <sip:2145550444@tekelec.com>
From: "Adam Roach" <sip:2145550500@tekelec.com>;tag=4at3wehz8c
Call-ID: 3c58339ed1f6-lvfoul2ixa8h
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:2145550500@172.17.1.247:2078>
Accept: multipart/mixed, application/sdp
Content-Type: multipart/mixed; boundary=sdlitskeiut-x
Content-Length: 307 Header
Body
--sdlitskeiut-x
Content-Type: application/sdp

v=0
o=- 1411917766 1411917766 IN IP4 172.17.1.247
c=IN IP4 172.17.1.247
t=0 0
m=audio 61586 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
--sdlitskeiut-x
Content-Type: application/isup; version=nxv3; base=etsi121
Content-Disposition: signal; handling=optional

PSTN Gateway

--sdlitskeiut-x--

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ISUP Message Tunneling (Egress)
Note: SIP Service Execution
INVITE sip:9725551111@tekelec.com SIP/2.0
has changed destination
Via: SIP/2.0/UDP 172.17.1.247:2078;branch=z9hG4bK-6vi6sa58smfx;rport
To: "Robert Sparks" <sip:2145550444@tekelec.com>
From: "Adam Roach" <sip:2145550500@tekelec.com>;tag=4at3wehz8c
Call-ID: 3c58339ed1f6-lvfoul2ixa8h
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:2145550500@172.17.1.247:2078>
Accept: multipart/mixed, application/sdp
Content-Type: multipart/mixed; boundary=sdlitskeiut-x
Content-Length: 307 Header
Body
--sdlitskeiut-x
Content-Type: application/sdp

v=0
o=- 1411917766 1411917766 IN IP4 172.17.1.247
c=IN IP4 172.17.1.247
t=0 0
m=audio 61586 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
--sdlitskeiut-x
Content-Type: application/isup; version=nxv3; base=etsi121
Content-Disposition: signal; handling=optional

PSTN Gateway

--sdlitskeiut-x--

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Three Main Mappings

› Mapping between message types


› Mapping between SIP header fields and ISUP parameters
› Mapping between ISUP cause codes and SIP response codes

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Message Type Mapping

› Driven by call state


› SIP-T defines formal state machine for mapping
› Semantics of message types should be very similar
› Several messages have no mapping (e.g., SIP OPTIONS,
ISUP CCR)

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High-Level Message Mapping

† INFO can also be used for SUS and RES under certain circumstances.
SIP-I uses INFO for all SUS and RES messages, since interworking
with native SIP terminals is out of scope.

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Message Mapping: Example Call Setup

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Message Mapping: Mid-Call Interaction

Note: SIP-I uses


INFO instead of
INVITE

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Header/Parameter Mapping

› Focused predominately on mapping in IAM/INVITE translation


› In practice, most mapping relates to calling and called party
identity
› Limited support for SIP indication of number portability, carrier
selection
› Requires several mandatory parameters to be provisioned with
default values in gateways (e.g., Nature of Connection,
Transmission Media Requirement)

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High-Level IAM Parameter Mapping

† The “P-Asserted-Identity:” and “Identity:” header fields will be used for this
purpose under certain circumstances.
‡ Mapping specified by SIP-I only; requires a priori knowledge of network
topology

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Response Code / Cause Code Mapping

› Convey reason for call setup failure when error is at called


party end
› SIP-I and SIP-T take radically different approaches
ƒ SIP-T attempts to map to semantically similar code
ƒ SIP-I maps almost all ISUP codes to SIP “500 Server Error”,
almost all SIP codes to ISUP “127 – Interworking, Unspecified”
› Mapping is not necessarily reversible (e.g., ISUP 34 → SIP
503 → ISUP 41)
› In practice, many products make this mapping configurable,
using either SIP-I or SIP-T table as default

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Representative Mapping: SIP to ISUP

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Representative Mapping: ISUP to SIP

† If no new number is provided by the ISUP diagnostic code, SIP-T will use
“410 gone” – the use of 301 allows automatic recovery in the SIP network.

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Response Code Mapping: Call Flow

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Challenges in SIP/ISUP Networks

› Overlapped Dialing
› Early Media
› Call Forking
› Service Parity

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Overlapped Dialing

› In locations with variable-length numbering plans, called party


number may be sent in multiple ISUP messages (IAM followed
by one or more SAM)
› SIP generally expects complete addresses in INVITE
messages
› No problem for SIP-originated calls
› SIP-T and SIP-I define complicated procedures for handling
overlapped dialing in ISUP-to-ISUP and ISUP-to-SIP calls
› Generally, requires that SIP service proxies understand
overlapped dialing procedures
› Problem can be bypassed, with degraded user experience, by
using digit collection timers

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Overlapped Dialing: Very Simple Example

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Early Media

› SIP generally expects that the session will be established before


media is passed
ƒ Tones and errors are indicated by the calling party device
ƒ SIP does specify that clients should receive and play out media prior to
call establishment to avoid clipping
› ISUP generally anticipates the ability to send media down a circuit as
soon as it is seized
ƒ Call progress tones are often sent by the called party end office
ƒ Many legacy IVRs take advantage of this by not sending an ACM until
(and unless) the call reaches a human
› In other words, SIP provides a “best effort” attempt at passing media
prior to the call, while ISUP has an absolute requirement
› This leads to additional signaling in SIP to establish more reliable
early media channel
ƒ This additional signaling is not supported by all clients

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Call Forking

› One feature inherent in SIP is the idea of “forking” a single


INVITE message to multiple destinations simultaneously
› ISUP has no analog to this behavior – such services require
servers which effectively terminate and re-originate the call
› Forked calls that reach multiple gateways will result in two
disparate sets of ISUP messages, which cannot simply be
merged
› Forked calls can also result in multiple early media sessions
› SIP/ISUP gateways must recognize such situations and
provide sensible behavior to the PSTN side
› Neither SIP-I nor SIP-T provide explicit guidance here;
equipment providers need to handle on a case-by-case basis

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Service Parity

› Many services already enabled by current mapping (e.g.,


Calling Party Identification, Anonymous Call Rejection)
› More advanced services need further definition (e.g.,
Automatic Call Completion)
› Ongoing work to define and document interworking of services
between SIP and ISUP/TCAP networks:
ƒ IETF BLISS – Relatively new working group formed to formalize
specific SIP mechanisms for services already deployed in the
PSTN
ƒ ETSI TISPAN – Interfaces with IETF to provide requirements for
specified services; ensures BLISS-specified SIP services work
with PSTN

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Summary

› SIP-I and SIP-T are two similar approaches that define


interworking between SIP and ISUP networks
ƒ Support different use cases (SIP-T interworks with native SIP
terminals; SIP-I includes mapping for BICC)
› Designed to allow ISUP to pass transparently through a SIP
network if both endpoints are ISUP
› Allow deployment of SIP services for use in both SIP/IMS and
PSTN/PLMN networks
› Fully defined and widely deployed for basic calling, some class
services
› Ongoing work to add remaining class services for mixed-
network interworking

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Next Steps…

› Follow-up email
ƒ Webinar slide deck
ƒ Recording archive
› Visit www.tekelec.com:
› Archived webinars
› Whitepaper downloads
› SSR Application Handbook
› SIP and SS7 Pocket Guides
› SIP Sessions blog
ƒ blog.tekelec.com

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Q&A and Contact Information

Adam Roach
Principal Engineer
Office: +1.214.329.0491
Mobile: +1.972.248.7835
adam.roach@tekelec.com

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