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Tekelec. For What's Next.
Adam Roach
Principal Engineer
October 27, 2009
SIP-I & SIP-T
Introduction and Overview
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Mike Sies (Moderator)
Global Marketing
Tekelec
Adam Roach
Principal Engineer
Tekelec
Speakers
About Tekelec
Global STP leader with 40% market share
Number Portability leadership with over 91
customers in 32 countries
Comprehensive SIGTRAN experience
SIP experts Chair 3 IETF SIP Working Groups
Over 150 IAS customers
Unique end-to-end network visibility with
integrated signaling and performance
management solutions
Leading next-gen messaging provider: SMS
routing, offload, security and advertising
49 new customers in the last ten quarters
worldwide
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
Tekelecs solutions are in 8 of 10 of the worlds largest wireless
operators and in 6 of 10 of the worlds largest wireline operators
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 dont
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:
SIP-I specifies rigid architectures, profiles, and services; cannot
make use of new SIP features as they are developed
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 were delaying or
canceling deployments
5.No experience yet
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SIP-T Only
Basic Mixed Protocol Architecture
SIP
Network
SIP
Network
PSTN
PSTN
PSTN
PSTN
PSTN Phone
SIP Phone
PSTN Phone
PSTN Gateway PSTN Gateway
SIP Application Servers
SIP Phone SIP Phone
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Overview of Operation: PSTN to PSTN
SIP
Network
SIP
Network
PSTN
PSTN
PSTN
PSTN
PSTN Phone
SIP Phone
PSTN Phone
PSTN Gateway PSTN Gateway
SIP Application Servers
SIP Phone SIP Phone
ISUP

Message
SIP Message

With Original

ISUP Message

Attached
Performs service-

related SIP

message

Modifications
Maps from ISUP

parameters to

SIP Header
Modifies attached

ISUP message

to match changes

made to SIP

message
Modified

ISUP

Message
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Overview of Operation: PSTN to SIP
SIP
Network
SIP
Network
PSTN
PSTN
PSTN
PSTN
PSTN Phone
SIP Phone
PSTN Phone
PSTN Gateway PSTN Gateway
SIP Application Servers
SIP Phone SIP Phone
ISUP

Message
SIP Message

With Original

ISUP Message

Attached
Performs service-

related SIP

message

Modifications
Maps from ISUP

parameters to

SIP Header
Ignores

attached ISUP

Message
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Overview of Operation: SIP to PSTN
SIP
Network
SIP
Network
PSTN
PSTN
PSTN
PSTN
PSTN Phone
SIP Phone
PSTN Phone
PSTN Gateway PSTN Gateway
SIP Application Servers
SIP Phone SIP Phone
SIP Message

(No ISUP

Attachment)
Performs service-

related SIP

message

Modifications
Creates new ISUP

message, based on

content of SIP

message
ISUP

Message
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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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Header
Body
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>

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: application/sdp

Content-Type: application/sdp

Content-Length: 168

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
Header
Fields
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Anatomy of an ISUP Message
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Header
Body
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-l vfoul2ixa8h
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
--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
--sdlitskeiut-x--
ISUP Message Tunneling (Ingress)
PSTN Gateway
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Header
Body
INVITE sip:9725551111@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-l vfoul2ixa8h
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
--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
--sdlitskeiut-x--
ISUP Message Tunneling (Egress)
PSTN Gateway
Note: SIP Service Execution
has changed destination
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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
SIP-T attempts to map to semantically similar code
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
Tones and errors are indicated by the calling party device
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
Call progress tones are often sent by the called party end office
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
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:
IETF BLISS Relatively new working group formed to formalize
specific SIP mechanisms for services already deployed in the
PSTN
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
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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Follow-up email
Webinar slide deck
Recording archive

Visit

www.tekelec.com:

Archived webinars

Whitepaper downloads

SSR Application Handbook

SIP and SS7 Pocket Guides

SIP Sessions blog
blog.tekelec.com
Next Steps
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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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