SIP Tutorial1

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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.

SIP OverviewSIP Educational Series: Part I

March 18th, 2008


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Tekelec Proprietary 2

Speakers

Mike Sies (Moderator)

Product MarketingTekelec

Jiri Kuthan

AVP, Engineering (SIP Technologies)Tekelec


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About Tekelec

Signaling Session Management Number Portability Messaging

Service Mediation Flexible Routing Service Assurance Revenue Assurance

Service Creation

Tekelecs solutions arein 7 of 10 of the worlds

largest wireless carriersand in 6 of 10 of theworlds largest wireline

carriers

TekelecTekelecs solutions ares solutions are

in 7 of 10 of the worldin 7 of 10 of the worldss

largest wireless carrierslargest wireless carriers

and in 6 of 10 of theand in 6 of 10 of theworldworlds largest wirelines largest wireline

carrierscarriers


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Outline

Q1: SIP Overview (March 18th) Introduction

Why SIP

SIP History

Where SIP was born: IETFstandardization

Use cases: voice, messaging,presence & conferencing

SIP definition & standards

SIP with respect to SS7 signaling

Q2: Introduction to SIP Protocol &Procedures (June 3rd)

SIP Architecture SIP Servers, ENUM

SIP protocol & message elements

SIP procedures

Q3: Deploying SIP in Real Networks(Exact date not finalized yet)

Quality of service (QoS)

NATs and Firewalls

PSTN Real-time Protocol (RTP) formultimedia applications

Q4: SIP Security & Services (Exactdate not finalized yet)

SIP security

SIP services Future SIP uses cases


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Introduction: Why SIP?


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What Is SIP? Depends on Who You Are

Visionary: missing piece for running all servicesover IP, including your browser, telephone and coffeemachine. Richer user interface than PSTN.

Productivity/collaboration applications. Work fromanywhere.

VP for Business Development: technology for all-IP-based telephony that allows integration with

Internet services and surpasses investment barriers CFO: reduction of costs by running homogenous all-

IP technology.

Techie: HTTP-like protocol specified in RFC3261 andassociated standards and running similarly like Emailruns over all-IP.



but it is always about ALL-IP.

Services available to all users,on-site, off-site, multi-site,underway, home office, in office.

Single infrastructure for data andvoice.

Improved productivity tools.

Service operation can beoutsourced in a Centrex-likemanner. Like with web/email,single server may host multipledomains for better efficiency.DSL

T1

E1

WaveLAN

PSTN
https://shop.lindows.com/shopping/cartProcessor.php?addItem[0][sku]=DSXXXUSRSIPPHN&addItem[0][fulfillment]=ship



Why SIP?

Challengers: All-IP nature opens up competition and removes investment barriers.

Incumbents: isnt VoIP a cannibalization threat?

No it is a Darwinist test in that well-adapting species profit whereas othersthat dont will disappear.

Whats the adaptation chance?

- Running homogenous all-IP networks greatly reduces cost and increasescompetitiveness.

- If I was an incumbent, I would pay most attention to key assets: access, identity,& retail capability.

Attack other market segments like challengers do.

Why not SkypeTM?

Thats a single party game: too few devices because of proprietarytechnology and reportedly the only party to make $ with Skype is Skype.



SIP Works Quite Fine in 2008

Working standardized technology for running Telephonyover the Internet (and in the future more real-timeapplications, such as messaging, gaming, etc.)

We have today a variety of interoperable equipment:

Clients: hardphone (snom, Cisco, Mitel, Nortel, Avaya, .),softphones (Microsoft, Counterpath, ), dual phones (Nokia),IADs (Linksys, AVM), terminal adapters (Sipura)

Gateways: Tekelec, Cisco, Sonus,

Servers: Tekelec/iptelorg, Oracle/HotSip, Ubiquity/Avaya, ...

Service providers: ISPs (T-Online, Earthlink), ASPs(SIPphone, Vonage), fixed-mobile-convergence providers

(Telio, Truvoice)



Summary of Introduction

SIP as of today (2008) is a mature all-IP technology andis embedded in a wide array of equipment from variousvendors.

Todays ISP/ASP market is moving to mobile markets. The cost-savings promise holds, applications are

coming slowly.

Learning and integration efforts are considerable.Troubleshooting in distrubuted networks still takes deepexpertise.

Upcoming concerns: security and unsolicitedcommunication.



Poll Question #1

How do you rate your SIP knowledge?

a) Expert

b) Intermediate

c) Basic

d) None


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IETF Standardization


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Where SIP Was Born

Internet Engineering Task Force(IETF www.ietf.org) is astandardization body, which hascreated a large variety of Internetprotocols:

TCP/IP for interconnection SMTP for E-email FTP for data transfer RTP for voice, etc.

Participation is open: participants from both data and telecom industryparticipate, as well as from academia. Contribution comes fromindividuals (as opposed to companies or institutions).

SIP was developed by the IETF (within the Real-Time Applications andInfrastructure area) and has grown to encompass several offshoots:

SIP (core SIP protocol)

SIPPING (telephony and multimedia applications and extensions)

SIMPLE (SIP-based instant messaging and presence)
http://www.ietf.org/http://www.ietf.org/http://www.ietf.org/



MobileNetworks

3GPPand 3GPP2

(defined IMS)

Cable/Broadband

CableLabs(defined

PaketCable)

WirelineNetworks

ETSI, ITUand ATIS

(defined TISPAN)

SIPs Migration as a Standard

SIP has successfully migrated tobecome the global signalingprotocol that underpins broadarchitectural efforts (especiallythe IP Multimedia Subsystem orIMS) for wireless, wireline andbroadband networks

SIP, SIP-T, RTPand many

other protocols



How IETF Working Groups Relate(Abbreviated Listing)

Core Specs SupportingSpecsApplicationSpecs

SIPCore

SignalingProtocol

Specifications

SIPPING

SIP Extensions(Filter for CoreSIP Working

Group)

AVT

Voice andReal-time

Transport

mmusic

Conferencingutilities

iptel

SupportingProtocols

ENUM

IP-PSTN

Numbering

XCON

CentralizedConferencing

GeoPriv

Locationand Related

Privacy

SIMPLE

InstantMessaging

and Presence

Chaired byRobert Sparks

Chaired byRobert Sparks

Chaired byAdam Roach



S in SIP Doesnt Stand for Simple


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SIP Use Cases



SIP Use Cases

Traditional PBX services Some reproducible with SIP 1:1, such as call forwarding

Some hard to reproduce, such as call parking

Some reproducible in different ways, such as voicemailby E-mail

Meshed SIP-based Communications Instant messaging and presence

Web integration such as click-to-dial

Killer applications and other nonsense is out-of-scope of this tutorial.



Meshed SIP-based Communications

Use the same signalingvehicle for more services.

SIP already supports: Notion of user location

Application-layer logic (incl.forking, redirection, etc.)

Security: authentication

The applicationsdemanded & deployed aremostly about serviceintegration:

E-mail: replacement of IVRannoyance with voicemail-2-e-mail

Web: read list of missed callsfrom your webpage (both off-line

& on-line) Web: online phonebook, click-

to-dial Instant Messaging and

Presence, Notification services

(T-storm alarm), SMS delivery Telephony: conferencing

Technical challenge: makeservice programming easy

Example: Voicemail-2-emailserver w/video



Poll Question #2

Which SIP-based services have you deployed or do youplan to deploy in the next 6-12 months? (you can selectmore than one response)

a) VoIP

b) Push-to-x (talk, video, etc.)

c) Presence

d) Messaging

e) Conferencing

f) Fixed mobile convergence (FMC)

g) Other

h) None


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Essential SIP Definitions


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RFC 3261

RFC3261 is the key specification for SIP technology

It establishes the SIP elements (User Agents, Proxyservers), their functions and topologies, and theprotocol they use to speak with each other.

SIP is a text-based, request-reply, client-server

protocol. The next slide shows an RFC3261 call-flow: two SIP

phones establish a connection through a SIP proxy

server. The proxy server implements an authenticationpolicy and locates the call recipient.


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Basic SIP Call-Flow (Proxy Mode)

[email protected]:[email protected]

Location Database

Proxy

INVITE

sip:[email protected]:

sip:[email protected];tag=12To:

sip: [email protected]:

[email protected]#4

DNS SRV Query ? iptel.org

#0Reply: IP Address of iptel.org SIP Server

INVITE

sip:[email protected]:


sip: [email protected]

Call-ID: [email protected] #1

[email protected]

#3jir

i

#2

OK 200From:


sip: [email protected];tag=34

Call-ID:

[email protected]

#5K 200From:


sip: [email protected];tag=34

Call-ID:

[email protected]

#6

ACK sip:[email protected]

Media streams #8

SIP Proxy looks up next hops for requests to served users in locationdatabase and forwards the requests there.


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All-IP Protocol Zoo (Hourglass Model)

UDPSCTPTCP

DNSSIP RADIUS

AALx

GPRS V.xSONETEthernet ATM

PPP

IPv4/IPv6

HTTP RTP STUN

WWW signaling interdomain AAA media NAT

TLS

iLBC, G.711, ...ENUM


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Minimum SIP Setup

Protocols

SIP for signaling

RTP for voiceDNS for finding SIPserver

Database for back-end

services

Boxes

SIP end-devices

PSTN gateways

Database

SIP server

Web-based provisioning


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For Production Setup Add

CDR processing External authentication and authorization server (RADIUS, HSS)

NTP for accurate CDRs

ICE, STUN and TURN servers for NAT traversal Failover protocols (VRRP, UCARP, )

For mobile environment: signaling compression

Applications: message store, presence server, conferencingserver, ...

SNMP for monitoring

SIP extensions for 911, identity management, SIP-T for PSTNtrunking, REFER for CTI

See draft ietf sip hitchhikers guide for a complete overview of SIP-

related technologies
http://www.ietf.org/internet-drafts/draft-ietf-sip-hitchhikers-guide-01.txthttp://www.ietf.org/internet-drafts/draft-ietf-sip-hitchhikers-guide-01.txt


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Interworking with PSTN


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PSTN: What are the Challenges?

If SIP can do it all, why bother with PSTN? Well thereare hundreds of millions of PSTN users.

How do you convert SIP calls to PSTN calls?

Function provided by the PSTN gateways

How do you find the right gateway?

SIP proxy servers with specialized routing logic How do you convey caller ID?

Asserted Identity header fields

How do you convey legacy signaling? DTMF, overlap dialing,


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Call Flow: SIP to PSTN

Request-URI in the INVITEcontains a Telephone Numberwhich is sent to PSTN Gateway.

The Gateway maps the INVITE toa SS7 ISUP IAM (Initial AddressMessage)

183 Session Progress establishes

early media session so caller hearsRing Tone.

Two way Speech path isestablished after ANM (Answer

Message) and 200 OK

Slide courtesy of Alan Johnston, WorldCom.

ISDN/ISUP: RFC 3398

QSIG: RFC 4497


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Example Scenario: SIP Trunking

Region 1

Region 2

Region 3SSR SSR

SSR SSR

Centralized SIProuting table

SIP servers centrally locate termination soft-switches, and assert caller identity.

Softswitches AKA PSTN gateways convert calls from SIP to PSTN and vice versa.


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Next Steps

SIP II: Introduction to SIPProtocol & Procedures (Jun 3rd)

SIP Architecture

SIP Servers, ENUM SIP protocol & message elements

SIP procedures

Follow-up email Webinar slide deck with self-testquestions & resource info

Registration details for SIP II

Visit www.tekelec.com for: SIP II tutorial registration

4-part SIP signaling whitepaper

series


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Q & A

h k f li i


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Thanks for listening!

Jiri Kuthan

AVP, Engineering

SIP Technologies

[email protected]
mailto:[email protected]:[email protected]

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