SIP Overview
SIP Educational Series: Part I
March 18th, 2008
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.
Speakers
Mike Sies (Moderator) Product Marketing Tekelec
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 are Tekelecs solutions are in 7 of 10 of the worlds in 7 of 10 of the worlds largest wireless carriers largest wireless carriers and in 6 of 10 of the and in 6 of 10 of the worlds largest wireline worlds largest wireline carriers carriers
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Outline
Q1: SIP Overview (March 18th)
Introduction Why SIP SIP History Where SIP was born: IETF standardization Use cases: voice, messaging, presence & conferencing SIP definition & standards SIP with respect to SS7 signaling
Q3: Deploying SIP in Real Networks (Exact date not finalized yet)
Quality of service (QoS) NATs and Firewalls PSTN Real-time Protocol (RTP) for multimedia applications
Q2: Introduction to SIP Protocol & Procedures (June 3rd)
SIP Architecture SIP Servers, ENUM SIP protocol & message elements SIP procedures
Q4: SIP Security & Services (Exact date not finalized yet)
SIP security SIP services Future SIP uses cases
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Introduction: Why SIP?
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.
What Is SIP? Depends on Who You Are
Visionary: missing piece for running all services over IP, including your browser, telephone and coffee machine. Richer user interface than PSTN. Productivity/collaboration applications. Work from anywhere. VP for Business Development: technology for allIP-based telephony that allows integration with Internet services and surpasses investment barriers CFO: reduction of costs by running homogenous allIP technology. Techie: HTTP-like protocol specified in RFC3261 and associated standards and running similarly like Email runs over all-IP.
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but it is always about ALL-IP.
PSTN
E1 WaveLAN T1
DSL
Services available to all users, on-site, off-site, multi-site, underway, home office, in office. Single infrastructure for data and voice. Improved productivity tools. Service operation can be outsourced in a Centrex-like manner. Like with web/email, single server may host multiple domains for better efficiency.
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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 others that dont will disappear. Whats the adaptation chance?
- Running homogenous all-IP networks greatly reduces cost and increases competitiveness. - 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 proprietary technology and reportedly the only party to make $ with Skype is Skype.
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SIP Works Quite Fine in 2008
Working standardized technology for running Telephony over the Internet (and in the future more real-time applications, 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)
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Summary of Introduction
SIP as of today (2008) is a mature all-IP technology and is embedded in a wide array of equipment from various vendors. 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 deep expertise. Upcoming concerns: security and unsolicited communication.
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Poll Question #1
How do you rate your SIP knowledge?
a) b) c) d) Expert Intermediate Basic None
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IETF Standardization
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.
Where SIP Was Born
Internet Engineering Task Force (IETF www.ietf.org) is a standardization body, which has created a large variety of Internet protocols:
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 industry participate, as well as from academia. Contribution comes from individuals (as opposed to companies or institutions). SIP was developed by the IETF (within the Real-Time Applications and Infrastructure 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)
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SIPs Migration as a Standard
Mobile Networks
3GPP and 3GPP2 (defined IMS)
Wireline Networks
ETSI, ITU and ATIS (defined TISPAN)
Cable/ Broadband
CableLabs (defined PaketCable)
SIP, SIP-T, RTP and many other protocols
SIP has successfully migrated to become the global signaling protocol that underpins broad architectural efforts (especially the IP Multimedia Subsystem or IMS) for wireless, wireline and broadband networks
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How IETF Working Groups Relate
(Abbreviated Listing)
Supporting Specs mmusic
Conferencing utilities Instant Messaging and Presence
Core Specs SIPPING
SIP Extensions (Filter for Core SIP Working Group)
Application Specs SIMPLE
Chaired by Robert Sparks
SIP
Core Signaling Protocol Specifications
iptel
Supporting Protocols
GeoPriv
Location and Related Privacy
Chaired by Robert Sparks
AVT
Voice and Real-time Transport
ENUM
IP-PSTN Numbering
XCON
Centralized Conferencing
Chaired by Adam Roach
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S in SIP Doesnt Stand for Simple
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SIP Use Cases
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 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 voicemail by 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.
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Meshed SIP-based Communications
Use the same signaling vehicle for more services. SIP already supports:
Notion of user location Application-layer logic (incl. forking, redirection, etc.) Security: authentication
The applications demanded & deployed are mostly about service integration:
E-mail: replacement of IVR annoyance with voicemail-2-email Web: read list of missed calls from your webpage (both off-line & on-line) Web: online phonebook, clickto-dial Instant Messaging and Presence, Notification services (T-storm alarm), SMS delivery Telephony: conferencing
Example: Voicemail-2-email server w/video
Technical challenge: make service programming easy
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Poll Question #2
Which SIP-based services have you deployed or do you plan to deploy in the next 6-12 months? (you can select
more than one response)
a) b) c) d) e) f) g) h)
VoIP Push-to-x (talk, video, etc.) Presence Messaging Conferencing Fixed mobile convergence (FMC) Other None
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Essential SIP Definitions
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.
RFC 3261
RFC3261 is the key specification for SIP technology It establishes the SIP elements (User Agents, Proxy servers), their functions and topologies, and the protocol 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 authentication policy and locates the call recipient.
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Basic SIP Call-Flow (Proxy Mode)
SIP Proxy looks up next hops for requests to served users in location database and forwards the requests there.
Location Database #0 DNS SRV Query ? iptel.org Reply: IP Address of iptel.org SIP Server INVITE sip:
[email protected] From: sip:
[email protected];tag=12 To: sip:
[email protected] #1 Call-ID:
[email protected] [email protected] #2 jiri #3 INVITE sip:
[email protected] From: sip:
[email protected];tag=12 To: sip:
[email protected] Call-ID:
[email protected]#4
#6
OK 200 From: sip:[email protected];tag=12 To: sip: [email protected];tag=34 Call-ID: [email protected] #7 ACK sip:[email protected]
Proxy
[email protected]
sip:[email protected]
Media streams
#8
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All-IP Protocol Zoo (Hourglass Model)
ENUM
WWW signaling interdomain AAA
iLBC, G.711, ...
media NAT
HTTP
TLS
SIP TCP
DNS SCTP IPv4/IPv6 PPP
RADIUS UDP
RTP
STUN
AALx V.x ATM
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Ethernet
GPRS
SONET
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Minimum SIP Setup
Protocols SIP for signaling RTP for voice DNS for finding SIP server 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, conferencing server, ... SNMP for monitoring SIP extensions for 911, identity management, SIP-T for PSTN trunking, REFER for CTI See draft ietf sip hitchhikers guide for a complete overview of SIPrelated technologies
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Interworking with PSTN
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.
PSTN: What are the Challenges?
If SIP can do it all, why bother with PSTN? Well there are 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 INVITE contains a Telephone Number which is sent to PSTN Gateway. The Gateway maps the INVITE to a SS7 ISUP IAM (Initial Address Message) 183 Session Progress establishes early media session so caller hears Ring Tone. Two way Speech path is established after ANM (Answer Message) and 200 OK
ISDN/ISUP: RFC 3398 QSIG: RFC 4497
Slide courtesy of Alan Johnston, WorldCom.
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Example Scenario: SIP Trunking
Region 2
Region 1
SSR
SSR
Region 3
SSR
SSR
Centralized SIP routing 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 SIP Protocol & Procedures (Jun 3rd)
SIP Architecture SIP Servers, ENUM SIP protocol & message elements SIP procedures
Follow-up email
Webinar slide deck with self-test questions & 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
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Thanks for listening!
Jiri Kuthan AVP, Engineering SIP Technologies [email protected]
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