hgs/SIP Services 2001 Keynote 1

Why SIP?
Henning Schulzrinne
Dept. of Computer Science
Columbia University
New York, New York
(sip:)[email protected]

SIP Services and Applications – Washington, D.C.

April 20th, 2001

February 2001
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Overview

• A brief history
• Service models

• SIP design principles

• Extensions in progress
• Potential hazards

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Whence SIP?

Feb. 1996: earliest Internet drafts

Feb. 1999: Proposed Standard

March 1999: RFC 2543
April 1999: first SIP bake-off

November 2000: SIP accepted as 3GPP signaling protocol

December 2001: 6th bake-off, 200+ participants
March 2001: 7th bake-off, first time outside U.S.

February 2001
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SIP years

Year development trade rags

1996-1998 R&D “academic exercise’, “distraction from H.323”
1999 standard & skunk works “what does SIP stand for again?”
2000 product development “SIP cures common cold!”
2001 pioneer deployment “Where are the SIP URLs?”
2002 kmart.com/sip SIP product comparisons

February 2001
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VoIP signaling architectures

• master-slave ➠ MGCP, Megaco

• (mostly) single administrative domain ➠ H.323

• peer-to-peer, cross domain ➠ SIP

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Master-Slave Architecture

• master-slave: MGC controls one or more gateways

• allows splitting of signaling and media functionality
• “please send audio from circuit 42 to 10.1.2.3”
• uses MGCP (implemented) or Megaco/H.248 (standardized, but just beginning to
be implemented)
• gateway can be residential
• basis of PacketCable NCS (network control system) architecture
• service creation similar to digital PBX or switch
• end system has no semantic knowledge of what’s happening
• −→ can charge for caller id, call waiting

February 2001
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VoIP architectures

SIP H.323 Megaco/MGCP

multiple domains x ? –
Third-party control x – single-domain
multimedia x fixed set not likely
end system control x x –
extensible x ? limited
generic events x – –
cgi scripting x – –
servlets x – –
CPL x x –

February 2001
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SIP inheritance

• URLs:
– general references (“forward to email”)
– recursive embeddding
• HTTP:
– basic request/response format, status codes, . . .
– proxies (but no caching)
– cgi programming interface
• email/SMTP:
– addressing
– MX −→ SRV records for load balancing, redundancy
– header/body separation, MIME

February 2001
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SIP design choices

Transport protocol neutrality: run over reliable (TCP, SCTP) and unreliable (UDP)
channels, with minimal assumptions

Request routing: direct (performance) or proxy-routed (control)

Separation signaling vs. media description: can add new applications or media
types, SDP −→ SDPng
Extensibility: indicate and require proxy and UA capabilities

February 2001
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Personal mobility

[email protected]
(also used by [email protected])

yahoo.com

tel:12128541111
[email protected]

[email protected]

[email protected]

columbia.edu tel:12015551234
[email protected]

[email protected]

February 2001
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Example: Columbia CS phone system

Expand existing PBX via IP phones, with transparent connectivity

MySQL
Cisco 7960 user database sipconf rtspd
LDAP server
conferencing RTSP
server media
(MCU) server

Sun Solaris RTSP

PC Linux/FreeBSD/NT

unified
messaging
server

Nortel Cisco sipd sipum

Meridian 2600 proxy/redirect server

PBX
T1/E1
RTP
SIP

PhoneJack interface

e*phone
sipc

SIP−H.323
converter
plug’n’sip
sip−h323

802.11b
wireless

February 2001
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Events as universal glue

• currently, don’t have general event notification in the Internet

• email is too slow: pull on the last hop (server to user)

• generic problem:
– “voicemail has arrived”
– “called party is reachable”
– “new configuration data available”
– “IR sensor has detected movement”
– “boiler temperature above threshold”
– ...
• same delivery (SIP), different data (XML DTDs)

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SIP as a presence & event platform

• minimal SIP extension: SUBSCRIBE to request notifcations, NOTIFY when

event occurs
• also, MESSAGE for IM, sessions for multi-party chats
• transition to true “chat” (and video)

• services such as reaching mobile phone while in meeting

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Events: SIP for appliances

SUBSCRIBE [email protected]

NOTIFY [email protected]
SIP proxy
SIP user agent DO [email protected] (RGW)

INVITE [email protected]

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SIP service architectures

classical: Media and signaling in one box

distributed: request routing and coordination, with service components (storage,
IVR, location, . . . )

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Challenges and obstacles

• scalable device configuration

• PSTNv3

• “walled garden”
• service infrastructure
• standardization

• invisible Internet telephony

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Device configuration

• need to plug in store-bought phone, without more than personalization

• limited user interface

• configuration from local (visited) network and from home network

• don’t want current PBX single-vendor tie-ins
• cannot rely on California-style upgrades

• notifications of new configurations ➠ SUBSCRIBE/NOTIFY

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Device configuration

visited network
visited.net

DHCP
IP address, router
DNS domain, server
SIP outbound proxy
tftp server

tftp
SIP boot image

SIP timers home network

SIP preloaded routes [email protected]

address book
CPL scripts
dialplan

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Potential obstacles

• SIP as transport – for legacy signaling

– due to proxies, UDP not designed for volume data
– doesn’t add significant value
• NATs and firewalls – can engineer around them, but ugly
– leads to IP-over-HTTP solutions, defeating firewall
– proxy boxes outside NATs

February 2001
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PSTN legacies to avoid

• E.164 numbers – might as well wear bar codes

• overlap dialing
• tones and announcements
• in-band signaling for features (DTMF)
• systems with user-interface knowledge (12 keys, voice)
• voice-only orientation (BICC, MGCP/Megaco)
• integration of bit transport and services
• service-specific billing ➠ separate signaling & billing
• trusted networks without crypto

➠ confine PSTN knowledge to edge of network

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“Walled garden” model

• 3G wireless carriers adopting SIP, but used to closed services

• SIP users should be able to use any proxy for services, not just carrier service
• typical users have many identities (and, thus, servers):

work [email protected]
travel [email protected]
home [email protected]
professional [email protected]

• hard to prevent: SIP can use any port number

• if not, requires draconian restrictions on IP packets, not just filtering port 5060
(SIP port)
• also, services may be split across servers

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So I want to build a SIP network. . .

Ready for trials, but probably not quite for shrink-wrap status:

• installation and operation still requires fair amount of expertise

• lots of web and email experts, few SIP experts

• needs some external infrastructure: DHCP and SRV, possibly AAA
• inconsistent configuration for Ethernet phones (being worked on)

• SIP phones still more expensive than analog phones ➠ hard to justify PBX
replacement (incremental cost)

• no just-download or ship-with-OS “soft” clients

February 2001
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Need for service infrastructure

• need carriers that offer SIP gateways

• without having to provide SS7 connectivity

• with outbound PSTN calling

• with inbound calls and number portability – need to be able to keep old PSTN
numbers
• either IP Centrex model or in-house servers – like ISP services for email or web

• for commercial-grade conferences, need nailed-up Internet connectivity, orderable

(at least) by web page – across providers!

• PBX revenue already decreasing

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Why aren’t we junking switches right now?

What made other services successful?

email: available within self-contained community (CS, EE)

web: initially used for local information

IM: instantly available for all of AOL

All of these . . .

• work with bare-bones connectivity (≥ 14.4 kb/s)

• had few problems with firewalls and NATs

• don’t require a reliable network

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Why aren’t we junking switches right now?

Telephone services are different:

• reliability expectation 99.9% % 99.999%

• PC not well suited for making/receiving calls – most residential handsets are
cordless or mobile

• business sets: price incentive minor for non-800 businesses

• services, multimedia limited by PSTN interconnection
• initial incentive of access charge bypass fading (0.5c/min.)

• international calls only outside Western Europe and U.S.

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Standardization

• SIP working group is one of the most active in IETF

• located in “transport” area, but really an application

• about 80 active Internet drafts related to SIP
• typically, 400 attend WG meetings at IETF

• but few drafts are working group items

• 80-20% – 80% of the technical work takes 20% of the time

February 2001
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Invisible Internet telephony

“VoIP” technology will appear in

• Internet appliances

• home security cameras, web cams

• 3G mobile terminals

• fire alarms and building sensors

• chat/IM tools
• interactive multiplayer games

• 3D worlds: proximity triggers call

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Conclusion

• SIP maturing – base stable, extension in progress

• avoid creating PSTN replica

• leverage, not inhibit, Internet flexibility
• significant deployment challenges remain

February 2001
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For more information. . .

SIP: http://www.cs.columbia.edu/sip

RTP: http://www.cs.columbia.edu/˜hgs/rtp
Papers: http://www.cs.columbia.edu/IRT

February 2001