SIP: Status and Directions - Columbia University · SIP: Status and Directions Henning Schulzrinne...

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1 SIP: Status and Directions Henning Schulzrinne Dept. of Computer Science Columbia University New York, New York [email protected] Bell Atlantic January 26, 2000

Transcript of SIP: Status and Directions - Columbia University · SIP: Status and Directions Henning Schulzrinne...

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SIP: Status and DirectionsHenning Schulzrinne

Dept. of Computer ScienceColumbia UniversityNew York, New York

[email protected]

Bell Atlantic

January 26, 2000

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Overview� SIP overview/review

� SIP services

� SIP standardization status

� SIP bake-off

� Columbia Internet telephony activities

� SIP for notification

� SIP for mobility

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Architecture

Megaco/MGCP/MDCP

SIP

RTP

circuit-switched voice

H.323

proxy

MGC

MG

GK GK

proxy

PSTN

gateway

circuit-switched voice (POTS, ISDN)Internet

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SIP 101� SIP = signaling protocol for establishing

sessions/calls/conferences/. . .

� session = audio, video, game, chat, . . .

1. called server may map name touser@host

2. callee accepts, rejects, forward (! new address)

3. if new address, go to step 1

4. if accept, caller confirms

5. . . . conversation . . .

6. caller or callee sendsBYE

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SIP Operation in Proxy Mode

10 media stream

4

[email protected]

8

7

1

[email protected]

9

6

5

3

?

henn

ing

hgs@

play

tune

play

cs.columbia.edu

200 OK

location server

ACK hgs@play

200 [email protected]

cs.tu-berlin.de INVITE hgs@play2

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SIP Operation in Redirect Mode

1

4

32

6

7

8

5

?

henn

ing

ACK [email protected]

INVITE [email protected]

302 Moved temporarily colu

mbi

a.ed

u

locationserver

columbia.edu

hgs

tu-berlin.de

INVITE [email protected]

200 OK

ACK [email protected]

ieee.org

Contact: [email protected]

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SIP Advanced Features� operation over UDP or TCP

� multicast invitations➠ basic ACD

� “interactive web response” (IWR)

� UA $ proxy = proxy/redirect$ proxy/redirect

� stateless proxies: self-routing responses

� forking proxies: call several in sequence and/or parallel

� security: basic (password), digest (challenge/response), PGP

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More SIP Internet Telephony Services� camp-on without holding a line

� short message service (“instant messaging”)

� schedule call into the future

� call with expiration date

� add/remove parties to/from call➠ mesh

� “buddy lists”

� can support CLASS features (see tech report)

� call transfer under discussion

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Internet Telephony – as Part of Internet

� email address = SIP address

� SIP URLs in web pages

� forward to email, web page, chat session, . . .

� include web page in invitation response (“web IVR”)

� RTSP: choose your own music-on-hold

� include vCard, photo URL in invitation

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SIP Security� signaling re-uses existing implementations and protocols�!

bugs less likely:

IPsec hop-by-hop

SSL hop-by-hop

basic SIP plaintext password authentication

p

digest SIP challenge-response

p

pgp SIP PGP encryption/authentication for parts/all

p

� media:

IPsec negotiation via SDP

RTP key in SDP (“k=”)

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Interdomain Communications and Security

� no difference in protocol betwen intra/inter domain

� any proxy server can insist onProxy-Authentication

� intermediaries can sign requests and responses (e.g., “thisphone belongs to a Bell Atlantic employee”)

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SIP Extensibility

� headers that receiver may ignore, e.g.,Photo

� new methods and inquire about those supported (OPTIONS)

� features that receivers needs to understand:Required �!

Unsupported

� e.g.,Required: com.sylantro.feature

� proposed: features supported viaSupported header

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An IETF VoIP Architecture

serveraccountingRADIUS

COPS?

sip-cgi

DIAMETER?

firewall

RTSP

MGCP/MEGACOP

LDAP

RTPMG

CPL

RTSP server

record

SIP

SIP server

message

promptsvoice

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SIP Voice Mail

SIPRTSP

create URL entry for

email URLor media

voice

recordmessage

prompts

voice message

media playerRTSP server

SIP server

RECORD msgPLAYSETUP

ogm

PLAY

INVITE

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IP Centrex

Internet Heads Up Barber

IP

PSTNISP

Ralph’s Pretty Good Grocery

Chatterbox Cafe

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SIP Standardization Status

� Feb. 2, 1999: IETF Proposed Standard

� March 17, 1999: IETF RFC 2543

� eligible for Draft Standard: 6 months, 2 implementations

p

� new SIP working group (move frommmusic)

� working on updated draft based on implementation experience

� mostly clarifications + optional headers, no new version

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SIP Work Items� sip-cgi

� call processing language(CPL)

� reliable provisional (1xx) re-sponses

� caller preferences

� third-party call control

� SIP for subscribe/notify

� server feature negotiation

� SIP–ISUP interworking

� SIP–H.323 interworking

� billing

� reverse channel setup for callprogress tones

� pre-ringing resource reserva-tion

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SIP Bake-Off

� 3 bake-offs: April, August, December

� from 15 to 33 groups

� hardware, PSTN gateways, proxy/redirect servers, clients, testinstrument, . . .

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SIP Bake-Off Participants

3Com dynamicsoft Mitel

8x8 Ellemtel Netspeak

Agilent Ericsson Nortel

Alcatel Facet Nuera

Broadsoft Helsinki Univ. OZ.com

British Telecom Hewlett-Packard Pingtel

Catapult Indigo Radcom

Cisco IPcell Telogy

Columbia University Lucent Vovida

Dialogic MCI Worldcom VTEL

Mediatrix

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SIP Bake-Off Goals

� basic call set-up

� registration, user location

� proxies and redirect server operation

� advanced features: security

� identify implementation bugs and robustness issues

� identify spec ambiguities

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SIP Bake-Off Results� almost all implementations could establish basic calls – either

on arrival or after minor on-site fixes

� tested redirection, proxying, security, registration, . . .

� generated interoperability test cases and tools

� will fold clarifications into Draft revision of RFC and webpage athttp://www.cs.columbia.edu/˜hgs/sip

� install public testing mechanisms (Pulver OpenTestNet,www.siphappens.com)

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On-Going Columbia IP Telephony Projects� e*phone: Ethernet SIP phone

� sipc SIP user agent for Windows, Linux, Solaris, . . .

� sipd SIP server

� SIP-H.323 translation

� SIP/RTSP unified messaging

� SIP gateways to the telephone network (T1 E&M, ISDN,. . . )

� SIP mobility

� programming servers: CPL, sip-cgi

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On-Going Columbia IP Telephony Projects

� light-weight resource reservation: YESSIR

� resource-reservation aggregation: BGRP

� charging for resource reservations: RNAP

� TRIP telephony gateway routing

� QOS fault detection

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Planned Internet Telephony Projects� detailed security evaluation/implementation

� caller preferences implementation

� service creation environments for end users and admins

� SIP for notification (“buddy lists”)

� interaction of signaling and QOS

� SIP and ISUP

� 911 services

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e*phone

stand-alone Ethernet phone will full SIP implementation

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sipc

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sipd

SIP registrar, proxy and redirect server:

� user registration

� forking proxy capability: one call, many destinations

� security mechanisms (basic, digest, pgp) for authentication

� sip-cgi: interface to scripting languages

� caller preferences

� CPL in progress

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Integrating Signaling and Instant Messaging: Some Ideas� “reverse” signaling: callee indicates availability

� buddy lists = special case ofevent notification

� other events: “sensor 17 smells smoke”, “Beanie Babies are onsale”, “(voice) mail has arrived”, . . .

� subscribe – notify – set up call

� useful for call parking

� many SIP mechanisms apply: security, redirection, proxying,content negotiation, . . .

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SIP for Event Notification

� add two methods:SUBSCRIBE andNOTIFY

� proxy server may interceptSUBSCRIBE

� use message body for event description

� default: presence, indicated byREGISTER

� one ofmanyproposals for presence (IETF WG!)

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SIP for Event Notification

SUBSCRIBE

SUBSCRIBE

NOTIFYNOTIFY

REGISTER

SUBSCRIBECarol

publisher

subscriber

Alice

Bob

proxy

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Mobility� new network ➠ new

IP address (DHCP)

� mobile IP hides addr.changes

� but: little deployment

� �: encapsulationoverhead

� �: dog-legged routing

� �: IP address filtering

CN

CH

HA

FAtunnelleddatadata

data

data

home network

foreignnetwork

mobile hostcorrespondent hostrouter with home agentfunctionalityrouter with foreign agentfunctionality

MH

CH

HA

HA

MH

MH

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SIP Mobility Overview

� pre-call mobility➠ SIP proxy, redirect

� mid-call mobility ➠ SIP re-INVITE, RTP

� recovery from disconnection

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SIP Mobility: Pre-call� MH acquires IP ad-

dress via DHCP

� optional: MH findsSIP server via multi-castREGISTER

� MH updates home SIPserver

� optimization: hierar-chical LR (later)

CH

redir

3

1

2

5

foreignnetwork

homenetwork

4

mobile hostcorrespondent host

SIP redirect server

MH

CH

redir

3

1

2

5

4

SIP INVITE

SIP 302 moved temporarily

SIP INVITE

SIP OK

dataMH

MH

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SIP mobility: mid-call

� MH!CH: new IN-VITE, with Contactand updated SDP

� re-registers with homeregistrar

CH

13

2

foreignnetwork

homemobile hostcorrespondent host

SIP redirect server

MH

CH

redir

3

1

2

SIP INVITE

SIP OK

data

redir

network

MH

MH

MH

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SIP mobility: multi-stage registration

Don’t want to bother home registrar with each move

Contact: alice@CAFrom: alice@NY

Contact: 193.1.1.1

REGISTERINVITE

Los Angeles

San Francisco

Contact: 192.1.2.3From: alice@NY

CA NY

From: alice@NY

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Conclusion� SIP basic standard stable

� multiple interoperating implementations

� backward-compatible features:

– interoperation with legacy signaling systems

– mobility

– caller preferences

– call transfer

– . . .

� programming of services: cgi, CPL, applets

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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