IP telephony overview and IP telephony overview and demonstrationdemonstration

Prof. Henning Schulzrinne(presented by Andrea Forte, Ron Shacham, Sangho Shin, Kundan Singh and Xiaotao Wu)

http://www.cs.columbia.edu/IRT

Research topics in IRT labResearch topics in IRT labInternet telephony Internet radio/TV

Content distribution

Wireless ad hoc networksResource reservation

Quality of serviceSecurity

VoIP and wireless

Internet service discovery

Peer-to-peer systems

What is IP telephony?What is IP telephony? Phone call + Internet User identifier Session Initiation

Protocol – SIP

home.com columbia.edu

office.com

alice@columbia.eduBob

(2)(1) pc4.columbia.edu

(3)

Personal mobilityPersonal mobility

alice_95@yahoo.com

alice@cs.columbia.edu

desk@cs.columbia.edu

Alice.Cueba@cs.columbia.edu

tel:12129397000

tel:17185551234

host.cs.columbia.edu

cs.columbia.edu

yahoo.com

Home

Mobile

Office

Call setupCall setup

Bob

columbia.edu

Office

Mobile

Home

Visiting universityRedirect

Forking

Media pathControl path

ProgrammabilityProgrammability

Common gateway interface (CGI)

Call processing language (CPL)

SIP servlet

Make call when boss is online …

Enter your authentication PIN for billing…

B2BUA

Double ringing sound when boss calls…

Endpoint

Forward to office phone during day, and home phone during evening…

Proxy/registrar

Endpoint

Language for End System Services (LESS)

Use finger for locating user…

ServerProxy, register, redirect.Conferencing.Voicemail, IVR.

Clients and serversClients and serversIP phones

script

SIP server

Urgent

Low-priority

Voicemail

Hardware phones

Software phones

Phone

Interworking with PSTNInterworking with PSTN

Translating: Audio – better codecs on IP Signaling – some features are lost Identifiers – phone numbers

Determining transition points

Telephonenetwork (PSTN)

SIP/PSTN gateway

SIP server IP endpointTelephonesubscriber

x7040 sip:bob@cs

PBX

(212)5551212

Enterprise VoIPEnterprise VoIP

InternalTelephonee.g., 7040

SIP/PSTN Gatewaye.g., Cisco 2600

Department PBX

Web based configuration

Web server

Telephoneswitch

SQLdatabase

sipd:proxy, redirect, registrar

H.323

rtspd: media server

sipum: unified messaging

RTSP clientse.g., Quicktime

RTSP

713x

CINEMA servers

sipconf: conference server

siph323: SIP-H.323 translator

Local/long distance e.g., 1-212-5551212

PSTN

SIP

VXML

vxml

cgi

7134

7136

alice@cs.columbia.edu(software phone) H.323 clients

e.g., NetMeeting

VoIP and wirelessVoIP and wireless Which wireless network?

802.11a/b/g Infrastructure mode (security) Ad-hoc mode

What is handoff? Handoff happens when a

mobile node moves beyond the radio range of one access point and enters another.

Internet

VoIP and wirelessVoIP and wireless What is the problem?

L2 Handoff time is too big (~500 ms) for seamless VoIP sessions (90 ms).

0100200300400500600

1 2 3 4 5 6 7 8 9 10Experiments

Handoff Time (ms) Asso + AuthScanning

VoIP and wirelessVoIP and wireless Improvement in our solution

Handoff Time

0

100

200

300

400

500

600

1 2 3 4 5 6 7 8 9 10Experiments

mse

c

Original HandoffSelective ScanningCaching

Session MobilitySession Mobility Focus on communication media: audio,

video, instant messaging Location sensors and presence, along with

service discovery yields a list of local devices

Seamlessly transfer an active session between devices

Transfer all media to a single device or split over multiple devices Privacy: keep audio on handset, watch video on

large screen Take advantage of benefits of different devices

Session MobilitySession MobilityInternet

CorrespondentNode (CN)

SIP UA

SLP UA

SIP SM

Local Devices

SLP SA SLP UA

SIP SM SIP UA

SLP DA

Mobile Node (MN)

SLPSIPRTP

SIP UA

Transcoder

Serverless (P2P) VoIPServerless (P2P) VoIP Server-based

Cost: maintenance, configuration Central points of failures Controlled infrastructure (e.g., DNS)

Peer-to-peer Robust: no central dependency Self organizing, no configuration Scalability

P2P-SIP Efficient, interoperable, hybrid Prototype implementation

C

C

C

C

C

S

P

P

P

P

P

SummarySummary SIP-based architecture Heterogeneous endpoints

Telephone, SIP phone, H.323 Devices like lamp, video encoder

Multimedia collaboration Conference, IM, discussion board, voicemail, file

sharing Advanced services

Programmable call routing, voice mail, interactive voice response

Fast handoff for WirelessLAN P2P-SIP for serverless VoIP

• Web configuration• Audio mixing• Video replication• SIP, PSTN or H.323sipcsipc

SIP/PSTNSIP/PSTN

ConferencingConferencing

Voicemail and IVRVoicemail and IVR Multi-platform (phone, PC) access Standard based (SIP, RTSP) Programmable dialogues

Room confLocation agent

Bob

RFID readeriButtonreader

ProxyLS

Bob is in conf

NOTIFYLocation

You areIn conf

SLP DA

SLP SA

Device GW SLinke

X10

Turn onconf’s light

Turn on light

What’savailable

sip:conf_pingtelfor audio

sip:conf

Tracking

Triggeran action

Resourcediscovery

Location-basedLocation-basedServices in our Services in our lablab

Location-basedLocation-basedServices in our Services in our lablab

Location agent

Bob

RFID readeriButtonreader

ProxyLS

Bob is in conf

NOTIFYLocation

You areIn conf

SLP DA

SLP SA

Device GW SLinke

X10

Turn onconf’s light

Turn on light

What’savailable

sip:conf_pingtelfor audio

sip:conf

INVITE sip:anyone_roomconf

Guard communicatio

n behavior

‘Talk’ to alocation

Room conf