Internet RealTime Laboratory

1
Internet Real-Time Laboratory

• Prof. Henning Schulzrinne
• (presented by Kundan Singh)
• http//www.cs.columbia.edu/IRT

2
Overview

• Columbia Networking Research Center
• CSEE networking groups 15 faculty

• IRT Lab
• 13 PhD
• 6 MS GRA
• 3 visitors
• 1 post doc

3
Research topics
Internet radio/TV
Internet telephony
Peer-to-peer systems
Quality of service
Security
Internet Real Time
Internet service discovery
Content distribution
VoIP and wireless
Resource reservation
Wireless ad hoc networks
4
Programmable Internet telephony
Double ringing sound when boss calls
Enter your authentication PIN for billing
Use finger for locating user
B2BUA
Endpoint
Make call when boss is online
Proxy/registrar
Endpoint
Forward to office phone during day, and home
phone during evening

• Common gateway interface (CGI)
• Call processing language (CPL)
• SIP servlet

• Language for End System Services (LESS)

5
Messaging and device control
Internet
6
VoIP for wireless LAN

• Problem
• L2 Handoff time is too big (500 ms) for seamless
  VoIP sessions (90 ms).
• L2 Handoff procedure
• Scanning APs, Authentication, Association
• The biggest component of handoff time is the
  scanning time ( gt 90)
• Solution
• Selective scanning (100-130 ms)
• Caching (2-4 ms)

7
911 for Internet telephony

• Problems
• Identify
• Route to PSAP
• Location information
• Other issues
• Record media
• Call taker GUI

ALI Server
DHCP Server
verified
TCP Socket
civil
DHCP Inform
Telephone
location
MAC Address
DNS Server
Number
Location
HTTP SOAP
DNS Query
Info
geo location
civil location
PSAP
PSAP
Info
Info
civil location

geo location
sip

sos
_at_
domain
911
w
/
location or
112
w
/
out location
IP Network
POTS
/
Wireless
Network
8
Next step in signaling (NSIS)

• As part of the IETF NSIS working group, we are
  standardizing a protocol for signaling
  information about a data flow along its path in
  the Internet.
• The protocol supports various signaling
  applications, such as Quality of Service (QoS)
  and Network Address Translation (NAT) and
  Firewall traversal.
• The protocol design adopts a two-layer approach,
  a lower layer for transport, and an upper layer
  specific to each signaling application.
• We conducted Internet routing dynamics
  measurement and evaluated route change detection
  methods in typical NSIS deployment models.

9
Email by phone
Email by phone
Inbox
Email
procmail
important mails
Internet
Email to IM

• Login
• Email formatting
• Listen, reply, delete, compose, forward
• Navigation -next, previous, jump

SIP

• Email formatting
• SIP based
• Text-to-speech

SIP
VoiceXML browser
TTS
SIP
HTTP
Internet
Email servlet
JSP
Inbox
DB
Email to phone
10
sipjohn_at_cs.columbia.edu
INVITE sipjohn_at_cs.columbia.edu
My owners SIP address is sipjohn_at_cs.columbia.edu
Help!!! (invoke sipc to call sipjohn_at_cs.columbia.
edu)
11
Other past projects

• Location-based services i-button, badge
• Voice and video mail
• Emergency 911 call routing
• File sharing among conference participants
• Phone announcement server
• Event notification to Phone, IM, email
• UDP-based link simulator
• Wireless ad-hoc networks
• QoS for audio conferencing
• Conference load balancing
• Conference floor control
• Peer-to-peer protocol analysis
• . . .

12
Summary

• Many research projects
• Multimedia, wireless, VoIP, services,
• Prototype, measurement, study,
• PhD, MS GRA and grad/undergrad project students
• More at http//www.cs.columbia.edu/IRT or email
  hgs_at_cs.columbia.edu

13
Reliable and scalable IP telephony

• Reliable and scalable SIP-based call routing and
  user registration services for multimedia
  communication in our CINEMA architecture.
• Server-based improving single server
  performance, and load sharing.
• Peer-to-peer using distributed hash table such
  as Chord.
• Goal Carrier grade performance using commodity
  hardware

14
DotSlash how to deal with 15 min of fame

• Web hotspots
• Existing mechanisms are costly
• DotSlash
• Enables a site utilize spare capacity of other
  site via dynamic collaboration

15
Location-basedServices
16
...from VoIP to multimedia collaboration
CINEMA servers
Telephone switch
rtspd media server
Local/long distance e.g., 1-212-5551212
sipconf conference server
PSTN
RTSP
RTSP clients e.g., Quicktime
Department PBX
sipum unified messaging
Internal Telephone e.g., 7040
sipd proxy, redirect, registrar
713x
SQL database
cgi
Web based configuration
vxml
SIP/PSTN Gateway e.g., Cisco 2600
7134
7136
siph323 SIP-H.323 translator
H.323
alice_at_cs.columbia.edu (software phone)
H.323 clients e.g., NetMeeting
Session Initiation Protocol (SIP)- based
enterprise VoIP infrastructure