Transcript IRT-overview-2006 - Computer Science, Columbia University

The Internet Real-Time
Laboratory (IRT)
http://www.cs.columbia.edu/IRT
Prof. Henning Schulzrinne
Dept. of Computer Science
Columbia University
New York, NY
January 2006
Networking research at
Columbia University
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Columbia Networking Research Center
both Electrical Engineering & Computer
Science Department
13 faculty – one of the largest
networking research groups in the US
about 40 PhD students
spanning optical networks and wireless
channels to operating systems, security
and applications
theory (performance analysis) to
systems (software, protocols)
Steve Bellovin
Keren Bergman
Ed Coffman
Predrag Jelenkovic
Angelos Keromytis
Aurel Lazar
Nick Maxemchuk
Vishal Misra
Jason Nieh
Dan Rubenstein
Henning Schulzrinne
Xiaodong Wang
Yechiam Yemini
Laboratory overview
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Dept. of Computer Science: 35 faculty
IRT lab staff:
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1 post-doc, 3 researchers
10 PhD students
7 MS GRAs
visitors (Ericsson, Fujitsu, Mitsubishi, Nokia,
U. Coimbra, U. Rome, NTT, …)
China, Finland, Greece, India, Japan,
Portugal, Spain, Sweden, US, Taiwan
~10 MS and undergraduate project students
Laboratory support
Equipment grants and student support
Overall IRT lab goals
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Reliable, flexible and programmable
communication infrastructure for Internetbased collaboration applications
Systematic evaluation by analysis and
simulation
Demonstrate capability via prototypes
Contribute protocols to standardization (IETF)
Convert prototypes into products and opensource software
Train students at all levels in current Internet
research and engineering
IRT research topics
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Internet telephony and
multimedia
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CINEMA – VoIP/multimedia and
collaboration system
QoS measurements
network application reliability
performance and server
architecture
APIs for SIP IM and presence
systems
ubiquitous computing using SIP
application sharing
P2P SIP systems
emergency services (“911”)
SIP security
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reputation systems, spam
firewalls
service creation languages
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CPL
LESS
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Mobile and wireless systems
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802.11 handoff acceleration
802.11 VoIP performance
improvements
SIP-based terminal mobility
personal, service and session
mobility
Peer-to-peer messaging  7DS
Service and event discovery
(GloServ)
Generic signaling protocols
(GIMPS) for QoS, NAT/FW, …
Autonomic computing
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service discovery  mSLP
automated server pooling 
DotSlash
IRT and standards
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PI and researchers active in IETF since 1992:
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RTP – lead author
RTSP – lead author
SIP – original design & core team
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SIP mobility
rich presence
privacy and geo-services
emergency calling
SLP – extensions for scalability
GIST – network signaling protocol
Graduated PhD students
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Internet telephony services, GSM interoperation (J.
Lennox)
QoS and reliability measurements (W. Jiang)
Federated CDNs (L. Amini)
Pricing for QoS, LDAP performance (X. Wang)
Multicast QoS fairness & signaling (P. Mendes)
Internet telephony topics (J. Rosenberg)
Mobile peer-to-peer systems (M. Papadopouli)
Scalable resource reservation (P. Pan)
VoIP service creation (J. Lennox)
Multimedia systems problems
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Old problems and
approaches:
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efficient codecs
ubiquitous reachability
audio/video
synchronization
network-layer mobility
quality-of-service
APIs and middleware
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New problems:
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controlled reachability
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spam
cell phone ringing in
lecture
service availability
information privacy
service & personal
mobility
service creation by nonexperts
CINEMA components
Cisco 7960
MySQL
sipconf
user database
rtspd
LDAP server
conferencing
server
(MCU)
sipd
RTSP
media
server
RTSP
proxy/redirect server
unified
messaging
server
Pingtel
Nortel
Meridian
Cisco
2600
sipum
VoiceXML
server
PBX
T1
T1
SIP
sipvxml
PhoneJack interface
sipc
SIP-H.323
converter
sip-h323
plug'n'sip
wireless
802.11b
SIP emergency calling
GPS
INVITE sips:sos@
48° 49' N 2° 29' E
outbound
proxy server
DHCP
48° 49' N 2° 29' E  Paris fire department
SIP for ubiquitous computing
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Focus on inter-domain, scalable systems
Components:
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context-aware communications
context-aware service and event discovery
location-based services
global-scale event notification
service creation by end users
terminal, personal, session and service mobility
Context-aware communication
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context = “the interrelated conditions in
which something exists or occurs”
anything known about the participants in
the (potential) communication relationship
both at caller and callee:
time
CPL
capabilities
caller preferences
location
location-based call routing
location events
activity/availability
“rich” presence
sensor data (mood, bio) not yet, but similar in many
aspects to location data
RPIDS: rich presence data
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Basic IETF presence (CPIM) only gives you
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contact information (SIP, tel URI)
priority
“open” or “closed”
Extend to much richer context information
everything
watcher
PUA
PA
watcher
PUBLISH
NOTIFY
"vague"
watcher
INVITE
CPL
Session mobility
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Walk into office, switch
from cell phone to desk
phone
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call transfer problem 
SIP REFER
related problem: split
session across end
devices
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e.g., wall display + desk
phone + PC for
collaborative application
assume devices (or
stand-ins) are SIPenabled
third-party call control
Service mobility: user-adaptive
device configuration
802.11 signal
strength  location
SLP
“all devices that are in the building”
RFC 3082?
device
controller
REGISTER
To: 815cepsr
Contact: alice@cs
PA
HTTP
SUBSCRIBE
to each room
1. discover room URI
2. REGISTER as contact for room URI
tftp
SIP
SUBSCRIBE to configuration
for users currently in rooms
room 815
Location-based services
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Presence-based approach:
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UA publishes location to presence agent (PA)
becomes part of general user context
other users (human and machines) subscribe to
context
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call handling and direction
location-based anycast (“anybody in the room”)
location-based service directory
Languages for location-based services
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building on experience with our XML-based service
creation languages
CPL for user-location services
LESS for end system services
Location-based IM & presence
Service creation
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Promise of faster service creation
traditionally, only vendors (and sometimes carriers)
learn from web models
end user
network
servers
programmer,
carrier
SIP servlets,
sip-cgi
end system
VoiceXML
VoiceXML (voice),
LESS
CPL
Service creation environment for
CPL and LESS
GloServ: Hierarchical P2P Global Service Discovery Architecture
Knarig Arabshian and Henning Schulzrinne
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Classify services using OWL
Use service classification to map ontology to a hierarchical P2P
network (using CAN for p2p)
Bootstrap servers using information in ontology
Intelligent registration and querying
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CAN DHT
distribution of properties
1) Query for “inn”
is issued
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(Wyoming) 50
4
Hotel
3
hostel
Service
hasState
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2) Map the word
“inn” to “hotel”
Restaurant Travel Medical Communication
(Arizona) 3
(Alaska)
inn rooming lodging motel
2
<1,2> <3,2>
<1,3> <3,3>
<2,1>
<2,2> . . .
<2,3> <10,2>
<10,3>
(Alabama) 1
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(Sports) (Adventure) (Sightseeing)
hasActivity
4) Send the query to the
closest high-level server
that is known
Destination Flights Agencies
Hotel
domain: hotel.destination.service
Bed&Breakfast
3) Look up the domain of
the equivalent server or
closely related server in
the primitive skeleton
ontology
QoS in VoIP Wireless Networks:
Adaptive Priority Control (APC)
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Decides the priority of the AP adaptively
based on
Wireless channel condition
 Uplink and downlink traffic volume
Controls the transmission rate of the AP
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according to the priority of the AP using
Content Free Transmission.
No changes in wireless nodes
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Number of VoIP Flow
Capacity
Adaptive Priority Control
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Uplink
600
End-to-end Delay (ms)
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Unbalanced uplink and downlink
delay due to fairness in DCF.
Uplink and downlink delay need to
be balanced for better QoS and
capacity for VoIP.
AP needs to have a higher priority
than the wireless nodes for fairness
between uplink and downlink.
Uplink
Downlink
End-to-end delay of VoIP in APC
600
End-to-end delay (ms)
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Downlink
End-to-end delay of VoIP in DCF
500
400
300
200
100
0
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Number of VoIP flow Capacity
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Accelerating DHCP: P-DAD
DHCP server
Address Usage Collector (AUC)
TCP Connection
IP4
DUID4
IP
MAC Expire Client ID MAC
IP1
MAC1
570
DUID1
MAC1
IP2
IP3
MAC2
MAC3
580
590
DUID2
DUID3
MAC2
MAC3
Broadcast-ARP/DHCP
Router/Relay Agent
SUBNET
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AUC builds DUID:MAC pair table (DHCP traffic only)
AUC builds IP:MAC pair table (broadcast and ARP traffic)
New pair is added to table, unauthorized IP detected  AUC sends
pair to DHCP server
DHCP server checks if pair is correct and records IP address as in use
ARP checking
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AUC scans unused IPs using ARP query periodically
Silent nodes can be detected
Ad-hoc wireless infrastructure
./: Rescue service for web servers
experiencing 15 minutes of fame
Extend Apache: mod_dots, dotsd, DNS, mSLP
State Transition
Conclusion
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Other topics:
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Focus on Internet multimedia services
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Skype analysis, QoS signaling, QoS for voice-over802.11
fixed & mobile applications
VoIP – protocols, presence, location-based
services, service creation, p2p networks
QoS in networks
Protocols, prototypes, performance evaluation