Transcript No Slide Title

Towards Junking the PBX:
Deploying IP Telephony
Wenyu Jiang, Jonathan Lennox, Henning Schulzrinne and Kundan Singh
Columbia University
{wenyu,lennox,hgs,kns10}@cs.columbia.edu
We describe our departmental IP telephony installation
What is a PBX ?
7040
212-8538080
External line
7041
Corporate/Campus
Private Branch
Exchange
Telephone
switch
Another
switch
7042
7043
Corporate/Campus LAN
March 12, 2001
Internet
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Telephony
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What is IP Telephony ?
Another campus
Corporate/Campus
7040
8151
External line
8152
7041
PBX
PBX
8153
7042
8154
7043
LAN
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Internet
Columbia University, Deploying IP
Telephony
LAN
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IP Telephony Protocols
Call “[email protected]”
SIP
server
home.com
office.com
• Contact “office.com” asking for “bob”
Session Initiation Protocol - SIP
• Locate Bob’s current phone and ring
• Bob picks up the ringing phone
Real time Transport Protocol - RTP
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• Send and receive audio packets
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Telephony
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Architecture
rtspd
RTSP media
server
sipconf
SIP
conference
server
Telephone
Telephone
switch
T1/E1
RTP/SIP
sipd
SIP proxy,
redirect
server
Quicktime
RTSP
RTSP clients
sipum
SIP/RTSP
Unified
messaging
Web based
configuration
Web server
SQL
database
e*phone
Cisco 2600 gateway
Hardware
Internet (SIP)
phones
sipc
NetMeeting
sip323
Software SIP
user agents
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SIPH.323
convertor
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Telephony
H.323
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Example Call
• Bob signs up for the service from the
web as “[email protected]”
• sipd canonicalizes the destination
to sip:[email protected]
• sipd rings both e*phone and sipc
• He registers from multiple phones
• Alice tries to reach Bob
INVITE sip:[email protected]
• Bob accepts the call from sipc and
starts talking
Web based
configuration
sipd
SIP proxy,
redirect
server
Call Bob
Web server
SQL
database
e*phone
Hardware
Internet (SIP)
phones
sipc
cs.columbia.edu
Software SIP
user agents
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Telephony
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Other Services
• Programmable servers
– Time-of-day, caller identification
– CPL, SIP CGI
• Unified messaging
– Centralized voice mail and answering machine
– SIP, RTSP
• Conferencing
– Dial-in bridges; centralized audio mixing
– Audio, video and chat
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Telephony
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PSTN to IP Call
PBX
PSTN
External T1/CAS
1 Call 9397134
713x is called a part of
Coordinated Dial Plan
(CDP) in a Nortel PBX
Gateway
Internal T1/CAS
(Ext:7130-7139)
2
Call 7134
Ethernet
5
Regular phone
(internal)
3
SIP server
• DID - direct and simple
• No-DID - dial extension,
supports more users
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sipc
Bob’s phone
Columbia University, Deploying IP
Telephony
SQL
database
sipd
4
7134 => bob
8
IP to PSTN Call
PBX
PSTN
External T1/CAS
5 Call 5551212
Gateway
(10.0.2.3)
Internal T1/CAS
4 Call 85551212
3
Ethernet
5551212
Regular phone
(internal, 7054)
1
Bob calls
5551212
SIP server
sipc
2
SQL
database
sipd
Use sip:[email protected]
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Telephony
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T1 Line Configuration
(From the PBX Side)
• Electrical/physical settings
– T1 type: Channelized, PRI
– Characteristics: line coding - AMI, B8ZS; framing
- D4, ESF
• Trunk type: DID, TIE
• Channel type: Data, Voice-only, Data/Voice
• Access permissions: adjust NCOS for internal
T1 trunk and CDP routing entry (713x)
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Telephony
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Security
• Prevent unauthorized users from making certain
(e.g., long-distance) calls
• IOS access control
• SIP authentication
Future:
• PIN numbers for telephone users
• Automated, electronic billing
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Telephony
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Conclusion and Future Work
• Initial field test experience with deploying IP
telephony in a campus environment
• The architecture and installation experience can be
used at other organizations
Future Work:
• Additional services, e.g., instant messaging,
VoiceXML
• Performance and scalability: sipd, rtspd, sipconf
• Firewall/NAT, SNMP
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Telephony
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