IP Telephony Deployment @ Columbia

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IP Telephony Deployment _at_ Columbia
Henning Schulzrinne Wenyu Jiang Sankaran
Narayanan Xiaotao Wu Columbia University Departmen
t of Computer Science hgs,wenyu,sankaran,xiaotaow
_at_cs.columbia.edu
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IP Telephony and Protocols
Corporate/Campus
Another campus
7040
8151
External line
8152
PBX
PBX
7042
8153
SIP server
VoIP Gateway
8154
VoIP Gateway
home.com
Internet
LAN
LAN
SIP server
Call bob_at_office.com
office.com
IP Phone Client/ Audio over RTP
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Architecture
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CINEMA Columbia InterNet Extensible Multimedia
Architecture

• Web interface
• Administration
• User configuration
• Unified Messaging
• Notify by email
• rtsp or http
• Portal Mode
• 3rd party IpTelSP

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PSTN-IP Inter-Operation

• Outgoing calls are similar

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sipc A SIP User Agent
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sipc Architecture
CPL script
CGI script
CPL script
CGI script
Service Creation Environment
Service Logic Execution Environment
Service Creation Environment
Service Logic Execution Environment
Controller
Controller
SIP Stack
SIP Stack
Media application
Media application
Transmission
Transmission
Media transmission
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What about Audio Quality?

• Voice Codec G.711 toll quality at 64 kb/s
• Bandwidth rarely an issue on campus networks
  with Gigabit core switches
• Measurement in the Columbia intranet
  (campus-wide), over a total of 24 hours
• Average (one-way) delay lt 1ms
• Jitter packets gt 10ms 0.003-0.05
• Loss 0.001-0.01, 0.005 average
• Using Ethernet switches instead of hubs prevents
  excessive delay/jitter.

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Scalability, Security and Other

• Scalability based on multiple servers
• SIP Server, via DNS SRV
• Gateway and LAN bandwidth
• Media servers (voice-mail and conferencing)
• Security
• Authenticate users Disallow auth-bypass.
• Gateway calls for only authorized users.
• Further issues of study
• Service availability/reliability, QoS
• Privacy/encryption, Electronic Billing