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Towards Junking the PBX: Deploying IP Telephony

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Presentation on theme: "Towards Junking the PBX: Deploying IP Telephony"— Presentation transcript:

1 Towards Junking the PBX: Deploying IP Telephony
Wenyu Jiang, Jonathan Lennox, Henning Schulzrinne and Kundan Singh Columbia University We describe our departmental IP telephony installation

2 Columbia University, Deploying IP Telephony
What is a PBX ? 7040 External line 7041 Corporate/Campus Telephone switch Private Branch Exchange Another switch 7042 7043 Corporate/Campus LAN Internet March 12, 2001 Columbia University, Deploying IP Telephony

3 Columbia University, Deploying IP Telephony
What is IP Telephony ? Corporate/Campus Another campus 7040 8151 External line 8152 7041 PBX PBX 7042 8153 8154 7043 Internet LAN LAN March 12, 2001 Columbia University, Deploying IP Telephony

4 IP Telephony Protocols
Call 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 Send and receive audio packets March 12, 2001 Columbia University, Deploying IP Telephony

5 Columbia University, Deploying IP Telephony
Architecture rtspd SIP/RTSP Unified messaging RTSP media server sipum Quicktime RTSP clients RTSP T1/E1 RTP/SIP Telephone Cisco 2600 gateway switch SIP conference server sipconf Web based configuration Web server SIP proxy, redirect server SQL database sipd e*phone sipc Software SIP user agents Hardware Internet (SIP) phones SIPH.323 convertor NetMeeting sip323 H.323 March 12, 2001 Columbia University, Deploying IP Telephony

6 Columbia University, Deploying IP Telephony
Example Call Bob signs up for the service from the web as sipd canonicalizes the destination to He registers from multiple phones sipd rings both e*phone and sipc Alice tries to reach Bob INVITE Bob accepts the call from sipc and starts talking Web based configuration Web server Call Bob SIP proxy, redirect server SQL database sipd e*phone sipc Software SIP user agents Hardware Internet (SIP) phones cs.columbia.edu March 12, 2001 Columbia University, Deploying IP Telephony

7 Columbia University, Deploying IP Telephony
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 March 12, 2001 Columbia University, Deploying IP Telephony

8 Columbia University, Deploying IP Telephony
PSTN to IP Call PBX PSTN External T1/CAS Regular phone (internal) Call 1 Gateway Internal T1/CAS (Ext: ) Call 7134 2 713x is called a part of Coordinated Dial Plan (CDP) in a Nortel PBX SIP server sipd Ethernet 3 sipc 5 Bob’s phone SQL database 4 7134 => bob DID - direct and simple No-DID - dial extension, supports more users March 12, 2001 Columbia University, Deploying IP Telephony

9 Columbia University, Deploying IP Telephony
IP to PSTN Call PBX Internal T1/CAS Call 4 Regular phone (internal, 7054) PSTN External T1/CAS Call 5 Gateway ( ) 3 Ethernet SIP server sipd sipc 1 Bob calls SQL database 2 Use March 12, 2001 Columbia University, Deploying IP Telephony

10 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) March 12, 2001 Columbia University, Deploying IP Telephony

11 Columbia University, Deploying IP Telephony
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 March 12, 2001 Columbia University, Deploying IP Telephony

12 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 March 12, 2001 Columbia University, Deploying IP Telephony


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