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Towards Junking the PBX: Deploying IP Telephony

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.

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Towards Junking the PBX: Deploying IP Telephony

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  1. 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

  2. What is a PBX ? 7040 212-8538080 External line 7041 Telephone switch Corporate/Campus Private Branch Exchange Another switch 7042 7043 Corporate/Campus LAN Internet Columbia University, Deploying IP Telephony

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

  4. IP Telephony Protocols Call “bob@office.com” 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 Columbia University, Deploying IP Telephony

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

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

  7. 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 Columbia University, Deploying IP Telephony

  8. PBX Gateway PSTN Internal T1/CAS (Ext:7130-7139) External T1/CAS Call 9397134 Call 7134 Ethernet 713x is called a part of Coordinated Dial Plan (CDP) in a Nortel PBX 2 1 3 4 5 Regular phone (internal) SIP server SQL database sipd sipc Bob’s phone 7134 => bob PSTN to IP Call • DID - direct and simple • No-DID - dial extension, supports more users Columbia University, Deploying IP Telephony

  9. PBX Gateway (10.0.2.3) PSTN External T1/CAS Internal T1/CAS Call 5551212 Call 85551212 Ethernet 4 2 5 1 3 5551212 Bob calls 5551212 Regular phone (internal, 7054) SIP server SQL database sipd sipc Use sip:85551212@10.0.2.3 IP to PSTN Call 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) Columbia University, Deploying IP Telephony

  11. 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 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 Columbia University, Deploying IP Telephony

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