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

Columbia University, Deploying IP Telephony What is a PBX ? 7040 212-8538080 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

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

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 March 12, 2001 Columbia University, Deploying IP Telephony

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

Columbia University, Deploying IP Telephony 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 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

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

Columbia University, Deploying IP Telephony PSTN to IP Call PBX PSTN External T1/CAS Regular phone (internal) Call 9397134 1 Gateway Internal T1/CAS (Ext:7130-7139) 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

Columbia University, Deploying IP Telephony IP to PSTN Call PBX Internal T1/CAS Call 85551212 4 Regular phone (internal, 7054) PSTN External T1/CAS Call 5551212 5 5551212 Gateway (10.0.2.3) 3 Ethernet SIP server sipd sipc 1 Bob calls 5551212 SQL database 2 Use sip:85551212@10.0.2.3 March 12, 2001 Columbia University, Deploying IP Telephony

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

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

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