Towards Junking the PBX: Towards Junking the PBX: Deploying IP TelephonyDeploying IP Telephony

Wenyu Jiang, Jonathan Lennox, Henning Schulzrinne and Kundan SinghColumbia University

{wenyu,lennox,hgs,kns10}@cs.columbia.edu

We describe our departmental IP telephony installation

March 12, 2001 Columbia University, Deploying IP Telephony

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What is a PBX ?What is a PBX ?

7043

7040

7041

7042

External line

Telephoneswitch

Private BranchExchange

212-8538080

Anotherswitch

Corporate/Campus

InternetCorporate/Campus LAN

March 12, 2001 Columbia University, Deploying IP Telephony

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What is IP Telephony ?What is IP Telephony ?

External line

7043

7040

7041

7042

PBX

Corporate/Campus

InternetLAN

8154

8151

8152

8153

PBX

Another campus

LAN

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IP Telephony ProtocolsIP Telephony Protocols

Call “bob@office.com”

• Contact “office.com” asking for “bob”• Locate Bob’s current phone and ring

office.comhome.com

• Bob picks up the ringing phone

• Send and receive audio packets

Session Initiation Protocol - SIP

Real time Transport Protocol - RTP

SIP server

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ArchitectureArchitecture

SIP proxy,redirectserver

SQLdatabase

sipd

e*phone

sipc

Software SIP user agents

Hardware Internet (SIP)

phones

SIPH.323convertor

NetMeetingsip323

H.323

rtspd

SIP/RTSPUnified

messaging

RTSP media server

sipum

Quicktime

RTSP clients

RTSP

SIP conference

server

sipconf

T1/E1 RTP/SIP

Telephone

Cisco 2600 gateway

Telephoneswitch Web based

configuration

Web server

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SIP proxy,redirectserver

SQLdatabase

sipd

e*phone

sipc

Software SIP user agents

Hardware Internet (SIP)

phones

Web based configuration

Web server

cs.columbia.edu

Call Bob

Example CallExample Call• Bob signs up for the service from the web as “bob@cs.columbia.edu”• He registers from multiple phones

• Alice tries to reach Bob INVITE sip:Bob.Wilson@cs.columbia.edu

• sipd canonicalizes the destination to sip:bob@cs.columbia.edu• sipd rings both e*phone and sipc

• Bob accepts the call from sipc and starts talking

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Other ServicesOther 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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PSTN to IP CallPSTN to IP CallPBXPSTN

External T1/CAS

Regular phone(internal)

Call 93971341

SIP server

sipd

Ethernet

3

SQLdatabase

4 7134 => bob

sipc

5

Bob’s phone

• DID - direct and simple• No-DID - dial extension, supports more users

GatewayInternal T1/CAS(Ext:7130-7139)

Call 71342

713x is called a part of Coordinated Dial Plan (CDP) in a Nortel PBX

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IP to PSTN CallIP to PSTN Call

Gateway(10.0.2.3)

3

SQLdatabase

2Use sip:85551212@10.0.2.3

Ethernet

SIP server

sipdsipc

1Bob calls 5551212

PSTN

External T1/CASCall 55512125

5551212

PBX

Internal T1/CASCall 85551212 4

Regular phone(internal, 7054)

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T1 Line Configuration T1 Line Configuration (From the PBX Side)(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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SecuritySecurity

• 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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Conclusion and Future WorkConclusion 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