towards junking the pbx: deploying ip telephony
DESCRIPTION
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. - PowerPoint PPT PresentationTRANSCRIPT
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 “[email protected]”
• 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
March 12, 2001 Columbia University, Deploying IP Telephony
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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 “[email protected]”• He registers from multiple phones
• Alice tries to reach Bob INVITE sip:[email protected]
• sipd canonicalizes the destination to sip:[email protected]• sipd rings both e*phone and sipc
• Bob accepts the call from sipc and starts talking
March 12, 2001 Columbia University, Deploying IP Telephony
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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
March 12, 2001 Columbia University, Deploying IP Telephony
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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
March 12, 2001 Columbia University, Deploying IP Telephony
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IP to PSTN CallIP to PSTN Call
Gateway(10.0.2.3)
3
SQLdatabase
2Use sip:[email protected]
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
March 12, 2001 Columbia University, Deploying IP Telephony
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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