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Integrating SIP and Legacy PBXs

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Title: PowerPoint Presentation Last modified by: Henning Schulzrinne Created Date: 1/1/1601 12:00:00 AM Document presentation format: On-screen Show – PowerPoint PPT presentation

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Title: Integrating SIP and Legacy PBXs


1
Integrating SIP and Legacy PBXs
  • Henning Schulzrinne
  • Dept. of Computer Science
  • Columbia University

2
Overview
  • Motivation
  • Migration strategy
  • Challenges
  • Example Columbia Dept. of CS
  • Scaling
  • Emergency calls

3
Motivation
  • Allow migration of enterprises to IP multimedia
    communication
  • Add capacity to existing PBX, without upgrade
  • Allow both
  • IP centrex hosted by carrier
  • PBX-style locally hosted
  • Unlike classical centrex, transition can be done
    transparently

4
Motivation
  • Not cheaper phone calls
  • Single number, follow-me even for analog phone
    users
  • Integration of presence
  • person already busy better than callback
  • physical environment (IR sensors)
  • Integration of IM
  • no need to look up IM address
  • missed calls become IMs
  • move immediately to voice if IM too tedious

5
Motivation
  • Cheaper wiring
  • with Ethernet power, no need for power brick
  • Flexible allotment of ports, without fixed
    RJ-11/RJ-45 boundary
  • No growth steps

6
Motivation
  • CTI never really worked
  • Used only for call centers, now for everyone
  • Integrate phone and PC
  • PC shows web page and photo of caller
  • PC shows call history
  • No more And whats your email address?

7
Migration strategy
  • Add IP phones to existing PBX or Centrex system
    PBX as gateway
  • Initial investment ?2k for gateway
  • Add multimedia capabilities PCs, dedicated video
    servers
  • Reverse PBX replace PSTN connection with
    SIP/IP connection to carrier
  • Retire PSTN phones

8
Implementation difficulties
  • Integration with PBX
  • typically, treat as adjunct
  • T1 PRI (much better!) or CAS
  • T1s have dozens of configuration combinations
  • AMI or B8ZS, SF or ESF, DID or TIE, voice/data,
  • two-stage dialing vs. DID
  • caller ID typically doesnt work
  • peculiar notions of privileges (caller callee)
  • arcane commands, undocumented
  • Voicemail integration
  • message deposit and retrieval
  • message-waiting light

9
Example Columbia Dept. of CS
  • About 100 analog phones on small PBX
  • DID
  • no voicemail
  • T1 to local carrier
  • Added small gateway and T1 trunk
  • Call to 7134 becomes sip7134_at_cs
  • Ethernet phones, soft phones and conference room
  • CINEMA set of servers, running on 1U rackmount
    server

10
CINEMA components
Cisco 7960
MySQL
rtspd
sipconf
user database
LDAP server
plug'n'sip
RTSP
conferencing
media
server
server
(MCU)
wireless
sipd
802.11b
RTSP
proxy/redirect server
unified
messaging
server
Pingtel
sipum
Cisco
Nortel
2600
Meridian
VoiceXML
PBX
server
T1
T1
SIP
sipvxml
PhoneJack interface
sipc
SIP-H.323
converter
sip-h323
11
Experiences
  • Need flexible name mapping
  • Alice.Cueba_at_cs ? alice_at_cs
  • sources database, LDAP, sendmail aliases,
  • Automatic import of user accounts
  • In university, thousands each September
  • /etc/passwd
  • LDAP, ActiveDirectory,
  • much easier than most closed PBXs
  • Integrate with Ethernet phone configuration
  • often, bunch of tftp files
  • Integrate with RADIUS accounting

12
Experiences
  • Password integration difficult
  • Digest needs plain-text, not hashed
  • Different user classes students, faculty, admin,
    guests,
  • Who pays if call is forwarded/proxied?
  • authentication and billing behavior of PBX and
    SIP system may differ
  • but much better real-time rating

13
Dialplans
  • Can be implemented in phone or proxy
  • timeout or explicit termination
  • canonicalize first, then find route
  • some may go PSTN, some IP
  • may depend on whos making the call
  • map to tel URLs or SIP URLs
  • tel translate at first proxy
  • tel212-939-7040
  • sip provide translation entity
  • sip212-939-7040_at_sip-provider.biz
  • 701?? tel1212939 (011) tel
  • ??????? tel1212
  • (8)1?????????? tel1
  • (8)(011) tel

14
Likely problems elsewhere
  • NATs
  • prevent inbound calls
  • make outbound UDP iffy
  • Low access bandwidth
  • need voice (UDP) prioritization
  • most IP phones support DSCP
  • possibly smaller MTU needed

15
Small gateways are dumb
  • No notion of users, passwords or authentication,
    accounting,
  • Thus, proxy needs to provide this
  • But avoid bypass users could talk to gateway
    directly and bypass pesky billing and
    authentication
  • Use built-in firewall and IP restrictions

16
Emergency calls
common emergency identifier sos_at_domain
EPAD
REGISTER sipsos Location 07605
302 Moved Contact sipsos_at_psap.leonia.nj.us Conta
ct tel1-201-911-1234
SIP proxy
INVITE sipsos Location 07605
INVITE sipsos_at_psap.leonia.nj.us Location 07605
17
Scaling and redundancy
  • Single host can handle ?10-100 calls
    registrations/second ? 18,000-180,000 users
  • 1 call, 1 registration/hour
  • Conference server about 50 small conferences or
    large conference with 100 users
  • For larger system and redundancy, replicate proxy
    server

18
Scaling and redundancy
  • DNS SRV records allow static load balancing and
    fail-over
  • but failed systems increase call setup delay
  • can also use IP address stealing to mask failed
    systems, as long as load lt 50
  • Still need common database
  • can separate REGISTER
  • make rest read-only

19
Large system
stateless proxies
a1.example.com
sip1.example.com
a2.example.com
sip2.example.com
sipbob_at_example.com
b1.example.com
sipbob_at_b.example.com
sip3.example.com
b2.example.com
_sip._udp SRV 0 0 b1.example.com 0
0 b2.example.com
_sip._udp SRV 0 0 sip1.example.com
0 0 sip2.example.com 0 0
sip3.example.com
20
Conclusions
  • VoIP with SIP attractive for upgrading PBXs
  • Add-on functions benefit even analog users
  • No feature difference between large and small
    installations
  • Adding gateway to PBX painful
  • PBX IP interfaces likely easier
  • Complete integration is difficult (voicemail)
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