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

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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
• tel[phone]
• sip provide translation entity
• sip212-939-7040_at_sip-provider.biz
• 701?? tel[phone] (011) tel
• ??????? tel1212
• (8)1?????????? tel1
• (8)(011) tel

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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 tel[phone]
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
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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)