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 what’s your email address?”

7. Migration strategy 1. Add IP phones to existing PBX or Centrex system – PBX as gateway Initial investment:  $2k for gateway 2. Add multimedia capabilities: PCs, dedicated video servers 3. “Reverse” PBX: replace PSTN connection with SIP/IP connection to carrier 4. Retire PSTN phones

8. Implementation difficulties Integration with PBX typically, treat as adjunct T1 PRI (much better!) or CAS T1’s have dozens of configuration combinations AMI or B8ZS, SF or ESF, DID or TIE, voice/data, … two-stage dialing vs. DID caller ID typically doesn’t 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 sip:7134@cs Ethernet phones, soft phones and conference room CINEMA set of servers, running on 1U rackmount server

10. CINEMA components RTSP sipum Cisco 7960 sipvxml SIP rtspdsipconf LDAP server MySQL PhoneJack interface sipc T1 sipd media server RTSP SIP-H.323 converter messaging server unified server (MCU) user database conferencing sip-h323 VoiceXML server proxy/redirect server Cisco 2600 Pingtel wireless 802.11b PBX Meridian Nortel plug'n'sip

11. Experiences Need flexible name mapping Alice.Cueba@cs  alice@cs Alice.Cueba@csalice@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 who’s making the call map to tel URLs or SIP URLs tel: translate at first proxy tel:212-939-7040 sip: provide translation entity sip:212-939-7040@sip-provider.biz 7[01]?? tel:+1212939$ (011)* tel:+$ ??????? tel:+1212$ (8)1?????????? tel:+1$ (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 EPAD INVITE sip:sos@psap.leonia.nj.us Location: 07605 REGISTER sip:sos Location: 07605 302 Moved Contact: sip:sos@psap.leonia.nj.ussos@psap.leonia.nj.us Contact: tel:+1-201-911-1234 SIP proxy INVITE sip:sos Location: 07605 common emergency identifier: sos@domain

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 < 50% Still need common database can separate REGISTER make rest read-only

19. Large system _sip._udp SRV 0 0 sip1.example.com 0 0 sip2.example.com 0 0 sip3.example.com a2.example.com sip2.example.com sip3.example.com a1.example.comsip1.example.com b1.example.com b2.example.com sip:bob@example.com sip:bob@b.example.com _sip._udp SRV 0 0 b1.example.com 0 0 b2.example.com stateless proxies

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)