integrating sip and legacy pbxs henning schulzrinne dept. of computer science columbia university
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Integrating SIP and Integrating SIP and Legacy PBXsLegacy PBXs
Henning SchulzrinneDept. of Computer Science
Columbia University
OverviewOverview
Motivation Migration strategy Challenges Example: Columbia Dept. of CS Scaling Emergency calls
MotivationMotivation 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
MotivationMotivation 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
MotivationMotivation
Cheaper wiring with Ethernet power, no need for
power brick Flexible allotment of ports, without
fixed RJ-11/RJ-45 boundary No growth steps
MotivationMotivation 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?”
Migration strategyMigration 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
Implementation difficultiesImplementation 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
Example: Columbia Dept. Example: Columbia Dept. of CSof 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
CINEMA componentsCINEMA components
RTSP
sipum
Cisco 7960
sipvxmlSIP
rtspdsipconfLDAP server
MySQL
PhoneJack interface
sipc
T1T1
sipd
mediaserver
RTSP
SIP-H.323converter
messagingserver
unified
server(MCU)
user database
conferencing
sip-h323
VoiceXMLserver
proxy/redirect server
Cisco2600
Pingtel
wireless802.11b
PBX
MeridianNortel
plug'n'sip
ExperiencesExperiences Need flexible name mapping
Alice.Cueba@cs alice@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
ExperiencesExperiences
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
DialplansDialplans 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
7[01]?? tel:+1212939$ (011)* tel:+$??????? tel:+1212$(8)1?????????? tel:+1$(8)(011)* tel:+$
Likely problems elsewhereLikely 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
Small gateways are dumbSmall 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
Emergency callsEmergency calls
EPAD
INVITE sip:[email protected]
Location: 07605
REGISTER sip:sos
Location: 07605
302 MovedContact: sip:[email protected]: tel:+1-201-911-1234
SIP proxyINVITE sip:sos
Location: 07605
common emergency identifier: sos@domain
Scaling and redundancyScaling 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
Scaling and redundancyScaling 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
Large systemLarge system
_sip._udp SRV 0 0 sip1.example.com
0 0 sip2.example.com
0 0 sip3.example.com
a2.example.comsip2.example.co
m
sip3.example.com
a1.example.com
sip1.example.com
b1.example.com
b2.example.com
_sip._udp SRV 0 0 b1.example.com
0 0 b2.example.com
stateless proxies
ConclusionsConclusions 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)