integrating sip and legacy pbxs henning schulzrinne dept. of computer science columbia university

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

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

Integrating SIP and Integrating SIP and Legacy PBXsLegacy PBXs

Henning SchulzrinneDept. of Computer Science

Columbia University

Page 2: Integrating SIP and Legacy PBXs Henning Schulzrinne Dept. of Computer Science Columbia University

OverviewOverview

Motivation Migration strategy Challenges Example: Columbia Dept. of CS Scaling Emergency calls

Page 3: Integrating SIP and Legacy PBXs Henning Schulzrinne Dept. of Computer Science Columbia University

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

Page 4: Integrating SIP and Legacy PBXs Henning Schulzrinne Dept. of Computer Science Columbia University

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

Page 5: Integrating SIP and Legacy PBXs Henning Schulzrinne Dept. of Computer Science Columbia University

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

Page 6: Integrating SIP and Legacy PBXs Henning Schulzrinne Dept. of Computer Science Columbia University

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?”

Page 7: Integrating SIP and Legacy PBXs Henning Schulzrinne Dept. of Computer Science Columbia University

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

Page 8: Integrating SIP and Legacy PBXs Henning Schulzrinne Dept. of Computer Science Columbia University

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

Page 9: Integrating SIP and Legacy PBXs Henning Schulzrinne Dept. of Computer Science Columbia University

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

Page 10: Integrating SIP and Legacy PBXs Henning Schulzrinne Dept. of Computer Science Columbia University

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

Page 11: Integrating SIP and Legacy PBXs Henning Schulzrinne Dept. of Computer Science Columbia University

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

Page 12: Integrating SIP and Legacy PBXs Henning Schulzrinne Dept. of Computer Science Columbia University

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

Page 13: Integrating SIP and Legacy PBXs Henning Schulzrinne Dept. of Computer Science Columbia University

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

sip:[email protected]

7[01]?? tel:+1212939$ (011)* tel:+$??????? tel:+1212$(8)1?????????? tel:+1$(8)(011)* tel:+$

Page 14: Integrating SIP and Legacy PBXs Henning Schulzrinne Dept. of Computer Science Columbia University

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

Page 15: Integrating SIP and Legacy PBXs Henning Schulzrinne Dept. of Computer Science Columbia University

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

Page 16: Integrating SIP and Legacy PBXs Henning Schulzrinne Dept. of Computer Science Columbia University

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

Page 17: Integrating SIP and Legacy PBXs Henning Schulzrinne Dept. of Computer Science Columbia University

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

Page 18: Integrating SIP and Legacy PBXs Henning Schulzrinne Dept. of Computer Science Columbia University

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

Page 19: Integrating SIP and Legacy PBXs Henning Schulzrinne Dept. of Computer Science Columbia University

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:[email protected]

sip:[email protected]

_sip._udp SRV 0 0 b1.example.com

0 0 b2.example.com

stateless proxies

Page 20: Integrating SIP and Legacy PBXs Henning Schulzrinne Dept. of Computer Science Columbia University

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)