towards junking the pbx: deploying ip telephony wenyu jiang, jonathan lennox, henning schulzrinne...

Towards Junking the PBX: Towards Junking the PBX: Deploying IP TelephonyDeploying IP Telephony

Wenyu Jiang, Jonathan Lennox, Henning Schulzrinne and Kundan SinghColumbia University

{wenyu,lennox,hgs,kns10}@cs.columbia.edu

We describe our departmental IP telephony installation

6/26/2001 Columbia University, Deploying IP Telephony

2

OutlineOutline

• Introduction to IP telephony

• System architecture

• Call flows

• System configuration

• Security

• Scalability


3

Traditional Telecommunication Traditional Telecommunication InfrastructureInfrastructure

7043

7040

7041

7042

External line

Telephoneswitch

Private BranchExchange

212-8538080

Anotherswitch

Corporate/Campus

InternetCorporate/Campus LAN


4

What is IP Telephony ?What is IP Telephony ?

External line

7043

7040

7041

7042

PBX

Corporate/Campus

InternetLAN

8154

8151

8152

8153

PBX

Another campus

LAN

IP Phone Client

VoIP Gateway VoIP Gateway


5

IP Telephony ProtocolsIP Telephony Protocols

Call “[email protected]”

• Contact “office.com” asking for “bob”

• Locate Bob’s current phone and ring

office.comhome.com

• Bob picks up the ringing phone

• Send and receive audio packets

Session Initiation Protocol- SIP

Real time Transport Protocol- RTP

SIP server

audio over RTP


6

ArchitectureArchitecture

SIP proxy,redirectserver

SQLdatabase

sipd

SIPH.323convertor

NetMeetingsip323

H.323

rtspd

SIP/RTSPUnified

messaging

RTSP media server

sipum

Quicktime

RTSP clients

RTSP

SIP conference

server

sipconf

T1/E1 RTP/SIP

Telephone

Cisco 2600 gateway

Telephoneswitch

Web based configuration

Web server

Cisco 7960

3Com

e*phone

sipc

Software SIP user agents

Hardware Internet (SIP)

phones

Pingtel

SNMP


7

SIP proxy,redirectserver

SQLdatabase

sipd

e*phone

sipc

Software SIP user agents

Hardware Internet (SIP)

phones

Web based configuration

Web server

cs.columbia.edu

Call Bob

Example Call (IP only)Example Call (IP only)• Bob signs up for the service from the web as “[email protected]”

• He registers from multiple phones

• Alice tries to reach Bob INVITE sip:[email protected]

• sipd canonicalizes the destination to sip:[email protected]

• sipd rings both e*phone and sipc

• Bob accepts the call from sipc and starts talking


8

CanonicalizationCanonicalization

Bob.Wilson

canonicalize

bob@cs


9

Other ServicesOther Services

• Programmable servers– Time-of-day, caller identification– CPL, SIP CGI

• Unified messaging– Centralized voice mail– SIP, RTSP

• Conferencing– Dial-in bridges; centralized audio mixing– Audio, video and chat


10

PSTN to IP Call (Direct Inward Dial)PSTN to IP Call (Direct Inward Dial)

PBXPSTN

External T1/CAS

Regular phone(internal)

Call 93971341

SIP server

sipd

Ethernet

3

SQLdatabase

4 7134 => bob

sipc

5

Bob’s phone

• DID - direct and simple• No-DID - dial extension, supports more users

GatewayInternal T1/CAS(Ext:7130-7139)

Call 71342

713x is called a part of Coordinated Dial Plan (CDP) in a Nortel PBX


11

IP to PSTN CallIP to PSTN Call

Gateway(10.0.2.3)

3

SQLdatabase

2Use sip:[email protected]

Ethernet

SIP server

sipdsipc

1Bob calls 5551212

PSTN

External T1/CASCall 55512125

5551212

PBX

Internal T1/CASCall 85551212 4

Regular phone(internal, 7054)

Note: In this direction there is no distinction between DID and non-DID calls.


12

T1 Line Configuration T1 Line Configuration (From the PBX Side)(From the PBX Side)

• Electrical/physical settings– T1 type: Channelized, PRI– Characteristics: line coding - AMI, B8ZS; framing

- D4, ESF

• Trunk type: DID, TIE

• Channel type: Data, Voice-only, Data/Voice

• Access permissions: adjust NCOS for internal T1 trunk and CDP routing entry (713x)


13

VoIP Configuration in the VoIP Configuration in the Gateway: Dial PeersGateway: Dial Peers

• Dial Peer for PSTN to IP calls:dial-peer voice 1 voip destination-pattern 713. voice-class codec 1 session protocol sipv2 session target ipv4:128.59.19.141

• Dial Peer for IP to PSTN calls:dial-peer voice 1000 pots destination-pattern ((70..)|(71[0-24-9].)) no digit-strip port 1/0:1

• Regular expressions to avoid ambiguity


14

Dial Peers for non-DID callsDial Peers for non-DID calls

• Example for a mix of DID and non-DIDtranslation-rule 7138 rule 1 71381.% 1 ANY abbreviateddial-peer voice 1 voip destination-pattern 713[0-79] …dial-peer voice 2 voip destination-pattern 7138T translate-outgoing called 7138 …

• Caller dial 939-7138, then punch in a 3-digit extension of the form 1xx.


15

VoIP Configuration in VoIP Configuration in sipdsipd:: Dial PlanDial Plan

sip:7134@sipd-hostcanonicalize using dial plan

tel:+12129397134Find tel: uid in SQL PrimaryUser Table

Locate user’scontact information

sip:5551212@sipd-host

• PSTN to IP call • IP to PSTN call

sip:bob@sipd-host

canonicalize using dial plan

tel:+12125551212

Verify caller’sprivilege

Locate propergateway

sip:85551212@gw


16

Example Dial PlanExample Dial Plan

• Dial plan mapping for IP to PSTN calls# Intra-department calls7[01]?? tel:+1212939$# Local (same area code) calls??????? tel:+1212$# Remove dial-out prefix ‘8’(8)???????tel:+1212$# International numbers(011)* tel:+$(8011)* tel:+$


17

SecuritySecurity

• Goal: prevent unauthorized users from making certain (e.g., long-distance) calls

• Where to put authentication modules:– In the gateway (requires vendor’s support)– Or, its associated SIP proxy server

• Prevent direct calls that bypasses the proxy• Enforce signaling path using IOS access control

• SIP authentication– Digest, Basic, PGP


18

Gateway Selection and PrivilegesGateway Selection and Privileges

• Approaches– RFC 2916: ENUM, E.164 based on DNS

– RFC 2871: TRIP, allows optimization

– Static routing file, used in sipd(+1212939)7[01]?? full,guest sip:[email protected]

– full and guest are user’s gateway classes

– The server may terminate the call if caller has no sufficient privileges.


19

Sample Access Control List (ACL)Sample Access Control List (ACL)• Configure NIC to use ACL 101 (in packets)

interface FastEthernet0/0 ip address 128.59.19.28 255.255.248.0 ip access-group 101 in

• Definition of ACL 101access-list 101 permit ip host 128.59.19.141 anyaccess-list 101 permit udp 128.59.16.0 0.0.7.255 \ range biff 65535 host 128.59.19.28 neq 5060

• SIP requests (destination port 5060) allowed from only the designated proxy host

• Multimedia (RTP) packets treated otherwise


20

CINEMA: Columbia InterNet CINEMA: Columbia InterNet Extensible Multimedia ArchitectureExtensible Multimedia Architecture

• Web interface– Administration– User configuration

• Unified Messaging

– Notify by email

– rtsp or http

• Portal Mode– 3rd party IpTelSP


21

Scalability via DNS SRVScalability via DNS SRV

• A simple load balancing schemeexample.com_sip._udp 0 40 a.example.com

0 40 b.example.com0 20 c.example.com1 0 backup.somewhere.com

• a and b each receives 40% of total request

• c receives remaining 20%

• backup server for fault tolerance


22

Scalability ContinuedScalability Continued

• 2-stage load balancing based on DNS SRV

• Stage 1: stateless routing based on hashing

• Stage 2: – Hashed clusters

– Stateful proxy

• Redirect feature


23

Scalability of Media ServersScalability of Media Servers

• Media packets => more load than signaling

• rtspd: multiple server selection: static/dynamic

• sipconf: tree structure• Bandwidth savings similar to multicast

• Added packetization and playout delay


24

Scalability of Gateway and LANScalability of Gateway and LAN

• 1 T1 line = maximum 24 voice channels– Multiple T1 lines or gateways– IP Centrex service by carrier PBX with ethernet

• LAN bandwidth limitations (gateway calls)

• Silence Suppression: 40-45% activity factor

• Faster Ethernet interface (10 => 100 Mb/s)

Codec Bit-rate Net bandwidth Gross (IP/RTP/UDP)

PCM µ-law 64 kb/s 3.072 Mb/s 3.84 Mb/s

G.729, 20ms 8 384 kb/s 1.152 Mb/s

G.729, 40ms .. .. 768 kb/s


25

SNMP Support in SNMP Support in sipdsipd

• sipd status

• Details of active transactions

• User contact info


26

Detailed SNMP MIBsDetailed SNMP MIBs

• Details of active transactions

• User contact info


27

Future WorkFuture Work• Additional services

– PIN numbers for telephone users– Automated, electronic billing– Instant messaging– VoiceXML (e.g., email access via PSTN)

• Performance and scalability: – sipd, rtspd, sipconf– SQL main-memory vs. disk database

• Firewall/NAT interoperability• Details of system to appear in Tech Report


28

ConclusionConclusion

• Initial field test experience with deploying IP telephony in a campus environment

• The architecture and installation experience can be used at other organizations

• Issues raised for further study:– Service availability/reliability– Quality of Service (QoS)– Privacy/encryption– Electronic billing policies

towards junking the pbx: deploying ip telephony wenyu jiang, jonathan lennox, henning schulzrinne...

Documents

ip calls

dial peers dial peer

dial extension

dial plan sip

caller dial

plan dial plan mapping

pstn calls

abbreviated dialpeer