Towards Junking the PBX: Deploying IP Telephony

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Towards Junking the PBX Deploying IP Telephony
Wenyu Jiang, Jonathan Lennox, Henning Schulzrinne
and Kundan Singh Columbia University wenyu,lennox
,hgs,kns10_at_cs.columbia.edu
We describe our departmental IP telephony
installation
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Outline

• Introduction to IP telephony
• System architecture
• Call flows
• System configuration
• Security
• Scalability

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Traditional Telecommunication Infrastructure
7040
External line
212-8538080
7041
Corporate/Campus
Telephone switch
Another switch
Private Branch Exchange
7042
7043
Internet
Corporate/Campus LAN
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What is IP Telephony ?
Corporate/Campus
Another campus
7040
8151
External line
8152
7041
PBX
PBX
8153
VoIP Gateway
8154
7042
VoIP Gateway
7043
Internet
LAN
LAN
IP Phone Client
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IP Telephony Protocols
audio over RTP
Call bob_at_office.com
SIP server
home.com
office.com

• Contact office.com asking for bob

Session Initiation Protocol - SIP

• Locate Bobs current phone and ring

• Bob picks up the ringing phone

Real time Transport Protocol - RTP

• Send and receive audio packets

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Architecture
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Example Call (IP only)

• Bob signs up for the service from the web as
  bob_at_cs.columbia.edu

• sipd canonicalizes the destination to
  sipbob_at_cs.columbia.edu

• He registers from multiple phones

• sipd rings both ephone and sipc

• Alice tries to reach Bob
• INVITE sipBob.Wilson_at_cs.columbia.edu

• Bob accepts the call from sipc and starts talking

cs.columbia.edu
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Canonicalization

• Bob.Wilson

canonicalize
bob_at_cs
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Other 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

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PSTN to IP Call (Direct Inward Dial)

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

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IP to PSTN Call
Note In this direction there is no distinction
between DID and non-DID calls.
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T1 Line Configuration (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)

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VoIP Configuration in the Gateway 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 ipv4128.59.19.141
• Dial Peer for IP to PSTN calls
• dial-peer voice 1000 pots
• destination-pattern ((70..)(710-24-9.))
• no digit-strip
• port 1/01
• Regular expressions to avoid ambiguity

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Dial Peers for non-DID calls

• Example for a mix of DID and non-DID
• translation-rule 7138
• rule 1 71381. 1 ANY abbreviated
• dial-peer voice 1 voip
• destination-pattern 7130-79
• 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.

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VoIP Configuration in sipd Dial Plan

• IP to PSTN call

• PSTN to IP call

• sip7134_at_sipd-host

sip5551212_at_sipd-host
tel[phone]
tel[phone]
sip85551212_at_gw
sipbob_at_sipd-host
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Example Dial Plan

• Dial plan mapping for IP to PSTN calls
• Intra-department calls
• 701?? tel[phone]
• Local (same area code) calls
• ??????? tel1212
• Remove dial-out prefix 8
• (8)??????? tel1212
• International numbers
• (011) tel
• (8011) tel

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Security

• Goal prevent unauthorized users from making
  certain (e.g., long-distance) calls
• Where to put authentication modules
• In the gateway (requires vendors 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

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Gateway Selection and Privileges

• Approaches
• RFC 2916 ENUM, E.164 based on DNS
• RFC 2871 TRIP, allows optimization
• Static routing file, used in sipd
• (1212939)701?? full,guest sip_at_gw.office.com
  
• full and guest are users gateway classes
• The server may terminate the call if caller has
  no sufficient privileges.

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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 101
• access-list 101 permit ip host 128.59.19.141 any
• access-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

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CINEMA Columbia InterNet Extensible Multimedia
Architecture

• Web interface
• Administration
• User configuration
• Unified Messaging
• Notify by email
• rtsp or http
• Portal Mode
• 3rd party IpTelSP

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Scalability via DNS SRV

• A simple load balancing scheme

example.com _sip._udp 0 40 a.example.com 0 40
b.example.com 0 20 c.example.com 1 0
backup.somewhere.com

• a and b each receives 40 of total request
• c receives remaining 20
• backup server for fault tolerance

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Scalability Continued

• 2-stage load balancing based on DNS SRV
• Stage 1 stateless routing based on hashing
• Stage 2
• Hashed clusters
• Stateful proxy
• Redirect feature

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Scalability of Media Servers

• Media packets gt more load than signaling
• rtspd multiple server selection static/dynamic

• sipconf tree structure
• Bandwidth savings similar to multicast
• Added packetization and playout delay

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Scalability 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 gt 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
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SNMP Support in sipd

• sipd status
• Details of active transactions
• User contact info

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Detailed SNMP MIBs

• User contact info

• Details of active transactions

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Future 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

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Conclusion

• 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