Download Voice over IP at UCSB

Bruce Miller – UCSB Communications Services
Outline
 Traditional Telephone Service
 Voice over Internet Protocols (VoIP)
 Implementations at UCSB
 Issues at UCSB
Background
 In 1982 the divestiture or “breakup” of AT&T, “The Phone
Company,” created the opportunity for private entities to
own and manage their own telephone systems, thereby
saving money.
 Communications Services was created in 1983 to manage
campus communications wiring and in part to take
advantage of the opportunity to bring a private telephone
system (PBX) to campus.
 In 1984, when the campus acquired its first PBX, it also
acquired the underground wiring.
Communications Services Provides
 Voice Services to Business and Residential Customers
 Telephone Service – Service, Installation, and Repair
 Connectivity to off-site locations
 Campus Operator
 Emergency Telephones (about 300)
 Pass-thru Billing of Carrier Services and Records Management
 Network Infrastructure
 Campus Copper Cable Underground maintenance and management
 Campus Fiber Optic Cable Underground (and as far as West Campus & Railroad)
 Data circuit connectivity
 Financial and infrastructure support of campus “Internet” connection
 Planning and design for connectivity of buildings
 Review of wiring designs during construction/renovation
 Responding to “Dig-Alerts” to locate underground infrastructure
 Minor wiring projects
 Wireless Services
 800MHz Radio System used by Police, FM, Parking, H&RS, and others
 Coordination of Cellular Carrier (Cell-Site) installation activities
 Support for campus/carrier wireless issues
 Cable Television to Business and Residential Customers
Traditional Telephone Service
 “Plain-Old-Telephone-Service” (POTS) based on analog voice
transmission.
 An individual pair of wires is used to connect each telephone
line to a central switch which may owned by the telephone
company or privately (a PBX).
 Originally a multi-line telephone
required additional
cable pairs for each line.
 Telephone switches are connected together by tunk lines
(originally also analog), which are not dedicated to individual
telephone lines.
Trunks
Public Switched Telephone Network (PSTN)
Switch
Switch
Digitized Voice
 While the standard telephone instrument today remains analog,
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telephone switches and telephone trunks use digital transmission.
Today voice is generally digitally encoded at 64 kbps (kilobits-persecond).
This is the “standard” for “voice-quality” transmission.
Like music and video, many encoding schemes (codecs) are available.
Traditionally this encoding and decoding happens at the telephone
switch.
Analog signal still sent to the “standard” telephone instrument.
Caller-ID to an analog telephone is transmitted basically as “modem”
tones.
Beyond Caller-ID, very few “advanced” features are available on an
analog telephone line.
Digital Trunks
 Trunk lines today generally use digital transmission with a
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conversation encoded at 64 kbps (kilobits-per-second).
The standard trunk line is a T-1 1.544 Mbps (Megabits-per-second)
circuit which is divided with a technique called “Time Division
Multiplexing” (TDM).
TDM on the T-1 provides 24 64kbps “timeslots” or channels for
digitized voice. Each timeslot is allocated to a call for the duration of
that call.
Usually 1 channel allocated to provide call setup signaling
information such as Caller Number and Name.
The ISDN Primary Rate Interface (PRI) is the “standard” digital trunk
service.
Traditional telephone networks are sometimes referred to as
“Circuit-Switched,” indicating that an end-to-end circuit (or channel)
is established for the session. This is different than the “packet”
nature of Internet protocols.
Digital Telephones
 Over the last several decades telephone manufacturers
have developed digital telephone instruments which
provide numerous features not available with analog
instruments.
 These are generally proprietary and work only with that
manufacturer’s equipment.
 These digital telephones may connect directly to a PBX or
may be part of a departmental or building digital
telephone system (sometimes called a “key-system”)
which locally distributes telephone lines provided by a
central telephone switch.
Hybrid Analog/Digital Phone Environment at UCSB
 The telephone switch (PBX) at UCSB provides analog telephone lines which support any
standard analog telephone instrument.
 Telephone instruments are purchased by departments or projects.
 An analog telephone line may be connected to a departmental or building digital telephone
system.
 At UCSB there are about 120 Panasonic Digital Telephone Systems (DBSs) which require digital
Panasonic DBS telephones. Many of these were acquired as part of a building construction or
renovation project.
 These systems allow multiple lines to appear on a set and include features such as intercom.
 Approximately 1800 of about 4300 campus customer telephone lines are connected to a
Panasonic DBS system. There are more instruments than there are telephone lines.
Voicemail
PBX
Digital Telephone
System
Digital
Trunks
Public Switched Telephone Network (PSTN)
Analog
Lines
Emergency
Telephones
Voice over Internet Protocol
 Voice over Internet Protocol (VoIP) is a general term for a family of transmission
technologies for delivery of voice communications over IP networks such as the
Internet or other packet-switched networks. Other terms frequently encountered
and synonymous with VoIP are IP telephony, Internet telephony, voice over
broadband (VoBB), broadband telephony, and broadband phone.
 Internet telephony refers to communications services — voice, facsimile, and/or
voice-messaging applications — that are transported via the Internet, rather than
the public switched telephone network (PSTN). The basic steps involved in
originating an Internet telephone call are conversion of the analog voice signal to
digital format and compression/translation of the signal into Internet protocol (IP)
packets for transmission over the Internet; the process is reversed at the receiving
end.
 VoIP systems employ session control protocols to control the set-up and tear-down
of calls as well as audio codecs which encode speech allowing transmission over an
IP network as digital audio via an audio stream. Codec use is varied between
different implementations of VoIP (and often a range of codecs are used); some
implementations rely on narrowband and compressed speech, while others
support high fidelity stereo codecs.
-- Wikipedia
VoIP elements
 Basic elements of a VoIP system
 Data Network
 Set of common Protocols (e.g. RTP “Real-time Transport
Protocol”, SIP “Session Initiation Protocol”)
 A client (hardware “instrument” or software “soft-phone”)
 A call control mechanism or proxy
 A gateway may be used to connect a VoIP system or
network to a traditional telephone switch or network
VoIP means different things to different people
 There are both open and proprietary (manufacturer
specific) protocols for VoIP.
 VoIP can be implemented in many different ways:
 “Person to Person” (peer-to-peer or “P2P”) with software
such as Skype.
 As part of a traditional telephone network with VoIP
trunking between systems or connected to VoIP “Carriers.”
 With VoIP telephone systems connected to traditional
trunks.
 With VoIP telephone systems which can connect to other
VoIP telephone systems.
A few of the different VoIP models
SIP client app. using data
connection on Cellular Phone
VoIP Carrier Service
(e.g. Vonage)
Peer-To-Peer
(e.g. Skype)
VoIP Based
Carrier (e.g.
Vonage,
magicJack)
Wireless Network
VoIP Gateway
Service
magicJack
Internet
P2P to
gateway
Departmental/Building Network
Campus Backbone Network
“Private” Wide-Area Network
Central
Gateway
Central
Gateway
Analog to VoIP
adapter (FXO)
Departmental/Building Network
PBX
“Soft”
Phone
Trunks
Trunk to Dept. VoIP Gateway
SIP
Phones
Public Switched Telephone Network (PSTN)
Analog
Phones
What is Communications Services
Doing with VoIP?
 Evaluating opportunities to use the technology for serving off-
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campus locations and as a potential replacement for digital key
systems
Evaluated Cisco Call Manager
Prototyped extending analog lines with Linksys FXS adapters
VoIP over radio link to Santa Cruz Island
Prototyping of 3CX system as a potential replacement instead
of digital key systems
Evaluated Linksys, Aastra, Polycom instruments
Started evaluating SipXecs
Campus VoIP Implementations
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KITP – Cisco Call Manager (101 numbers)
Physics – Asterisk (FreePBX) (128 numbers)
Chemistry – Asterisk (FreePBX) (102 numbers)
These 3 implementations are connected to the campus
PBX with trunks. Calls are routed through the campus
PBX to other campus stations, local, and toll destinations.
Calls are recharged and statements generated along with
standard telephone lines.
 Various niche implementations
Dynamics of the VoIP landscape
 SIP based telephone instruments are becoming commodities.
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This provides competition, interoperability, and choice.
But there are still differences between manufacturers.
Durability and lifespan of VoIP instruments still unclear (the
average Panasonic on campus is probably 10 years old).
Voice Enabled applications are increasingly using VoIP
technologies instead of proprietary telephone interfaces.
2 major open source VoIP platforms, Asterisk and SipXecs.
“Unified Communications” and “Collaboration” tools are a
competitor of “traditional” VoIP models.
Wireless Carriers are a major competitor of wired voice
services of all kinds (traditional and VoIP).
A Few Potential VoIP Benefits
 Relocating instruments without central administrative support
 Possible savings on wiring costs for new buildings and
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renovations projects, with reduced copper “riser” cable.
However, inside “lateral” wire is already largely shared with
data, and it is unlikely copper underground can be eliminated
entirely any time soon.
Low cost or free toll calls to some locations (does require
configuration and management)
Advanced features and voice-data applications
“Roaming” between instruments (login to a phone)
Off-campus access to office number
Major VoIP Issues
 Reliability, Availability and Support
 E-911 Services
 Directory Services
 Loss of Standardization
 Shifting Support Roles
 Funding of existing “embedded” Communications
Infrastructure and Services
Reliability, Availability, Support - Some form of reliable
communications is considered a life-safety issue
Finding People – 911 Calls
 911 Emergency calls must be directed to the correct (closest) Public
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Safety Answering Point (PSAP) to obtain the quickest response.
The UCSB Campus Police dispatch is the PSAP for campus.
911 calls from 893 numbers are directed to our campus PSAP.
Calls from non-893 numbers are directed to ???
Owners of private telephone systems are responsible for
maintaining location information for telephone numbers they serve.
Location data for 893 telephone numbers is provided to our PSAP by
Communications Services’ Service Location Database.
Customers can (and do) move their own telephones. Updates can
(and should) be made on Communications Services’ Web site.
Customers can (and should) confirm their location information by
dialing 893-2300 (ALICIA).
Location data for non-893 numbers is provided by ???
Finding People - Directory
 The only nearly universal address for voice
communications is still the telephone number.
 Currently the primary single source for locating
individuals on campus is the LDAP directory.
 Campus operators rely on the LDAP directory data to
direct callers.
 The LDAP directory data is validated against working
telephone numbers for 893 numbers.
 Regardless of their telephone system, people should
update their directory data.
Loss of Standardization
 VoIP system dialing plans (how you dial a number) can be
different.
 VoIP instruments and systems provide an abundance of
advanced features – not all of them intuitive and not all
implemented alike.
 Even some basic dialing features can be different.
 Divergence of telephone numbers away from 893- over to
services like magicJack increases complexity.
 Call charges can result from dialing off-campus numbers.
Shifting Support Roles
 The current departmental VoIP implementations still relieve
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the implementers of the routing complexity and
billing/records management complexity.
However, departmental CNTs are taking on new
responsibilities for voice services.
Implementation of solutions such as magicJack which do not
interface with the PBX put the burden for recharge billing and
records management on the department.
Issues with departmental VoIP solutions require more
technical involvement on the part of Communications
Services.
Any “central” VoIP solution would most likely require either a
separate network or visibility into departmental building
networks and shared diagnostic efforts.
Funding of Existing Infrastructure and Services
 Communications Services receives no direct core funding
 Telephone Line and Usage charges support many of the services
listed on slide 3 (residential services are “self-supporting”)
 The expenses for many of these services are not reduced
appreciably by a decline in telephone lines or usage
 The current “standard line” telephone line charge includes a
$6.50/line “Data Network Surcharge” which funds:
 Campus use of CENIC Network
 Campus membership in Abilene & Internet2
 Campus ISP traffic to the commodity Internet
 Acquisition & Maintenance of Border Router Equipment
 Common infrastructure supporting connection