Voice over IP at UCSBBruce Miller – UCSB Communications Services

OutlineTraditional Telephone ServiceVoice over Internet Protocols (VoIP)Implementations at UCSBIssues at UCSB

BackgroundIn 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.

Public Switched Telephone Network (PSTN)

Trunks

Switch Switch

Digitized VoiceWhile the standard telephone instrument today remains analog,

telephone switches and telephone trunks use digital transmission.Today voice is generally digitally encoded at 64 kbps (kilobits-per-second).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 TrunksTrunk lines today generally use digital transmission with a 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 TelephonesOver 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.

Public Switched Telephone Network (PSTN)

Analog Lines

Digital

PBX

Digital Telephone System

Trunks

Voicemail

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 elementsBasic elements of a VoIP system

Data NetworkSet 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 proxyA gateway may be used to connect a VoIP system or

network to a traditional telephone switch or network

VoIP means different things to different peopleThere 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

Public Switched Telephone Network (PSTN)

Central Gateway

“Soft” Phone

Trunks

“Private” Wide-Area Network

Internet

Departmental/Building Network

Departmental/Building Network

Trunk to Dept. VoIP Gateway

Analog to VoIP adapter (FXO)

Peer-To-Peer (e.g. Skype)

VoIP Based Carrier (e.g. Vonage, magicJack)

VoIP Gateway Service magicJack

SIP client app. using data connection on Cellular Phone

Wireless Network

SIP Phones

Campus Backbone NetworkCentral Gateway

P2P to gateway

VoIP Carrier Service (e.g. Vonage)

Analog Phones

PBX

What is Communications Services Doing with VoIP?Evaluating opportunities to use the technology for serving

off-campus locations and as a potential replacement for digital key systems

Evaluated Cisco Call ManagerPrototyped extending analog lines with Linksys FXS adaptersVoIP over radio link to Santa Cruz IslandPrototyping of 3CX system as a potential replacement

instead of digital key systemsEvaluated Linksys, Aastra, Polycom instrumentsStarted evaluating SipXecs

Campus VoIP ImplementationsKITP – 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 landscapeSIP based telephone instruments are becoming commodities. 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 BenefitsRelocating instruments without central administrative supportPossible savings on wiring costs for new buildings and

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 IssuesReliability, Availability and SupportE-911 ServicesDirectory ServicesLoss of StandardizationShifting Support RolesFunding of existing “embedded” Communications

Infrastructure and Services

Reliability, Availability, Support - Some form of reliable communications is considered a life-safety issue

Finding People – 911 Calls911 Emergency calls must be directed to the correct (closest) Public

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 - DirectoryThe 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 StandardizationVoIP 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 RolesThe current departmental VoIP implementations still relieve 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 fundingTelephone 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 usageThe current “standard line” telephone line charge includes a

$6.50/line “Data Network Surcharge” which funds:Campus use of CENIC NetworkCampus membership in Abilene & Internet2Campus ISP traffic to the commodity InternetAcquisition & Maintenance of Border Router EquipmentCommon infrastructure supporting connection