Voice over IP at UCSB

1. Voice over IP at UCSB Bruce Miller – UCSB Communications Services

2. Outline • Traditional Telephone Service • Voice over Internet Protocols (VoIP) • Implementations at UCSB • Issues at UCSB

3. 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.

4. 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

5. 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 tunklines (originally also analog), which are not dedicated to individual telephone lines. Trunks Switch Switch

6. Digitized Voice • While 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.

7. Digital Trunks • Trunk 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.

8. 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.

9. 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 Analog Lines Emergency Telephones

10. Voice over Internet Protocol • Voice over Internet Protocol (VoIP) is a general term for a family of transmission technologies for delivery of voice communicationsover IP networks such as the Internet or other packet-switchednetworks. 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

11. 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

12. 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.

13. A few of the different VoIP models SIP client app. using data connection on Cellular Phone VoIP Carrier Service (e.g. Vonage) VoIP Based Carrier (e.g. Vonage, magicJack) Peer-To-Peer (e.g. Skype) VoIP Gateway Service magicJack P2P to gateway Central Gateway Central Gateway Analog to VoIP adapter (FXO) PBX “Soft” Phone Trunks Trunk to Dept. VoIP Gateway SIP Phones Analog Phones

14. 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 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

15. Campus VoIP Implementations • 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

16. Dynamics of the VoIP landscape • SIP 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).

17. A Few Potential VoIP Benefits • Relocating instruments without central administrative support • Possible 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

18. 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

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

20. Finding People – 911 Calls • 911 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 ???

21. 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.

22. 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.

23. Shifting Support Roles • The 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.

24. 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