Trends in Enterprise VoIP VoIP as a catalyst for enhanced Business Services.
VoIP for Business
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Transcript of VoIP for Business
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VoIP for Business
A discussion of issues relating to deploying Asterisk based VoIP systems in a business setting
Tim FrichtelTimFrichtel at yahoo dot com
Scale 4X – February 12, 2006
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Disclaimer
● Views expressed are my personal opinions, not those of my employer
● Not affiliated with any vendor mentioned
● I'm not an Asterisk developer
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Presentation Goals
● Overview of VoIP● Business implementation of Asterisk● Lessons learned● Costs● Resources
● Cheap long-distance not covered● Configuring Asterisk not covered
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VoIP vs PSTN
● PSTN– Public Switched Telephone Network– Calls go through physical circuit paths– 100+ years of maturation/optimization
● VoIP– Calls are packetized, carried over IP– Can be peer-to-peer or through a PBX– Systems can be open or proprietary
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Why install VoIP?
● New calling features● Better integrate voice & data (CTI)● Easier/cheaper administration● Increased redundancy
● Maybe to reduce long distance costs
● Not to get “better” voice● Not necessarily to get better support
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Open Source VoIP Pitfalls
● Compared to PSTN, VoIP is immature● No single vendor “Turn-Key” solutions● More features = greater complexity● Total control = Total responsibility
– For configuration– For testing– For integration– For troubleshooting
● You are the integrator!
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Why Open Source VoIP?
● To eliminate vendor lock-in● To take total control of system● To avoid license costs● Choice in:
– Phones– PBX Hardware– PSTN Gateways– SIP Service providers
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Why we chose Open Source
● Cost● Speed of deployment● Desire to avoid vendor lock-in● Wanted to really “own” system● Comfort with Linux and other OSS
● No big vendor would do a pilot
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Our Requirements
● Single Building● 100 employees + conference rooms etc.● Receptionist● Overhead Paging● Voicemail● Conferencing● A few Fax & Modems● NOT a call center
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Business VoIP System
BackupAsterisk
PBX
Switch
SIPRegister
SIPRegister
T-1(s)
T-1(s)
Analog line(s)
Analog line(s)
Fax, Modem, Alarm analog Lines
DHCPFTP
Server
Phone Config & Log Files
SIP Provider (optional)
Asterisk PBX
Spkr
pagegateway
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Choosing a Prototype
● Implement a small system for test group● Researched OSS & Commercial
– Cisco, Nortel, Mitel, Zultys, Siemens– Asterisk, Pingtel– Full system quotes - from $85,000 to $110,000
● Asterisk looked (and IS) hard
● Hybrid OSS/Commercial chosen– Switchvox (based on Asterisk)
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Switchvox selected
● Switchvox targets smaller deployments● Vendor support valuable● Low cost, Low Risk
– Standard Dell Server– Polycom Telephones– Pre-configured to our needs
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Switchvox Pilot Solution
● Dell 1800 Server● Switchvox software
– Fedore Core– Asterisk– Proprietary Switchvox Admin Interface
● Digium TDM 400P Analog Card● Polycom 300 and 600 telephones● PBX & phones came pre-configured● It worked perfectly when we plugged it in!
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Test successful but...
● Analog PSTN interface prone to echo – eventually mostly resolved
● Didn't replicate every feature of existing Nortel Meridian
● Network and power challenges became obvious
● Pilot demonstrated how involved replacing a 10-year old stable system would be...
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Pilot Phase II
● Upgraded to 4 port Digium T-1 Card● Excellent tech support from Switchvox● T-1 cable was a guess● Using phones from other vendors was a
steep learning curve – but worked– Cisco 7940– Grandstream GX2000
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Pilot Lessons Learned
● Features are split between phone & PBX● Phones are “partners” with PBX● Configuring phones manually is tedious● Lots of data in a phone system● “Closed” Switchvox system:
– Great at what it does – No direct access to Asterisk– Out of luck if you want to do something it doesn't
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Prep to Deployment
● Designed PoE Network● Chose Telephones● Created database to track names,
extensions, telephone MAC address and PBX passwords for telephones
● Created app to generate XML files to configure Phones
● FTP server for phone configuration files & phone log files
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Business VoIP System
BackupAsterisk
PBXSIP
Register
SIPRegister
T-1(s)
T-1(s)
Analog line(s)
Analog line(s)
Fax, Modem, Alarm analog Lines
DHCPFTP
Server
Phone Config & Log Files
SIP Provider (optional)
Asterisk PBX
Spkr
pagegateway
PSTNInterface
PSTNInterface
PSTNInterface
UPS
PSTNInterface
UPS
UPS
Switch
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“Component” Costs
● Linux - $0● Asterisk - $0● PBX Hardware - $1,000 ~ $3,000● Telephones - $80 ~ $500 each● PoE Switch - $800 ~ $3,500● UPS for Server and PoE switch● PSTN Interface $200 ~ $5,000
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PSTN to VoIP Integration
● Telcom people don't know VoIP and IT people don't know Telecom
● Be prepared to learn Telecom
● You are the integrator
● Good Telecom support is critical
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PSTN Interfaces
● Analog– Analog Telephone Adapter (ATA) – Sipura– Analog Interface card* (Digium)
● Digital (PRI – T1/E1)– Internal Interface Card* (Digium, Sangoma)– External Router (Cisco)
Caution! - Internal PSTN cards aren't always compatible with, and may not fit in all PCs/servers
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PSTN Interface - Analog
● Analog Lines– Good for low external call volume– ATAs are inexpensive– Echo is a bigger issue– Analog lines cost tens of dollars/month– “Built in” fall-back – plug a telephone in!– Your phone number is tied to Central Office– Expensive for multiple DID (direct inward dial) lines
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PSTN Interface - Digital
● Digital Lines– T-1 PRI provides 23 circuits– Can be less expensive than same number of analog– Echo rarely an issue– PRI has more features than analog– Need expensive, specialized test equipment to
troubleshoot -- (or a vendor)– T-1 Interface card more expensive, but only real way
to go for many DID lines– Channel banks can be used to split analog lines out
of T-1 for fax/modem
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PBX Server Selection
● Buy reliable hardware– Redundant power supplies– RAID– ECC memory
● Moderate speed is OK● RAM > 512 (more for conferencing)● Disk – voicemail is biggest consumer● Check compatibility with PSTN cards!● UPS recommended
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Asterisk PBX Provides
● PBX functions to control calls● Voicemail● Call Conferencing● Music On Hold● Call Queues● Integrated Voice Response (IVR)● Integration with Data System● Call Reporting/Logging
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Asterisk Configuration
● Create SIP channels● Create PSTN channels● Create dialplan● Configure voicemail, email integration● Create IVR/Call Queues
● “User Friendly” interfaces are important– Switchvox (can't see Asterisk files)– Sigman (can see Asterisk files)– Asterisk@Home (can see Asterisk files)
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Telephones
● Choose SIP● Features, Cost & Quality vary widely● “Feel” and “Sound” of phones important● Message lights, buttons matter● Headsets support generally integrated● Check CODEC support● Power choices
– AC Adapter– Power Over Ethernet (may require special cable)
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Voice Quality Factors
● Handset versus softphone– Quality of sound card– Quality of microphone/earphones– PC stability and availability
● Codec - G711– free, high quality & bandwidth – for LANs
● Codec - G729 – licensed, good quality, low bandwidth – for WANs
● Others
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Voice Quality Factors -2
● Echo– Echo is part of telephone life– Is less with digital (T-1/E-1) PSTN connections– Some echo is good (side tone)
● Latency– Delay due to equipment, routing, distance etc
● Jitter– Due to dropped, out of sequence packets
● Dropped Packets– A few lost packets won't be noticed
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Telephones – PoE
● No need for power outlet by phone● Reduce cabling at telephone● Phones may use bulky, expensive cable● Only reliable if supported with UPS● Not available on all phones or ATAs
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Polycom PoE Cable
● PoE cable is 9' long
● PoE block is 1.5' from connector
● Must be connected to standard cat-5
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Telephones - Configuration
● Telephones require configuration● Most phones support:
– Keypad entry (limited and painful)– In-phone web server– XML file – best way to go
● Typically XML file:– Features– Phone extension– PBX passwords– Multiple PBX registrations
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Network - Dedicated/Shared?
● VoIP needs good networking● Shared VoIP & Data
– Less cost (only 1 set of switches)– Less wire / power / space– Quality of Service (QoS) require network skills
● Dedicated VoIP Network– Expensive– “Throwing money at the problem”– Avoids contention with data network
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Power over Ethernet (PoE):
● Two PoE Options:– PoE Switch (Power and Switch together)– Power Injector (between phone and switch)
● PoE switches– Are more expensive– Draw more power– Generate more heat– Have louder fans due to heat– May require external power supplies for
redundancy
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VoIP “Sidekicks”
● Optional but recommended● User & telephone database● Backup system for PBX config files and
Voicemail● TFTP, FTP or HTTP server to deliver
telphone config files● DHCP server to provide IP addresses to
phones
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Username/Password Mania!
● Phone – password to access Asterisk– password to change in-phone settings– password to access FTP site for config files– password for telephone web config pages
● User– Voicemail password– Web interface password
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● Admin– PBX server root– PBX administration interface– Telephone FTP upload password– Vendor support web site login– Network Switches– Backup server (if separate)
More Username/Password Mania!
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Bringing it All Together
● Build the network● Install & configure the server● Build the user database● Provision the telephones● Connect to the PSTN● Train the users
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Our System
● Live since October 2005● 2,000 calls/day● $39,000
– 2 Dell PBX (1 switchvox, 1 just asterisk)– 2 T-1 cards (1 per server) – 120 telephones– Dedicated voice PoE network– Overhead speakers for 30,000' office
● Employees happy
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Good Luck!
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Miscellaneous
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“Selling” Management on Asterisk
● Hire a vendor● Positive reviews in PC Mag, E-week etc● Demonstrate with low-cost pilot system
– Switchvox, Signate, Asterisk@Home● Demonstrate using multi-vendor phones● Showcase cool new features
– Email a voicemail– Use softphone
● Don't Over Sell!
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VoIP Resources
● www.voip-info.org (wiki)● Asterisk mailing list (not for the meek!)● Books:
– Switching to VoIP (O'Reilly)– Asterisk (O'Reilly)– VoIP Telephony with Asterisk (Signate)
● White Paper– Voice over IP 101 (Juniper Networks)
● Asterisk overview video – revision3.com/systm/ episode #5
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Resources – Asterisk Training
● Digium – 3 days, $1,500● Signate – 4 days, $1,795● Astricon – 5 days, $3,000
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Resources – End User Training
● Download telephone manuals● Create “help sheets”● Telephone demonstration classes
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Softphone Issues
● Softphone success depends on– Quality of sound card– Quality of microphone/earphones– PC stability and availability– PC network
● User reaction– Headsets can be uncomfortable– Not as convenient as physical phone– OK for travel and occasional use
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VoIP – Protocols
● Signaling to control calls– SIP = Session Initiation Protocol– SCCP = Skinny Client Control Protocol (Cisco)– H.323 – older, more complex– MGCP = Media Gateway Control Protocol
● Voice Data (“payload”) – RTP = Real Time Protocol
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VoIP – Common Terms
● PSTN = Public Switched Telephone Network
● Central Office = phone company center● ATA = Analog Telephone Adapter
– connects analog phones to VoIP● Foreign Exchange (Station, Office)
– FXS = Interface to connect analog telephone device to VoIP
– FXO = Interface to connect analog line to VoIP
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Overhead Paging
● Another Integration Challenge– Speaker System Choice & Design– Cable Installation– VoIP to Speaker integration
● Cobbled solution works!– Grandstream 2000 phone cut open to feed
amplified speaker system● Better solution coming
– Sipura ATA and Bogen TAM-B interface