VoIP Open Source and TCO
We will endeavour to explain the likely costs of choosing VOIP as a telephone solution for your business and the choice between licensed proprietary or Free and Open Source Software (FOSS).
Telephony has evolved, traditional telephony providers have their network for their customers, with the internet we are all on one network making convergence inevitable. Voice is just another form of data, making it easy to integrate your telephony into your network infrastructure and cabling affordably, growing organisations will appreciate the flexibility that a robust VOIP solution will provide. By managing your own telephone system, and using Free and Open Source Software (FOSS) you will not only gain from substantial savings and flexibility but also add to the sustainability of your organisation and reduce the Total Cost of Ownership (TCO) of your communications systems. The world of telecommunications relies heavily on jargon therefore we will have introduce some of the common anachronisms you will encounter. Glossary
PSTN Public Switched Telephone Network. This is the familiar telephone service based on copper wire that carries analogue voice data. It is also called POTS (Plain Old Telephone Service).
ISDN Integrated Services Digital Network An international standard for end-to-end digital transmission of voice, data, and signalling.
PBX A PBX is a private telephone switch that provides switching (including a full set of switching features) for an office or campus. PBX often use proprietary digital-line protocols, although some are analogue-based.
VOIP Voice over Internet Protocol. The technology used to transmit voice conversations over a data network using the Internet Protocol. Such data network may be the Internet or a corporate Intranet.
SIP An application-layer control protocol, a Signalling protocol for Internet Telephony. SIP can establish sessions for features such as audio/video conferencing, interactive gaming, and call forwarding to be deployed over IP networks thus enabling service providers to integrate basic IP telephony services with Web, e-mail, and chat services.
Why to Consider A VOIP Solution? Cost Savings on your office PBX
Traditional PBX's are expensive and are usually totally proprietary, this means that any moves and changes (adding extensions, programming IVR's (Auto Attendant), adding ring groups, etc...) usually have to be carried out by the supplier. In addition the traditional PBX manufacturers have embraced VOIP in a proprietary manner, they may offer you a VOIP card for your PBX, but the licensing culture remains the same, typically you will be offered an additional 5 or 10 user licence and your need for more lines will require yet more licences.
Cost savings on Long Distance Calls
The cost savings off using Voip with a Provider can be tremendous, for example www.voipcheap.com and www.voipbuster.com let you sign up for a free account and give you 90 days of discounted calls (providing you deposit €10), calling landlines in the UK and most of Europe and North America are free and UK mobiles work out at 8p a minute.
Multi Sited Business
With VOIP you link all your sites across the Internet, typically providing an integrated dial plan, any calls made to those extensions will be in effect free, all you will pay for is the internet connection.
Wireless Building to Building links (Intranet)
Many Businesses operate from Business Parks, typically to trunk data from building to building has required that you contact the Park owner to find out who owns the ducting, then contact the owner (Normally BT) and negotiate a price to lay your cables in their ducting, this often costs £3-7K a year. By doing away with this and using Wifi or Laser links you can extend your Intranet, save the rental of fibre and ducting and provide telephony to your buildings.
Voicemail2Email
Many VOIP solutions provide voicemail to email, this may not sound that exciting but in the real world it can really make the difference, between missing an opportunity or not, for example when on the road all it takes is a quick visit to a wifi hotspot and you can collect your voicemail and respond then and there.
Sip URI dialling
Sip uri (universal resource identifier) dialling is one of the most future proof concepts of VOIP. Just as we use Domain Name Servers (DNS) instead of having to remember IP numbers for websites, URI dialling means you don't have to remember telephone numbers.
e.g. which is easier to remember
+441173250067
The above real life example rings a group of 4 phones in the bristol wireless main office and falls over to voicemail if unavailable. A good example would be web based click through dialling, much like clicking on an email address will open your mail client. clicking a sip address will open your phone and commence dialling, some VOIP phones have this built in as a standard feature.
Homeworkers
Homeworking is becoming more and more popular, rather than than renting expensive office space and wasting both time and resources commuting, work from the comfort of your own home. Using either a hardware telephone or software telephone staff have their extensions set to ring from 9-5 weekdays and transferred to another number or voicemail when out of hours. The IAX2 protocol even will work behind firewalls.
Conference Calling
Its good business hosting conference calls, depending on the frequency and number of users this can run into thousands of pounds over a year. A VOIP PBX e.g. Asterisk normally provides this feature internally, you decide how many users you want, ring them and transfer them to the conference or provide an external number for them to dial directly. In this way you will just pay for the call charges and still get the service.
Virtual Office
Does your organisation need a central office? With a VOIP PBX either co-located or at your home you can present your organisation as being geographically anywhere. For instance, providers such as gradwell.net and voiptalk.org, will provide you with incoming geographical telephone numbers for as little as £3 a month. Therefore if you want to look like a London based business you can answer your 0207 number from the comfort of your home in Cornwall or your Dumfries number from Paris. Anywhere in the world you can get Internet Access, off piste in the Alps perhaps?
Satellite Offices with limited telephone access
Providing telephone access to remote sites is costly with prices starting at £80 a month for a BT Business line, triple that to 3 lines and see the costs escalate. With just one line enabled with ADSL you can have multiple VOIP extensions. For instance you share an office with colleagues and you would all like to have own usual work extensions.
Types of VOIP PBX available
There are many manufacturers embracing VOIP. However most of them either use there own proprietary VOIP protocol, Nortel (UNISTIM) Cisco (MGCP, SCCP). However, SIP is an open standard that is becoming increasingly popular with large scale manufacturers who are now implementing this feature into their new products. The importance of an open standard is that as, SMTP POP and IMAP are standards for relaying and delivering email over the internet SIP is the standard for VOIP, just as you wouldn't buy an email system that was closed, one that couldn't integrate with anything else. Why would you buy a phone system that was closed? A majority of “off the shelf” proprietary systems are merely PBX's with a VOIP card added. SIP integration is essential.
Open Source PBX Asterisk
Asterisk has all of the features provided by the typical PBX, call centre queuing, auto attendant (IVR), call diversion, multiple hunt groups, Music On Hold (MOH), call recording, Call Logging (CDR) and much much more. It will run on Linux OSX and Open BSD. It's also the only telephony device that connects to every type of telephone system avialble (POTS/PSTN, ISDN2, ISDN30, SIP, H232, SCCP, MGCP, UNISTIM, IAX2 and Carrier Grade SS7), making it the most widely deployed VOIP PBX. Asterisk can also be integrated into any ODBC compliant database, allowing your organisation, for example, to integrate its Customer Relationship Management (CRM) with its telephone system. There are many simple web based interfaces written to configure asterisk (e.g. FreePBX), whilst not offering every feature available, they make moves and changes to your extensions and lines easy to manage and configure. It has a large community of developers and users willing to offer support.
SIPX
SipX is a pure SIP telephony solution. The SipX IP PBX has been successfully deployed in a lot of places. The largest known installation serves more than 5,000 users connected to one redundant High Availability (HA) system . Small installations for 2 to 10 phones can be served by very low hardware cost.
OpenPBX
OpenPBX is a community-driven vendor-independent cross-platform open source PBX software project that was originally derived from asterisk. OpenPBX is a fully featured PBX in software. It supports analogue and digital telephony, multi-protocol VOIP, fax, software-fax, T38 fax over IP and many more telephony applications ie. IVR (auto attendant), conferencing and call centre queue management.
SER (SIP Express Router)
SER is a high performance, configurable, free SIP server. SER's major advantage is its interoperability. it has been reliably tested with products from Microsoft, Cisco, Mitel, snom, Pingtel, Siemens, xten, and many others. Capable to throughput thousands of calls per second on a dual-CPU PC (the capacity needed to cover Los Angeles Bay Area) and hundreds on a Compaq IPAQ.
Glossary 2
PRI/E1 A digital transmission link with a capacity of 2.048 Mbps, used predominantly in Europe. E1 is channelized into 32 DS0s, each capable of carrying a single voice conversation or data stream.
BRI Basic Rate Interface One of two ISDN subscriber interfaces in ISDN. 2 voice (B) channels at 64 kbps channels and 1 data signal (D) channel at 16 kbps. The B-channels are for voice, video, and data. The D-channel is for signaling between telephone company switches and for carrying ISDN user-network messages.
H.323 Is an umbrella recommendation from the ITU-T, that defines the protocols to provide audio-visual communication sessions on any packet network. It is currently implemented by various Internet real-time applications such as NetMeeting and GnomeMeeting (the latter using the OpenH323 implementation). It is a part of the H.32x series of protocols which also address communications over ISDN, PSTN or SS7. A challenger to H.323 is SIP, a standard from the IETF.
IAX2 Is the Inter-Asterisk eXchange protocol used by Asterisk, an open source PBX application from Digium. It is used to enable VoIP connections between Asterisk servers, and between servers and clients that also use the IAX protocol.
ENUM Is a standard adopted by the Internet Engineering Task Force (IETF) that uses the domain name system (DNS) to map telephone numbers to Web addresses or uniform resource indicators (URI). The goal of the ENUM standard is to provide a single number to replace the multiple numbers and addresses for an individual's home phone, business phone, fax, cell phone, and e-mail. The ENUM standard is a joint effort of Telecordia and Verisign.
Codecs Codecs are used to convert an analogue voice signal to digitally encoded version. Codecs vary in the sound quality, the bandwidth required, the computational requirements, etc. GIPSFamily - 13.3 Kbps and up GSM - 13 Kbps (full rate), 20ms frame size ILBC - 15Kbps,20ms frame size: 13.3 Kbps, 30ms frame size ITU G.711- 64 Kbps, sample-based Also known as alaw/ulaw ITU G.722 - 48/56/64 Kbps ITU G.723.1 - 5.3/6.3 Kbps, 30ms frame size ITU G.726 - 16/24/32/40 Kbps ITU G.728 - 16 Kbps ITU G.729 - 8 Kbps, 10ms frame size Speex - 2.15 to 44.2 Kbps LPC10 - 2.5 Kbps DoD CELP - 4.8 Kbps
Bandwidth to Lines Calculator Use the calculator below to see many lines you can get. http://www.erlang.com/calculator/lipb/
Hardware Phones and Software Phones There are many different handsets headsets and software packages that work as telephones. In this section we compare some of the variety of solutions on the market. Prices vary dramatically as does software performance configure ability and interoperability. Below is a brief selection from the the most popular manufacturers.
Hardware VOIP Telephones ATA's or Analogue Telephone Adaptors These consist of a small box with ethernet, power and either 1 or 2 telephone sockets. An analogue telephone is plugged in the socket and ethernet into your network. Most ATA's available are SIP based, however Digium the company behind Asterisk do make an IAX2 version that works well behind firewalls. Linksys SPA-3000 Analogue Telephone Adapter SIP around £70 Zyxel Prestig 2002 ATA around £45 Grandstream HandyTone 286 ATA-286 around £40 Digium IAXY Analog Adapter IAXY around £55
SIP Hardware Phones SNOM Snom make excellent phones. They can support up to 12 lines making them an ideal reception tool. They are open source and running a linux kernel. Snom phones support direct sip:uri dialling. Priced around £80 - £200 each depending on model.
Cisco Cisco make a variety of VOIP telephones and PBX's, they are functional and fairly easy to configure, however they suffer from Cisco's usual licensing agreement, meaning that getting firmware for the phones is only legal from Cisco resellers, typically they require payments for even minor firmware upgrades, they also only support certain codecs. Cisco VOIP phones cost between £200 - £300 each.
Aastra Aastra took over Nortels deskphone division and produce a large range of SIP phones. Aastra phones can be configured without a PBX for calling each other, however if you want to connect them to the PSTN a PBX is required, Aastra themselves produce the VentureIP Gateway supporting up to 4 analogue PSTN lines. Aastra phones are available at £80 - £150 each.
Linksys Linksys produce the SPA9XX range of IP phones which are all robust and available in several configurations and supporting from 1 line to 4. Most models feature Power Over Ethernet (POE) as standard. Interestingly Linksys, being owned by Cisco they come with similar licensing issues. Phones are in the £50 to £130 range.
SIP and IAX Softphones A softphone will run on your laptop or desktop computer and provide telephony via a microphone and speakers or even better via a headset or bluetooth earpiece.
Counterpath provide the free xtensoftphone for Windows, Mac and Linux and also the pay for SIP Video Phone, Eybeam. The xtensoftphone supports direct URI dialling.
Sj-labs Sjphone works on multiple platforms including PDA's and IPAQ's.
Kiax Kiax is a software phone (soft phone, VoIP client) with a simple and comfortable user interface for making VoIP calls to Asterisk PBX. Using Asterisk's IAX2 protocol is easy for call configuration and audio setup.
Moziax MozIAX, a Firefox VoIP extension is a cross platform software IAX2 phone (softphone) that can be be used with Asterisk, the Open Source PBX. Like Asterisk, MozIAX is free software
Twinkle Twinkle is a soft phone for your voice over IP communications using the SIP protocol. You can use it for direct IP phone to IP phone communication or in a network using a SIP proxy to route your calls. Currently Twinkle is available for Linux only (GPL license).
Skype Whist fine for home use, Skype is not a telephone. It is proprietary, comparatively expensive and works in a peer to peer environment which raises concerns over security, viruses and malware.
Example PBX prices
Sample Specifications of An Asterisk PBX for 10-20 SIP/IAX2 extensions, 4 analogue phone lines Pentium3 800Mhz 512Mb RAM 40 GB hdd approx £50 secondhand Digium TDM04B 4 analogue lines approx £200 Your Linux Distribution of choice nothing!
Sample Specifications of An Asterisk PBX for 50-100 SIP/IAX2 extensions 30 digital (ISDN) lines HP Proliant 3.2Ghz Xeon 2x SATA 80Gb RAID approx £1100 Digium TE110P or Sangoma A101 approx £300 Your Linux Distribution of choice nothing!
Proprietory Hardware Comparison (remember the obligatory annual support and update costs) Epygi Quadro 4x 2 analogue lines, 4 analogue extensions, 16 SIP extensions approx £1,575.00
Epygi Quadro 16x 4 analogue lines, 16 analogue extensions, 48 SIP extensions approx £2,180.00
AYC Telecom IPcts-E IP PBX 4 IP trunks x 50 extensions approx £1300 (analogue and digital trunk cards extra)
Nortel BCM50, BCM400Only available from Nortel resellers so prices are not available for these products without a site visit from a saleman.
Cisco Gatekeeper Series
Literally the skies the limit!
summary The traditional telephony industry is a can of worms, salesmen can promise the earth but very rarely will provide it. Recent examples include; selling a customer an ISDN30 circuit (£10K annual rental), 9 months later telling the customer they cant get the cable on site! Selling a PBX that is capable of voicemail integration, but not mentioning until installation that it only worked with MS Outlook 2003 and so on... Similarly to most Free and Open Source Software (FOSS) Asterisk can be configured in the way that best suits you and your budget. Being open means that it is constantly being improved, new features are available and bugs removed. Vulnerabilities are continuously monitored, fixed and security patches are freely available. In contrast, Proprietary hardware is expensive and less configurable. Per seat licensing and support costs are considerably higher.
Technical Information The best source of information regarding VOIP is seems to be http://voip-info.org which is dedicated to everything related to VOIP, software, hardware, service providers, reviews, configurations, standards, tips & tricks and everything else related to voice over IP networks, IP telephony and Internet Telephony. It is a collaborative document format known as a WiKi, the content is generated by the people who actually implement and support VOIP systems.
ISP's and Telephony Providers are Blocking VOIP Traffic Just as recent legislation in the UK requires most ISP to block port 25, for reasons of "Security". Some Countries are becoming very protective of their national telephony infrastructure and see the adoption of VOIP as a threat to there revenue stream. For example in India it is ILLEGAL to connect a VOIP equipped PBX to the telephone network. In other countries like the United Arab Emirates Internet Providers physically block the ports required for VOIP. i.e. a colleague was going to use mobile telephone access to the internet (GPRS) but found out his contract agreement specified “if VOIP is detected it would result in his disconnection......” not just from the internet but from the mobile network! Changing providers is simple when using VOIP.