Awhile back, We posted on ESI’s Presence Management, which features a special device called a Presence Management Reader that has multiple functions.  It can act as a door phone, as security access for building personnel (using RFID-based security cards or fobs), and more.

ESI presence Management Reader

We are pleased to announce that ESI has developed an IP version of its Presence Management reader.  We think this is exciting, because it allows for owners of all-IP ESI Communications Server phone systems to have Presence Management capability.  It also removes the limitation of 1,000 feet that exists with the original Presence Management reader (this was the maximum distance it could be installed from the phone server, or KSU).  In fact, a remote IP Presence Management reader can even be installed in off-site locations!  This does present interesting possibilities, like the ability to unlock a door remotely for example (after verifying the caller’s identity, of course, by talking to them via VoIP right through the reader itself).

ESI (our manufacturer of choice for business phone systems) announced recently that ESI has achieved interoperability with Broadvox GO! SIP trunking.  SIP trunking, for those who may not know, is a relatively new type of dialtone that is delivered via an IP network by an ITSP, or Internet Telephony Service Provider.  It’s far more efficient than Plain Old Telephone Service, and it appears to be where dialtone technology is headed.  Like ISDN PRI, it offers both DID (Direct Inward Dial) capability so that individual phone users can have their own phone numbers, and Caller ID name and number service is available.  In addition, SIP trunking allows businesses to be geographically diverse by obtaining phone numbers outside their normal calling area, if desired.  Additionally, ITSP service providers tend to charge less for their services than the traditional dialtone providers (sometimes substantially less).

While ESI has not yet formally released SIP trunking capability as a feature on its phone systems, clearly such capability is in the works.  According to the press release, ESI Communications Server phone equipment is what is being used to achieve this interoperability, and we anticipate that ESI will make this capability available to both new and existing users of its Communications Server phone systems.

We think this is great news.  It will not only give ESI Communications Server owners additional flexibility when choosing what dialtone type makes the most sense to them, but we are excited about SIP trunking in general, because it can give businesses more flexibility and allow them to save a substantial amount of money on their phone bills.  We’ll post more as more details are released.

Occasionally, you’ll see the word “convergence” come up in regards to phone systems.  But what does it mean?  In a word, flexibility.

Traditional, “old school” phone systems have always had two different types of ports: one type to support phone lines (dialtone), and another type to support the phones.  Since all phones used to be analog phones, this second port type would also handle fax machines, credit card machines, and cordless phones, all of which have analog interfaces.

Over time, phone system manufacturers realized that they could increase flexibility and features by making digital phones rather than analog ones.  Today, most phone system manufacturers who have been around for awhile offer digital phones.  Because each manufacturer has its own idea of what would make for the best features in a phone system, its own “killer apps”, and implementation of features, these digital phones are proprietary and cannot move from one phone system type to another.  Phone companies (dialtone providers), as well, realized that there were certain advantages to going digital, and began to offer digital phone lines.  By far the most common of these is a T1 line, which can carry up to 24 conversations simultaneously.  One flavor of T1 called ISDN/PRI is now the most popular due to it’s support of Caller ID (which a “plain” T1 does not support), and its enhanced signaling and troubleshooting capability (this is accomplished by having messages on the setup, progress, and teardown of calls take place on one of the 24 channels, leaving 23 available for voice conversations).  Most phone system manufacturers now support T1/PRI.

In the last few years, we’ve seen the emergence of VoIP (or IP) phone technology, which promises to offer certain advantages over regular digital phones.  The greatest of these is the ability to effortlessly move phones from one desk to another and keep the same extension, and the ability to use a phone off-premises and participate as if you were still in the office.  This last item is significant, as it gives businesses flexibility to allow remote workers the ability to handle office calls just as if they were there in person. (There are actually several “flavors” or different types of VoIP, which is treated in a separate post.)

Even with the marvelous capabilities that IP phones enable, digital phones are here to stay for the foreseeable future.  The reasons include, but are not necessarily limited to, IP phones being more expensive than their digital counterparts, and more potential for quality issues with VoIP.  Some manufacturers have decided to keep their digital phone systems and IP phones separate, meaning that if you decide digital phones are best for you now, but would like to have IP phones in the future, you’ll need to upgrade the whole phone system.  Other companies, typically ones who began life as data-driven companies such as Cisco, don’t offer digital (non-IP) phones.  They mask their lack of experience and R&D in this arena by claiming that IP phones are the best solution in every situation.

Other companies, such as ESI (whose phone systems we sell), have decided to take a converged approach, meaning that their phone systems will support either digital or IP phones equally well, in any combination.  If it makes sense for you to have digital phones inside the office (as we argue here is generally a good idea), and IP phones outside the office, you can do that.  If you’d only like digital phones now but would like to add IP phones later, you can do that as well.  In fact, ESI Communications Server phone systems support everything I’ve mentioned, including digital, IP or analog phones/devices, as well as analog lines or T1/PRI.  It’s a great day we live in, and flexibility is the name of the game.

In the open-source PBX world, one platform is king: Asterisk.  Asterisk is an open-source software PBX platform originally created by Mark Spencer of Digium. Because of the open-source nature of Asterisk, many companies have used the Asterisk code as a base for their product and modified the code to suit their needs. There are a host of products built on top of Asterisk, such as Fonality’s products PBXtra and Trixbox, IntuitiveVoice’s Evolution PBX, etc. This post applies to all of them to some degree. Also, there are other PC-based PBX solutions out there that run on Windows, such as 3CX or NCH’s Axon product. These share many of the same pitfalls as described below (but have the additional problems associated with Windows-based instability and security issues).

Open Source.
Asterisk is open-source software. It is controlled by a company named Digium, which issues periodic updates and bug fixes to the code. Because it is open-source software, this means that (1) Asterisk is free to anyone and can be downloaded without charge; (2) Anyone with the proper skills is able to view the original programming code and modify it for their own purposes. There are a number of companies that have grown up around this model and are selling their own “take” on Asterisk.

Hardware.
Most Asterisk builds use off-the-shelf PC hardware. Because there is so much to consider in regards to PC-based phone systems (whether tower-based or rack-based), I can‘t enumerate it all here.  Suffice it to say, PC-based phone systems suffer from a slew of drawbacks, including longevity, expandability, security, complexity, migratability, and recoverability issues, and they are more of a headache than they’re worth.

There are also a number of appliance-based Asterisk systems out there. Many of these are PCs in disguise. Others, that are truly appliance-based (you’ll often see the term “embedded” when seeing references to these systems) are generally for the smaller-end market. They feature hardware that has specifically been engineered for that specific product. Many times there is a single circuit board inside that replaces the functionality of multiple boards in other systems. The reliability of these products varies, but is generally good. If something needs to be replaced, however, the whole system usually needs to be swapped out.

Telephony cards.
In the case of PC-based implementations, and some appliance-based installations, the box will require certain telephony cards in most cases. This is to provide support for dialtone (analog lines or T1/PRI lines), as well as other analog devices such as cordless phones, credit card machines or faxes. These cards can be purchased from Digium, but the majority of these are purchased from third-party vendors, such as Sangoma, Rhino, OpenVox, ZapMicro, etc. With all of these choices comes different implementation issues, as each card has its own drivers, idiosyncrasies, implementation, support, reliability, and integration with Asterisk. Some cards are supported better than others. Some companies will only allow certain cards to be used with their implementation.

A note on echo. One of the pervasive challenges with Asterisk is that of echo. This generally happens when a digital signal (like IP) is being converted to an analog signal, or vice versa, and there is an impedance mismatch on the line (there usually is-it’s just the nature of the beast). This causes one or both parties to hear their own voice echo in their ear. While software-based echo cancellers are standard and tend to cut down on this, many users have reported serious echoing issues. The only proper way to minimize echo with Asterisk-based systems is to make sure the cards have HARDWARE based echo cancellation (the Echo Cancellation, or EC, version of any given card is more expensive). The advantages of hardware-based echo cancellation are (1) it can take care of some echoing that software sometimes cannot; (2) it is faster (although you may still have a couple of seconds of echo at the beginning of the call; (3) it frees up precious CPU resources from having to do the echo cancellation.

Telephones.
Digium does not make the telephones for its Asterisk software, nor do the other vendors whose PBX products are based on Asterisk. These are third-party phones from various manufacturers, such as Aastra, Grandstream, Cisco, Snom, and others. Each phone model from each manufacturer has a different look and operation than the others.

While “freedom of choice” is touted by Asterisk proponents, this disparity between phone models, and the integration of phones with Asterisk, is actually one of Asterisk’s greatest drawbacks. By not having proprietary phones, integration with the phone system is looser. Some of the phone manufacturers illustrate some “whiz-bang” applications that can be done on their phones, but the reality is that it’s difficult to incorporate into the Asterisk platform (more on this in a moment), and most IP phones used on Asterisk-based phone systems function little better than plain old analog telephones.

Getting the phones to work with Asterisk in the first place has traditionally been a challenge, although many strides have been made in this respect.

One thing that anyone who has been looking at Asterisk for any length of time will notice is that the phones in general are severely restricted in the number of buttons available for lines and features. This requires people to remember and dial feature codes manually for most anything they would like to do feature-wise, or work their way through on-screen menus.

Features.
Now for a paradox on Asterisk. Asterisk is, admittedly, a platform with perhaps the greatest potential for flexibility, and yet, in practice most people find that they are locked into the features provided out-of-the-box. It is true that because their code is open-source that in theory someone who knows how to program can make it do special things. It is also true that Asterisk has a scripting language that allows for special things to be accomplished. By going into special, text-based configuration files, one can, in theory, open up the flexibility potential. Here are some of the configuration files:

amportal.conf; asterisk.conf; autofs_ldap_auth.conf; capi.conf; cbmysql.conf; cdr_mysql.conf; conman.conf; dhcpd.conf; disa-1.conf; enum.conf; extensions.conf; extensions_additional.conf; extensions_custom.conf; extensions_hud.conf; extensions_override_freepbx.conf; ez-ipupdate.conf; features.conf; features_applicationmap_additional.conf; features_applicationmap_custom.conf; features_featuremap_additional.conf; features_featuremap_custom.conf; features_general_additional.conf; features_general_custom.conf; fxotune.conf; globals_custom.conf; gpm-root.conf; grub.conf; gssapi_mech.conf; host.conf; iax.conf; iax_additional.conf; iax_custom.conf; iax_custom_post.conf; iax_general_additional.conf; iax_general_custom.conf; iax_registrations.conf; iax_registrations_custom.conf; idmapd.conf; indications.conf; initlog.conf; jwhois.conf; krb5.conf; ld.so.conf; ldap.conf; lftp.conf; libaudit.conf; libuser.conf; localprefixes.conf; logger.conf; logrotate.conf; manager.conf; manager_additional.conf; manager_custom.conf; meetme.conf; meetme_additional.conf; mke2fs.conf; modem.conf; modprobe.conf; modules.conf; mtools.conf; multipath.conf; musiconhold.conf; musiconhold_additional.conf; musiconhold_custom.conf; my.cnf; nscd.conf; nsswitch.conf; ntp.conf; oddjobd.conf; op_astdb.cfg; op_buttons.cfg; op_buttons_additional.cfg; op_buttons_custom.cfg; op_lang_de.cfg; op_lang_en.cfg; op_lang_es.cfg; op_lang_it.cfg; op_server.cfg; op_style.cfg; pam_smb.conf; pear.conf; phone.conf; phpagi.conf; privacy.conf; queues.conf; queues_additional.conf; queues_custom.conf; queues_custom_general.conf; queues_general_additional.conf; queues_post_custom.conf; reader.conf; request-key.conf; resolv.conf; rtp.conf; sensors.conf; sestatus.conf; sip.conf; sip_additional.conf; sip_custom.conf; sip_custom_post.conf; sip_general_additional.conf; sip_general_custom.conf; sip_nat.conf; sip_registrations.conf; sip_registrations_custom.conf; smartd.conf; snom.cnf; sysctl.conf; syslog.conf; updatedb.conf; vm_email.inc; vm_general.inc; voicemail.conf; warnquota.conf; wvdial.conf; xinetd.conf; yp.conf; yum.conf; zapata.conf; zapata-auto.conf; zapata-channels.conf; zaptel.conf

Because of this maddening array of configuration files (which are not in one single place on the disk, by the way), several projects (including the makers of Asterisk itself) have developed web-based configuration screens so that you don’t have to delve into all of these. By doing this, however, they are necessarily limiting the number and type of special configurations that can be done.

Asterisk features a special scripting language to tweak the behavior of the phone system. Again, in theory it’s great, but only if you want to learn this:

exten => s,n,Set(FROMCONTEXT=exten-vm)
exten => s,n,Set(VMBOX=${ARG1})
exten => s,n,Set(EXTTOCALL=${ARG2})
exten => s,n,Set(CFUEXT=${DB(CFU/${EXTTOCALL})})
exten => s,n,Set(CFBEXT=${DB(CFB/${EXTTOCALL})})
exten => s,n,Set(RT=${IF($[$["${VMBOX}"!="novm"] | $["foo${CFUEXT}"!="foo"]]?${RINGTIMER}:”")})
exten => s,n,Macro(record-enable,${EXTTOCALL},IN)
exten => s,n,Macro(dial,${RT},${DIAL_OPTIONS},${EXTTOCALL})
exten => s,n,Set(SV_DIALSTATUS=${DIALSTATUS})
exten => s,n,GosubIf($[$["${SV_DIALSTATUS}"="NOANSWER"] & $["foo${CFUEXT}"!="foo"]]?docfu,1) ; check for CFU in use on no answer
exten => s,n,GosubIf($[$["${SV_DIALSTATUS}"="BUSY"] & $["foo${CFBEXT}"!="foo"]]?docfb,1) ; check for CFB in use on busy
exten => s,n,Set(DIALSTATUS=${SV_DIALSTATUS})
exten => s,n,NoOp(Voicemail is ‘${VMBOX}’)
exten => s,n,GotoIf($["${VMBOX}" = "novm"]?s-${DIALSTATUS},1) ; no voicemail in use for this extension
exten => s,n,NoOp(Sending to Voicemail box ${EXTTOCALL})
exten => s,n,Macro(vm,${VMBOX},${DIALSTATUS})

If you do have something custom programmed, you’ll need to do your best to ensure that it doesn’t get overwritten with a software update, or knowledge of it lost if the person who programmed it moves away.

Here is a real-world example of how the practical reality of Asterisk differs from theory. Consider some features that Asterisk-based systems have only recently added in, or that in some cases STILL have not been implemented yet. These are things that traditional phone systems have had for over 20 years:

• Day/night mode button (Fonality just implemented in 2008?)
• BLF, or Busy Lamp Field (Fonality just implemented in 2008?). This is where, by looking at a button on your phone, you can see if another person in your office is on their phone
• Shared Line Appearances (ability to press a specific line button to make a call, and see visually whether that line is in use by looking at the line button). Asterisk just added this capability in the last year, and some Asterisk-based systems still do not have this feature (including Fonality, based on their website).

Support.
Most national Asterisk-based vendors do not hire local support. There is normally an 800 number you can call with the major vendors, some of whom have 24-hour support. Bear in mind, however, that if and when your system goes down, there is only so much that can be troubleshot remotely. If the remote connection is down, then what? Most of these vendors do not have a mechanism in place for same-day parts replacement, let alone getting a body out locally to troubleshoot.

Not too long ago, I posted about ESI’s line of Cordless Handsets.  They were flexible, easy to use, and had a ton of features.  Now, they’re even better.

Introducing the ESI Cordless Handset II.  Like its predecessor, it comes in a digital version, local IP (LAN) version, and remote IP version.  Like the original version, it’s no ordinary cordless phone like you would get at Best Buy; with ESI’s cordless handsets, you have access to features like transfer, hold, conference, call record, voicemail, call page, door unlock and more without having to remember any feature codes.  You can also easily access or view the status of lines and other phone users thanks to several programmable, lighted buttons.  Here’s how the Cordless Handset II improves on the original:

• No more “large” and “small” versions of the phone: the new phone is the smaller size only.  Enhanced design also allows for the external antenna to be absent, further reducing bulk.
• Greatly enhanced battery life versus the original: fully charged, the new model allows for up to 16 hours of talk time, and about 7 days of standby time.
• Advanced DECT 6.0 technology allows for higher security, crystal-clear audio quality, and minimized or zero interference from Wi-Fi or other devices.
• 8 programmable buttons, versus 4 for the original model.
• Built-in speakerphone (the original did not have a speakerphone).
• Optional repeaters to extend wireless coverage over larger areas or between floors in a building.

One thing I love about the DECT 6.0 technology this phone uses is that it is so clear-sounding due to its digital transmission.  It also uses a different frequency band than other phones (1.9 GHz), and when it does encounter another device in that spectrum, it negotiates with it quickly and silently so that the devices do not interfere with each other.

Some of the more knowledgeable readers out there may ask, “Why 1.9 GHz?  Isn’t that taking a step backward from the relatively new 5.8GHz phones, or even the 2.4 GHz phones that came out before that?”  The answer is simple.  Higher frequencies are not always better–in fact, the higher the frequency in general, the harder it is for the signal to pass through solid objects like walls.  To compensate for this, manufacturers of the 2.4 GHz and 5.8GHz phones tend to boost the signal strength of the phone, which takes more power, which shortens battery life (or requires a bigger battery).  By using a lower frequency, DECT 6.0 phones are actually at an advantage in a typical office environment.

In short, DECT 6.0 is the latest, greatest technology, and paired with ESI features and technology, this phone is a winner.  But in order to take advantage of it, it must be used in conjunction with an ESI Communications Server phone system.  Fortunately, even small businesses can benefit from (and afford) a Communications Server.  If you’re in the Phoenix area and thinking about replacing an outdated phone system, give us a call.

VoIP, or Voice over IP, has become very popular over the last few years, and most businesses we come in contact with who are considering purchasing a new phone system want VoIP in some form, or at least want to know that the system they buy can support it.

(If you’re new to VoIP, feel free to read my other post on the different types of VoIP.  You can also check out our web site for more information on ESI Communications Server phone systems, which support VoIP.)

To be sure, VoIP gives tremendous advantages over traditional solutions, and we are very excited to offer it.  It should be noted, though, that with the nature of VoIP, a little more planning is required, and there are even times where VoIP may not make sense in your installation.

In contrast to traditional telephones, which usually use a pair of dedicated wires to transmit voice, IP phones can use the same network connection that your computer uses.  This is because in VoIP, your voice is broken up into many small bits (packets) at lightning speed and sent quickly to its destination over an IP network (such as your LAN), where the voice packets are reassembled and played back to the recipient.  This VoIP technology offers several applications that traditional telephony can’t match, but it also has the potential of causing some issues if you’re not prepared.

Jitter

When voice packets travel over a network to their destination, some packets may take a different route than others (routers determine this based on what appears to be the most efficient route at the particular millisecond that each packet passes through).  Because there can be some variability in the time it takes for a packet to reach its destination, this can create some issues.  If a particular packet does not reach its destination in time to be assembled at the other end in the right order, or if it does not arrive at all, it is not part of the voice conversation, and the person at the other end will get missing chunks or “stuttering” in the conversation.  Phone systems generally address jitter, which occurs in all IP-based conversations to a greater or lesser degree, with an internal jitter buffer, which helps to smooth things out.  No phone system can cope with excessive jitter, however (the larger a jitter buffer is in order to cope with jitter, the more it contributes to latency; see below).

Latency

Latency is the average time it takes for IP voice packets to get from one end to the other on an IP network.  The greater the latency, the longer it takes for you to hear what the other person is saying, and vice-versa.  In VoIP, there is always some level of latency, which is not necessarily a problem.  For most people, latency above 250 milliseconds will make for a noticeable delay and will start affecting the conversation (generally in the form of people talking over each other).

Packet Loss

As mentioned, the voice packets in a conversation can (and usually do) take different paths to their destination.  Sometimes, a packet may reach an unintended “dead end” and not make it to its destination, or a router along the way will become overwhelmed with traffic and drop some packets.  This is normal, and a few packets will almost always be dropped from a given conversation.  If the percentage of packets being dropped approaches 1%, however, it will make a noticeable difference in the conversation.  Packet loss above 2% may result in the conversation quality being unacceptable.

Echo

Echo happens all the time in telephony, in fact, almost constantly, though you almost never notice it.  In simplified terms, echo tends to happen due to an impedance mismatch on phone lines.  This is common.  In traditional telephony, the reason we almost never notice it is because when your voice bounces back to you and you hear it, there is a delay of only a few milliseconds.  Thus, instead of perceiving it as echo, you perceive it as if it is something called sidetone (this is when you hear yourself talk on the phone, which gives you the comfort that you’re not on a “dead” line).  With IP communications, voice takes longer to get from point A to point B and back (because at each router along the way, there is a small delay as the router processes it).  Because of this increased delay, any echo is more noticeable because there is a greater time between when you speak and when you hear your voice in your own ear.  Most phone systems address echo through the use of echo cancelers.  ESI uses sophisticated hardware-based echo cancelers in all its phone system products.

Jitter, latency, and packet loss can usually be addressed by obtaining a managed network switch and/or a good router (such as Cisco).  This equipment offers something called QoS, or Quality of Service, which essentially looks at network traffic and determines which packets are voice traffic, and gives voice the highest priority.  Even when using a Remote IP phone or ESI-Link (ESI’s site-to-site IP connectivity technology) over the public Internet, we find that, most often, it is the internal network where most issues lie if there is trouble.  Having indicated that, voice quality is obviously dependent on the quality of the Internet connection as well.  We have found in general that if using the public Internet to carry your IP traffic, T1 is generally the best way to go, followed by cable modem, followed by DSL.  Satellite modems are NOT acceptable for VoIP traffic, nor are other forms of wireless Internet (such as microwave-based technologies) since they tend to be “bursty”.  Of course, while it is a more expensive option, a point-to-point T1 connection would be the best guarantee of quality, and this is often the best choice for full site-to-site connectivity.  Another good choice would be an MPLS circuit, which has QoS capabilities.

When determining in what fashion it makes sense to deploy IP, one should consider the extra time and effort it will take to ensure that things run smoothly, and compare it with the benefits you will receive.  With local phones in the office, this can be a toss-up.  This is because, all else being equal, IP phones generally cost more than their digital phone counterparts (although with ESI Communications Server phone systems, IP pricing is very competitive with its digital counterpart if you’re buying 20 phones or more).  Also, as mentioned above, the IP phones’ voice quality and connectivity can be subject to network conditions.  Keep in mind that if your network goes down for whatever reason, it will bring down the phones as well, which will certainly add to the stress of your IT person or team.  If you have digital phones, on the other hand, they will continue to operate normally in the case of a LAN outage.

Many people tout using IP phones on their LAN because of the convenience of moving phones around without having to pay for a phone technician to visit your site.  With ESI’s Communications Servers, this is now a moot point, as even digital phones can be swapped easily without a site visit or wiring change.  Still, many IT professionals prefer working with IP phones, because they understand them and how they operate, and local IP phones are certainly a viable option.

Where IP applications tend to make the most sense are for Remote IP phones (for telecommuters or executives), IP “soft” phones (for “road warriors”), and multiple-site connectivity where each site has its own phone system.  If you could benefit from any of these scenarios, it’s well worth the effort to take the plunge.

Voice over IP, or VoIP, has come a long way since the early days.  Back then, VoIP phones in general weren’t at all tolerant of network issues, nor were networks as good at prioritizing VoIP traffic to ensure quality.  QoS, or Quality of Service, a term used to indicate network traffic prioritization, was nonexistent on service provider networks.  And most IP phones out there were difficult to set up, could not be easily moved from site to site, and had very few features.

What a difference a product generation or two makes.  Not only are IP phone makers getting smarter and more capable in their offerings, but service providers are offering more bandwidth and many are routing VoIP traffic more intelligently.  Sure, things still aren’t perfect, but for the first time, business VoIP phones are truly useful.

ESI (the manufacturer whose phone systems we sell) was an early entrant in the VoIP market (they were the first, in fact, to be able to network 100 phone systems together via IP), and their experience has shown.  Unlike others in the market, they have for a long time offered the same feature set on their IP phones as their regular digital key phones (and believe me, they have a rich feature set).  On ESI’s Remote Feature Phone II, which is compatible with any IP-enabled Communications Server phone system, you’ll truly be able to participate as if you were sitting in the office.

ESI IP Feature Phone II (backlit)

ESI’s Remote Feature Phone II, while retaining the same form factor as its predecessor (which we like), has improved some things under the hood.  For example, it has become much more tolerant of network issues.  Not that there was anything wrong with the prior iteration, but the newer version is a bit like a car having better shocks on a bumpy road.  Of course, you’ll want to make sure that your network connection is as good as it can be, but much of the time we don’t need to do any special router configuration at the remote site for the user to have a good experience.

One thing that the Remote Feature Phone II also integrated was the ability to “plug & play” at virtually any site where there is a broadband connection.  In the prior generation, you would set up the phone and not move it, because it would generally take about an hour to get the phone operating wherever it was.  Not here–wherever you plug the Feature Phone II in, it begins communicating with the main phone cabinet right away, and connects automatically.

Another change ESI made was switching its codec to G.726 (a codec is the compression algorithm used to save Internet bandwidth).  I think this was a brilliant move; most other IP phones seem to be using G.729 or G.729a, which has a higher compression rate, and the higher voice quality of G.726 is noticeable.  While G.726 does use more bandwidth than G.729, it’s still much less than G.711, which is uncompressed.  Besides, provider bandwidth is increasing more all the time, and the 80K or so the phone requires (which includes voice in both directions, plus overhead for lighting buttons, updating the phone display, etc.) is very, very small compared to what standard cable modem connections provide around here (generally about 2Mbps upstream and 10Mbps or so downstream).

One thing I appreciate about ESI’s IP phones is that they don’t skimp on buttons.  I can’t believe that most manufacturers are getting away with putting only a few buttons on their IP phones (presumably to save on manufacturing costs?).  Sure, many of them try to compensate by having a larger screen and/or navigation buttons, but I have always found navigating this way to be cumbersome.  There is truly no substitute for having all the buttons you need for line appearances, station buttons, and feature keys.  I also note that ESI’s phones have tri-colored buttons; most competitors’ lighted buttons are either a single color, or in some cases, two colors.

Regarding features, I haven’t really delved into that in this post.  Features and ease-of-use are ESI’s strong suit, and their phones are truly a pleasure to use.  You can check out our web site to find out more on ESI products in general, or poke around in the other posts on this blog.

A Hosted PBX (sometimes referred to as Hosted IP or IP Centrex) is a relatively new phenomenon in the telecommunications world, springing up as a result of a promising new technology, Voice Over IP (VoIP).

A Hosted PBX differs from a traditional PBX or key phone system in that businesses using a hosted PBX are “outsourcing” their phone system to a service provider. This appears on the surface to provide a number of advantages:

• Your phone can follow you wherever you go
• You don’t have to worry about equipment maintenance
• Your initial out-of-pocket costs are lower

There are certain things, however, one should know in order to make a well-informed decision when selecting a solution for critical telephone communications.

Reliability.

Using a hosted PBX means that all of your telephone calls will travel over an IP network, often the public Internet-even your intercom calls to other phones in the same office! This is the same Internet that tends to bog down during certain periods of the day. When you browse the Internet or check your email, many of the common day-to-day interruptions go unnoticed-if your connection is suffering for a few seconds or fractions of a second, you won’t notice a problem. VoIP, on the other hand, is very unforgiving about data transmission delays, and you will experience a loss of quality, or even dropped calls, on a regular basis.  Some providers mitigate this by providing a point-to-point connection between your office and their facility.

Cost.

At first glance, it may seem that you will save money by going to a hosted PBX solution rather than purchasing your own equipment. But will you really save money? The going rate per seat, per month in a hosted PBX solution is approximately $50. Multiply this by the number of users over 5 years (the average time period a business will use their phone system), and you may be surprised at the results. Let’s take an example of 15 employees:

$50 per month x 15 employees x 5 years = $45,000!

If your goal is to avoid large upfront costs and have a reasonable, stable monthly cost, you would financially be far better off doing a lease purchase of an in-house phone system.

Features.

When you sign up for a hosted PBX solution, you should understand that your service provider will not be dedicating a server to your business; on the contrary, you will be sharing a remote server with hundreds or even thousands of other businesses. This is important because many of the advanced features offered by standalone phone systems require significant computing power and resources. Because of this, you will notice that if you compare the feature set of a hosted PBX provider against that of a standalone phone system, many features and capabilities will be missing from the hosted PBX solution or be at an additional monthly cost, such as conferencing, call recording, automatic call distribution, presence management, unified messaging, service observe, enhanced Caller ID functions (such as attaching the Caller ID of a caller to voicemail messages) and more. Additionally, because most hosted PBX providers utilize non-proprietary IP phones, they must offer a “lowest common denominator” feature-set. The number of buttons available for lines, extensions, and features is usually lacking, as is the number of button colors available (ESI, our manufacturer of choice, offers tri-colored buttons; most IP phones used in hosted IP solutions have either dual-colored buttons, or in most cases, only one color).

Expandability.

Because each phone in a hosted IP PBX requires a certain amount of bandwidth, you will ultimately be limited on the total number of phones that you can add to your setup. The more phones you have and are in use, the greater the bandwidth requirements. Additionally, the phones will have to compete with the computers on your network for bandwidth-when many phones are in use, you may notice that the voice quality suffers considerably. In many cases, you may purchase equipment that supports Quality of Service, or QoS to alleviate this and give priority to the phone traffic. The downsides to this are that good hardware to implement this properly can be expensive, and in addition, ensuring that the phones have the bandwidth they need will cause your computers to run slower on the Internet.

Security.

In a hosted PBX, voice traffic travels off of your network onto other networks-your service provider’s network at the least, and in most cases over the public Internet. There, your conversations can be vulnerable to interception and eavesdropping. If your hosted IP PBX provider implements encryption on all calls, your risk is somewhat mitigated in this area.

911 Service.

Something most VoIP providers don’t like to talk about is 911 service. The reason for this is that some of them still don’t have it. Fortunately, since 911 compliance has been made a federal mandate, the VoIP service providers are making rapid progress toward this goal and many of them have implemented it in some fashion. Still, you should be aware that even for providers who have implemented a solution, there will be at least one extra layer of separation between yourself and emergency services. This is because the provider has to either hire staff to handle emergency calls or contract it out to another call center, but in either case the party you reach when you dial 911 is not a governmental 911 dispatch center.

Provider Longevity.

VoIP services is a new and expanding area of commerce, with most companies offering services being fairly new business entities (or at least being relatively inexperienced at offering these particular services). How long has the company you’re considering been in business, and perhaps more importantly, what will be their condition a year or two from now? If they do go out of business, will your service remain intact? You can always port your phone numbers to another provider, but if the hosting company is unresponsive you could experience delays (perhaps on the order of weeks) in that respect.

Monetary Benefits.

Keep in mind that there are certain tax benefits associated with having your own in-house phone system equipment.  When you own your own phone system, it is also considered an asset, improving your company’s financial worth.

Final Thoughts.

Hosted IP providers often tout the portability of their phones, so that your phone can go with you wherever you go, as long as there is a broadband Internet connection.  But if this used to be an advantage for them, it is no longer (ESI has, for example, three offerings that will go anywhere you do: a desktop IP phone, a cordless IP phone, and a PC-based “soft” phone).

Hosted providers often also tout low long-distance rates in their offerings.  But it should be noted that this is more a function of your dialtone/long distance service than it is the choice of an in-house versus hosted PBX.  There are several providers here in the Phoenix area (and probably nationwide) that are offering free and low-cost long distance in their plans.

When it comes to critical applications like telephone communications, having an in-house PBX (phone system) is the clear winner.

Almost every phone system, universally, supports cordless phones when properly equipped with an analog station port.  These are the phones you can get practically anywhere and are primarily intended for home use.  Many phone systems, however, do not support cordless phones via a digital interface.  Why does this matter?  Features, and ease of use.

An analog interface to a phone system is a very simple one.  Cordless phones connected this way are generally able to make and receive calls, transfer, and that’s about the extent of it.  Sure, a lot of manufacturers allow you to do more by pressing the Flash button and dialing a code, but few people ever venture beyond the basic functionality because to try to do more is, well, a hassle.  Even putting a call on hold is usually a chore, because you can’t see by looking at the phone what line a call is on, to be able to pick it back up.

This is where digitally-interfaced cordless phones come in.  These phones can communicate digitally with the phone cabinet, allowing for much tighter integration, making for more features and a phone that’s much easier to use.  What features?  Well, let’s take an ESI Digital Cordless Handset, for example:

This baby is small and light, making it very portable.  Though you can’t quite make it out from the picture, this phone has dedicated Hold, Redial, and Voice Mail buttons, as well as a button used for Transfer, Conference, or station programming depending on when you press it.  What’s really great are the 4 buttons along the bottom that can be used for almost anything, according to your needs.  Here is a partial list:

• Line buttons
• Extension buttons (including extension status)
• Company day/night/holiday mode
• Do Not Disturb
• Call Forward
• Overhead Page
• ACD (Automatic Call Distribution) agent logon/logoff
• ACD administrator for viewing ACD queue status
• Personal greeting switch
• Virtual Answer
• Door Unlock (used with Presence Management)
• Account code entry

The display indicates when you have new voicemail messages.  Also on the phone is one-button access to change the volume (including ring volume), as well as mute.  There is a micro-mini headset port at the top, which supports generally the same headsets you would use for a cell phone.

In addition to the digital version of this phone, a local IP version is available (for inside the building), and a remote IP version is available for off-site.  All three versions operate in the same way, with full functionality.  The remote IP version will also allow you to connect a phone line (for example, if you’re using it at home), so that you have the choice of making local calls if you wish (otherwise, calls are routed via your office lines).  There is also a slightly larger version of this phone (in all 3 flavors) which offers longer battery life and an extended range.

There has been a real buzz over Voice over IP (VoIP) during the last couple of years.  We get asked about it often by businesses looking to purchase a new phone system, who want to make sure they’re not being “left out”.  How does it work?  Will it save money on long distance?  What other benefits does it provide?  In many people’s minds, the concept is a bit undefined; they’re not sure exactly what it will do for them, but they do know that their next system should support it.

First, a small explanation of what VoIP technology is, and the basic reason it’s so promising.  In traditional telephony, voice conversations take place over a dedicated line.  This is generally a pair of wires that carry your voice to the other person you are speaking to, and their voice back to you.  Granted, there is sophisticated equipment that routes your call from point A to point B, so there are usually a number of sets of wires that are actually being connected, but the point is that you are using a dedicated connection when you are on that call.  Since there are a limited total number of connections that can be made, even the phone company can run out of connections, and this is why you can sometimes try to make a call and hear the “all circuits are busy” recording.

VoIP technology is more sophisticated.  When you are on a conversation using VoIP technology, your voice is broken out into many small bits at lightning speed and sent to the other end very quickly, where it is reassembled and played to the person at the other end.  These small voice bits, or packets, have several pathways they can take to their destination and will take whatever path is the most efficient at that millisecond in time.  As long as they can be reassembled properly at the other end within a reasonable time frame (on the order of milliseconds), it does not matter what path they take to get there.  Because a dedicated path does not need to be established in this scenario, certain things can be done to make calls more efficient, such as cutting out the parts of the conversation where no talking is happening, like between words, and voice compression can also take place.  The net effect is that the same resources that could once carry say, 24 calls simultaneously, may now be able to carry triple that number or more.

It’s important to note that, while all VoIP technology is the same in a basic sense as has been described, there are some different applications  that take advantage of this technology.

VoIP Dialtone

Dialtone providers have learned early on the tremendous efficiency gains (and cost reductions) that could be had with packet-based calls, and most carriers have upgraded their equipment so that, at least behind the scenes, they are using this technology.  Most are also now offering VoIP-based phone lines to both business and consumer customers.  In some cases, the interface to your home or business may not even be any different, but they can offer more competitive dialtone rates to you because of this technology.

Some companies, such as Vonage, deliver dialtone over the Internet, and they provide you with a special box called an ATA (or Analog Terminal Adapter) that converts the IP technology into an analog signal that will work with your phone or business phone system.  Companies that provide dialtone via the Internet in this manner are called ITSPs, or Internet Telephony Service Providers, and they can usually offer very aggressive rates in the form of lower monthly rates or free long distance.  Some newer phone systems have circuitry that can understand IP dialtone natively, and in such a case an ATA box is not required.

Phone System VoIP

Phone system manufacturers have also realized that they can do some fantastic things with VoIP technology.  These applications offer tremendous benefits by taking advantage of two things: (1) the fact that most Internet connections are based on a monthly fee, and that you are not charged based on the amount of traffic you generate, and (2) the global nature of the Internet.

• Local IP Phones: these are used instead of traditional digital phones, and rather than requiring separate cabling can use the same cable that your computer uses on your office LAN (Local Area Network).  Their functionality is usually the same as a normal digital phone by the same manufacturer.
• Remote IP Phones: these can be used off-site, such as at a home residence for telecommuters or executives.  While remote IP phones from most manufacturers offer a more limited feature set than a local IP phone, remote phones from ESI (the manufacturer whose phone systems we sell) offer exactly the same experience that a local IP phone does, meaning that a remote user can see who is on their phone in the office at a glance, intercom, answer incoming calls to the business, act as a customer service agent as part of an ACD (Automatic Call Distribution) group, and more.
• Remote “Soft” Phones: this is software installed on a computer, such as a traveling sales rep’s laptop, that, when used in conjunction with a headset, allows the user to have similar functionality to a Remote phone, without the phone’s footprint.  This is extremely beneficial for “road warriors” and can be used anywhere in the world there is a decent broadband Internet connection (wired or wireless).
• Site-to-site VoIP: this is beneficial for businesses with multiple locations.  These locations can be virtually connected together to make it as if the users in all offices are together in one large office.  In addition to bringing everyone together without having to dial outside phone numbers from one site to another, this can eliminate long distance charges between offices.  ESI’s implementation of this is called ESI-Link, and supports tying up to 100 phone systems together.

ESI offers all of the above-mentioned types of phone system-based VoIP in their Communications Server phone systems, and does so with a very rich feature-set and high-quality voice codecs, making it an outstanding investment value.