Improve company productivity with a Business Account.Sign Up

  • Status: Solved
  • Priority: Medium
  • Security: Public
  • Views: 1615
  • Last Modified:

What is the current status of VoIP in companies

Not that I wanted to be the first to ask a question here :)

VoIP was discussed a lot in big companies
- much cheaper (through Internet - especially since bandwidth is huge everywhere)
- does not usually a PABX (usually)
- safe encrypting solutions are available
- VoIP can become Voice and Image over IP pretty easily, making conferencing systems much cheaper, with a wider usage in the company

What is the current status - do we have statistics about companies having done a (un)successful migration?
Do we know the level of satisfation?
What is the short/middle-term perspective?
  • 12
  • 7
  • 5
  • +4
5 Solutions
It is still the driving force and the marketing point for new networking products especially those which target the enterprise market

If bandwidths are good, VoIP should not be much of a problem ... with faster alternatives available in the form of VoFR etc. it is bound to further pick up (voice that is) ... Though I do not have any stats on (un)successful migration, but I do believe that most upcoming networks like to keep this option open at the very least ...
errr... my source of info is my idea of the R&D done by my company and the design goals and feature sets (I am into router development)
MercantilumAuthor Commented:
Yes, this is my point, I'd like to know how companies could take benefit from low bandwidth, VPN etc...
A couple of years ago, many companies were still afraid to put something on the Internet...
Improve Your Query Performance Tuning

In this FREE six-day email course, you'll learn from Janis Griffin, Database Performance Evangelist. She'll teach 12 steps that you can use to optimize your queries as much as possible and see measurable results in your work. Get started today!

At lower bandwidths voice quality is bound to suffer, esp for VoIP ... VoFR and brethen get better performances if QoS has been properly set up (solely due to possibly smaller payload size)

So what is it that you are trying to ask ? The real stats of how many companies use it and at what bandwidths or what technologies are used for making most of whatever low bandwidth is available ?
MercantilumAuthor Commented:
Regarding VoFR I've no doubt quality is (usually) better.
But frame relay has to cope with the fact that usually only one provider takes care...
While Internet can have many ways to be routed when a node dies...
Anyway FR is  *much*  more expensive.

Internet side, the VPN get some confidence, whiile some people are still afraid of the "crackability" of the keys (1024 bits is not enough).

So, the question is a bit out of the teckky beaten tracks :)

I'd like to hear from professionals about their VoIP experience (especially through the Internet):
- considering the high cost of a PABX, maintenance, configuration etc...
- what is the trend - is the level of confidence higher than 2-5 years ago in using VoIP (Internet)
- some statistics to show the trend and this level of confidence
- perspectives for the coming years

Well, this thread being quiet, it may take some time ...
I think we should wait for someone more suitable to reply ... I do not have any experience from the user's point of view so can't give first hand information ... But I would like to add a point ...

cost of setting up a FR network is lower than that of setting up a VPN ... It is the recurring costs which are lower incase of VPN and it takes a few months of operation to cover the costs and gain benefit over FR  ... this I knew for sure, so I shared ... Rest I would leave a real expert to comment ;o)
MercantilumAuthor Commented:
Hmmm, international FR lines ... will takes less than 3 months to ammortize the inital VPN to my mind.
Btw, Cisco ?
>Hmmm, international FR lines ... will takes less than 3 months to ammortize the inital VPN to my mind
Actually it was lot more than that (more than a year ... I do not exactly remember) unless you opted for some cheaper alternatives

>Btw, Cisco ?
Nope .. we're brand new in this field ... can't post the co name in open forum
Well, seeing as how VoIP only requires 14k per channel, it isn't very hard to inplement. I have installed a few flavors of Voip, Multitech and Avaya products, and found them to work very well indeed.

I have also installed Cisco's, but they have some issues with there POE to the phones. If there are proper QoS, and P/q switches, the implementation is quite easy. Toll bypass works very nicely, and The customers I have implemented it for are quite happy indeed with the product.

The intigration of Data and voice brings so many neat tools to the table as well, with exchange, etc. intigration. I have 4 new jobs upcomming, and the more I do the more I like.

Technology is a very, very good thing.

Have fun all..

Don Johnson
Mm, just noticed this topic.

Could someone please explain to me exactly WHAT VoIP is? I don't quite get it, and any
searches I do on it just talk about companies GOING to do something about this think called
'Voice over IP' without actually explaining what it is. All I seem to find is "Company XYZ will be releasing something soon"...
Here is a presentation I put together for one of my clients.

An overview of the “Voice over IP” or VoIP technology, benefits and
associated protocols, from an informed position.

The public telephone network and the equipment that makes it possible are taken for granted in most parts of the world. availability of a telephone and access to a lowcost, high-quality worldwide network is considered to be essential in modern society,
with expectations for reliability of the telephone network being set in the highest order. Such has been the reliability of traditional voice networks, that it soon became accepted to transport data over voice circuits, (e.g. facsimile & modems).

The capability for encapsulating voice into packetised data streams has been around for many years, but reliability of historical data networks, both in terms of uptime and delivering latency sensitive applications such as voice, has held back widespread adoption. The main drivers behind the inevitable move toward transporting voice,
(plus other multimedia applications such as video), in packetised format over data networks, is the Internet explosion and IP (Internet Protocol) becoming the dominant network layer protocol.

Support for VoIP, has become especially attractive given the low-cost, flat-rate pricing of the public Internet, in fact toll quality telephony over IP, has become one of the key steps leading to the convergence of the voice, video, and data communications
industries. The feasibility of carrying voice and call signaling messages over the  Internet is readily demonstrable, but delivering high quality commercial products, establishing public services, and convincing users to buy into the vision are just beginning.

Whilst we are a long way from complete adoption of VoIP, due to major existing investment in legacy voice networks, peoples reluctance to deploy immature technology and a current lack of large integrated voice & data rollouts in critical real world environments, most of the traditional voice vendors are investing more R&D costs into what will surely become a pervasive technology over the next few years. Applications & Benefits of VoIP
Some of the recognised benefits of VoIP include:

Cost reductions

This is still a subject for much conjecture and debate within the industry, although with the lowering costs of communication across the Internet, increasing popularity of secure VPN’s and indeed, an ever-growing number of corporate IP based WAN’s, there should certainly be many cost based arguments to justify a VoIP deployment.

Downsizing Since people are among the most significant cost elements in a network, any opportunity to combine operations, to eliminate points of failure, and to consolidate accounting systems would be beneficial. In the enterprise, SNMP-based management can be provided for both voice and data services using VoIP.

Related facilities such as directory services and security services may be more easily shared. Network Consolidation An integrated infrastructure that supports all forms of communication allows more standardization and reduces the total equipment costs. This combined infrastructure can support dynamic bandwidth optimization and a fault tolerant design.

The differences between the traffic patterns of voice and data offer further opportunities for significant efficiency. Universal use of the IP protocols for all applications holds out the promise of both reduced complexity and more flexibility. Advanced Applications
Even though basic telephony and facsimile are the initial  applications for VoIP, the longer term benefits are expected to be derived from multimedia and multiservice applications.

For example, Internet commerce solutions can combine WWW access to information with a voice call button that allows immediate access to a call center agent from the PC. Combining voice and data features into new applications will provide the greatest returns over the longer term. Some of the applications include:

Internet-aware telephones: Ordinary telephones (wired or wireless) can be enhanced to serve as an Internet access device as well as providing normal telephony.

Directory services, for example, could be accessed over the Internet by submitting a name and receiving a voice (or text) reply. Voice calls from a mobile PC via the Internet: Calls to the office can be achieved using a multimedia PC that is connected via the Internet. One example would be using the Internet to call from a hotel instead of using expensive hotel telephones. This could be ideal for submitting or retrieving voice messages.

Internet call centre access: Access to call center facilities via the Internet is emerging as a valuable adjunct to electronic commerce applications. Internet call centre access would enable a customer who has questions about a product being offered over the Internet to access customer service agents online. This could also be extended to remote branches or home workers, as an extension to the call center facility offered.

A lot of unnecessary grief can be avoided, with a little fore-thought to designing the correct topology and implementing a resilient and bandwidth scalable infrastructure. Whilst it is not always economically viable to increase bandwidth; in a LAN environment where it is more easily controlled, deploying an appropriate level of bandwidth and even over provisioning, can save considerable time and the complication of configuring and administering QoS mechanisms.

As part of the move from data only, to an infrastructure that now supports all communication forms, particular attention should be given to resilience, with rapid recovery in the event of failure. With features available today, such as Link Aggregation, VRRP, 802.1w (Rapid Spanning Tree), it is possible to create a very resilient and fault tolerant environment, even in the least complicated networks. QoS mechanisms are not mandatory when designing a network to support multimedia applications and, as discussed above, can be a burden and an overhead best avoided.

It is probably best to consider whether any major benefit can be gained and if possible, start with no QoS configured and monitor performance and response over a period, particularly covering times of high usage, with a view to adding QoS where appropriate. Two reasonable arguments for deploying QoS are:

1. When bandwidth is limited and/or traffic flows are of a high enough level, as not to guarantee timely delivery of latency sensitive traffic such as voice. In this case, inherent queuing mechanisms within switches and routers can be effectively utilized, by applying a prioritization policy, that corresponds to the priority of a specific traffic flow, thereby ensuring that the latency sensitive
flows get processed first.

2. In an environment where varying types of multimedia traffic, (e.g. voice, nonreal time video), are mixed with data, that also may have differing levels of importance.

In addition to prioritization techniques, rate limiting/bandwidth throttling methods may be deployed. These will typically restrict bandwidth for the less critical traffic, on a per application basis, offering more resource to the more important traffic flows. In a contention based environment such as Ethernet, this equates more to time slices than raw bandwidth allocation, but amounts to the same result in practice.

Latency is the measure of time it takes to send data from one point to another. The amount of latency will affect the efficiency of applications running across the network. This will have more of an impact on time sensitive applications like Voice over IP, which requires minimal latency between the end points to meet users
expectations. The following one-way transmission time guidelines have been defined by the ITU Recommendation G.114

Under 150 ms (with echo cancellation): acceptable for most
user applications

150-400 ms: acceptable provided that management is aware of the transmission time impact on the transmission quality of user

Above 400 ms: unacceptable for most applications
Due to the fact that Latency can impede the performance delivered by VoIP deployments, it is important that the end-to-end latency be kept at a minimum.

Although some latency is taken into account with VoIP protocols and equipment, it is critical that the amount of end-to-end latency not exceed the levels described in the definition. Once latency exceeds the acceptable level, a VoIP conversation can become unintelligible causing the deployment to be considered a failure.

Latency is caused by many parts of the system including the encoding algorithm used (G.711, G.729, etc.) and the network infrastructure. The encoding algorithm introduces minimal latency, which can be assisted by buffering, the majority of issues can come from the network infrastructure if it is not able to cope with bursty LAN traffic and VoIP traffic simultaneously.

Jitter In data networks, jitter refers to packet jitter, not bit jitter. When Data packets arrive at an intended destination, the time measured between arrival consecutive packets (delta time) in the stream must be a constant interval. If the delta times between
packets become varied, it is deemed as Jitter.

The amount of Jitter that can be tolerated before a call is considered to have bad quality is dependant on the 'Jitter Buffer' associated with receiving device. The purpose of the Jitter Buffer is to buffer the arriving VoIP packets, and 'smooth' out the latency between the arriving packets before delivering them to the end user.

Although Jitter Buffers can assist in delivering a good quality call, they can only offer little assistance. The issue becomes the amount of buffer used to 'smooth' out the call before delivering the packets. If the Jitter Buffer is too large, there can be minimal jitter experienced, but the end user will experience delays in the conversation that may not be tolerable.

Therefore, although Jitter Buffers help when jitter is experienced, it does not provide complete protection against quality issues relating from Jitter. There are several protocols associated with deploying a packet based multimedia environment such as VoIP, each with it’s own merits that should be considered according to individual requirements.

The first consideration is preference for a centralized or distributed architecture. The former is similar to the traditional PSTN, where all the ‘smarts’ are handled by central switches, (in VoIP nvironments these could be Call Agents or Media Gateway Controllers), and the end devices are relatively simple with limited features.

A distributed architecture relies on embedded intelligence in each end device, (which could be IP phones, VoIP gateways or any device that can initiate or terminate a VoIP call). The intelligence can be all aspects of the call handling, from call features and
routing through to billing information. Centralized MGCP (Media Gateway Control Protocol) as devised by the IETF and H.248 as
devised by the ITU, were designed with a centralized architecture in mind and lend themselves ideally to this deployment method.

Although they can also be used in a centralized environment, with the aid of user agents or gatekeeper call signaling mechanisms, the two recognized protocols for a distributed architecture are H323 (devised by the ITU) and SIP (Session Initiation Protocol), an IETF standard. H.323 is based on the Integrated Services Digital Network (ISDN) Q.931 protocol, which allows it to easily interoperate with legacy voice networks such as the PSTN or
Signaling System 7 (SS7).

It was one of the earliest multimedia standards designed for LAN’s, catering for all aspects of the VoIP call setup, tear down and routing and is therefore the most widely deployed. The call-control devices in an H.323 network are called gatekeepers, which act as a single point for co-ordinating routing, resource allocation, billing etc…

SIP, having been devised by the IETF, utilizes Internet protocols such as HTTP for message formatting and DNS/URL for naming, which makes it a popular choice going forward. Although similar in principle, SIP uses proxy servers rather than gatekeepers for call handling. It is feasible for these protocols to run concurrently on the same network, but in the interest of simplicity and performance, it would be more practical to implement a single protocol strategy.

Real-time Transport Protocol sits between the Transport Layer and Application Layer and is used in conjunction with RTCP (the control element). As an end to end protocol, it does not guarantee real time delivery, but is still suited for multimedia applications as it provides elements such as timestamp and control mechanisms for synchronizing multimedia streams with timing properties.

If you would like more information You can e-mail me....

MercantilumAuthor Commented:
Thanks nosnhojnod, the PDF is maybe more readable 
It answers cool12399's question.
Just a small comment from a SOHO user,
I switched to VoIP about a year ago. In Europe. The service providers for my (SOHO) type of business really have to get their acts together.
Standard SIP services are still being developed, for example:- Music on hold, conference calling, call forwarding, geographical diverse secretary manager, voice mail etc.
As far as I have found, no one in Europe offers a pachage that can really be defined as embracing the true spirit and impressive possibilities of SIP .
I find it a great shame and I have spent many frustrating hours with several VoIP ervice providers trying to:
a) make / receive calls with reasonable quality
b) implement the above features.

If any SOHO (or medium size) in Europe is considering (and you should for cost alone) then you are welcome to email me ( and I will share my pain with you can avoid repeating my teething problems.  
I am not associated in any way with the VoIP business.
There you are, a user opinion. Superb possibilities but as you no 100% reality. VoIP providers convince me I am wrong.
MercantilumAuthor Commented:
Thanks Muskett, I guess your VoIP was over Internet.
Yes, VoIP on a 300/300 connection using cable. I have now got to the stage where I can make and receive calls reliably. All the other features mentioned above are still "under development"

I am sure that (I hope) I will be deluged with emails from VoIP providers tolling me they can provide these services.
Hi Mercantilum ;)

> What is the current status - do we have statistics about companies having done a (un)successful migration?

Well my company (which unfortunately I can't mention) rolled out VoIP about 2 years ago. I feel on the whole that this was a successful migration and very worthwhile. That is, after the teething problems were ironed out. By teething problems, I mean the full range of issues, such as phones re-registering all the time, problems getting IP addresses/gateways, problems getting the right number, etc etc. The system we have is Cisco phones, servers and switches. Most of these problems were fixing eventually, with patches/updates from Cisco I believe.

> Do we know the level of satisfation?

Good, now. Now the system runs itself well, that is except for the time when a virus found it's way onto one of the VoIP boxes and almost killed the system... but thats was only cause Cisco originally wouldn't let us put anti-virus on them, which has now been solved thankfully!

We have 2 sites, connected by 2x100MBit links. These links carry the network, internet, internal email, VoIP traffic over them quite well. Sometimes, there has been quality issues when talking to someone at the other site, but I believe that was when we had 2x8MBit links, since the 2x100MBit links have been put in then all has been good.

> What is the short/middle-term perspective?

Not sure what you mean!?!
MercantilumAuthor Commented:
Hi simonprr (long time no see :)
Thank you for the interesting comment.

Can I ask the number of persons in your company (to get an idea of the size)?

"What is the short/middle-term perspective?"
This question is relative to the development of VoIP in companies: what is the trend
- most companies still wait?
- some companies plan to migrate to VoIP?
- many companies started to migrate or plan to?
Hi :)
Yeah long time... hehe :)

We rolled VoIP phones to approximately 2,000 users. Roughly 10 servers across both sites, some of which are purely for the VoIP system, 1 for the recording of calls and some for running the helpdesk phone system.

I think personally, that most companies wait because they are scared of it or cause they don't have the necessary infrastructure to support it. Or in probably quite a lot of cases, are happy with the solution they have at present. The main reason my company migrated was because they worked out they would save a few million in the next 5 years (or something like that), instead of continuing the contract they had with BT. Even though they had to pay me and quite a lot of other people overtime for many many weekends, fixing problems and plugging in phones all over the place! So to answer your question directly I believe that "most companies still wait". However, said that I believe there may be a split:

- most companies still wait? --> 60%
- some companies plan to migrate to VoIP? --> 35%
- many companies started to migrate or plan to? --> 5%

This is a pure guess really, but I reckon it's a good indication of the situation. I have in the past seen similar stats in computing magazines, etc.

Actually, thinking about this, I do know quite a lot about VoIP. Maybe I should hang around the section more often :)

PS. When you get a chance please check out my latest question, maybe you can help ;)
MercantilumAuthor Commented:
Thanks again.
Seemingly companies are still afraid of doing this (probably) cost-effective migration.

By the way, could you get a ROI (return on investment) of your migration?
What kind of savings to you expect do get in, let say, 2005?

NB: had a look to your questions, of course Perl would be (for html and ftp) a good alternative.
Why don't you drop a 20 pts link from Perl TA to your DOS questions?
Yes, I believe they are afraid of doing it. Actually, I only believe it's cost effective with medium to large companies. Smaller companies and some medium companies may not benefit, due the infrastructure, support, etc that is required when you have this type of system.

Yes, I mentioned that they worked out it would say my company a few million over 5 years... unfortunately I don't know any more detail than that.
Except for the obvious, which is that previously we all had normal phones on our desks, which connected to exchanges inside the building. Then each internal call that was made was chargable, including the calls to other other site (I guess that is external, but still). Daily, that would have been quite a large £££, especially as most of the calls were probably to our other site anyway. With VoIP, it just uses the network, both internally and externally to the other site (which we have and need anyway) so that daily cost is no more.

Hope that makes sense, sorry I don't have any more info on costs/savings, but that it's my area.

So whats your reason for this question? Thinking about going VoIP??? If you are... go for it! But defnitely allow for the "teething problems"!
Although it has moved on a LOT since 2 years ago.

> Why don't you drop a 20 pts link from Perl TA to your DOS questions?

For a number of reasons:
- I already have a very large amount of asked questions, doing this would increase this even more
- More questions to manage, some people post in both questions, then there is posts to read and respond to all over the place, makes things very disjointed (and a big mess!)
- People tend to ignore low points questions, so that would mean posting large points questions to get attention
- Would be fine to do it in Perl of course, if it gets the job done, but trouble is I don't know Perl, so when I will likely need to amend it/add to it/etc I won't be able to without help. In normal batch file language, I should be able to work it out, logically. Perl looks extremely complicated.

MercantilumAuthor Commented:
Just curious :)

About the Perl, Simon, do me a favour:
1. ask a first Q in perl TA: "What's the best way to get the basics of Perl?", and learn basics (not that difficult, I could do it :)
2. next requirements of scripts only in perl TA
3. forget this msdos forever

You'll thank me one day :)
Yes, good idea. I will do that one day. In the meantime, feel free to code for that question. That way, I will learn a bit about it as I go along and also we can discuss what I would need to amend to cater for more DJs that I want to search for on those pages. Cheers!

Feel free to ask any further questions about VoIP in companies...  this question does seem almost endless.

I forgot to mention that we have servers for a product called Unity, which is the voice mail system. I really like it actually, it interfaces with Outlook, so when you get a voice mail, the light on the phone turns on (to be able to listen to it that way) and also you get an email with a wav file attached, which is the voice mail message. Very neat. And very useful for when I'm remotely accessing my email! :)
MercantilumAuthor Commented:
Yes, sounds cool :)

Perl, better start today than never ;-)
I posted this to another question but I think it is relevant here to the extent that there are still many misconceptions about VoIP not the least of which is that most clients and many Professionals still think that you can eliminate all need for traditional phone lines.

As far as I know contrary to popular belief you will not do away with "ALL" of your copper line phone connections at best you will just defer or shift the cost to another provider.  The only way I know of completely going to VoIP is if everyone you do business and or plan to talk to is using VoIP.

At some point the call has to get back onto a traditional copperline and complete the call to those phones / businesses /clients that do not / will not use VoIP.  A prime example is emergency services.

I know some people will say what about providers like NuVOx Avaya etc.  Every "Total VoIP" solution that I have seen has to have provisions to interface with the POTS system or there will be some calls that you cannot make or recieve. Unless of course you shift the cost of that POTS line by paying a provider (NuVox Avaya etc.) to provide that interface.

Can this save costs yes it certainly can.  Can it eliminate some of your phone lines? In most cases yes.  Does it eliminate all the traditional phone lines ?  Thats questionable. At best you reduce the number of traditional lines/connections needed to complete non VoIP calls and shift the location of those lines to a VoIP provider which you pay to complete all the non VoIP calls.   Obviously you sould look very closely at the costs to see if this an answer for you.

From what I have seen the best application of VoIP application right now is to consolidate branch offices over your existing network.   I have worked with a few different VoIP solutions some use dedicated Voip phones and some use traditional analog or digital phones and complete the VoIP connection via a gateway or PBX.

When using dedicated VoIP phone technology it can be implemented in many different ways the worst way in my opinion deploys Dedicated VoIP phones at all sites that need to communicate and you dial the designated number/addess of the phone to make a call.  If you need to talk with someone who is not on the "VoIP" network then it has to be routed through a provider (shift cost by paying a provider) or make it on a different phone using your traditional system.

When using the gateway or pbx integrated system the user dials the phone extension of a phone in the "business network" and the PBX/Gateway routes the call via VoIP to the the branch office and the call is carried completely on the data network.  This can be as seemless as calling an extension in the next room once the system is set up correctly ands can get very sophisticated to the point of programming in least cost routing tables.  

An example of a least cost routing VoIP system.  You have a business with offices in New York London and Tokyo.  All offices are connected via a data network and a VPN.  Your PBXS are Voip capable and set up for least cost routing.  I can inter office call to any extension on the system by calling the extension # and it travels the VPN/Data network as a VoIP call.  Now Say I am in New York and I need to call a supplier in Okinawa Japan.  I dial the call as usual and the least cost routing tables recognize (If you set them up correctly) that it is cheaper to route the call via VoIP to Tokyo and then make a POTS call to the supplier from there.

As you can see even in this last example I still have to pay for a phone line and long distance at the Tokyo office.  Is the cost going to be less that if I called direct from New York. I would hope so.  Have I eliminated all my traditional lines no.  Do you really want to?

One of the questions I usually ask any provider of alternate phone technology is this.  If I pick up a phone at each branch office and dial for help 911 , fire dept  ambulance etc... will it ring at the appropriate place.  This may sound like a trivial question but having worked in communications for 20 years I have held many different positions not the least of which was communications planning for a 911 center.  Believe it or not there are systems out there that dont accomplish the described task correctly.  I have seen more than one recomended VoIP application that would be dangerous to deploy as designed.

Here is an example of bad design.
I have a main office with 23 lines and 100 extensions 20 miles away I have a wearhouse with 15 workers/extensions.  The office and wearhouse are connected via a frame relay cablemodem DSL or T1 etc.  The wearhouse makes infrequent outbound calls and most of those go to the main office.  The solution most frequently recommended is, to route all calls over VoiP to take advantage of the 23 lines and lower bulk rate long diatance. Eliminate the cost of the traditional lines at the wearhouse.  In this example I have eliminated all my traditional lines at the wearhouse and I will save the cost of those lines.

This is not a good solution especially when you are stuck in the back of a burning wearhouse and you pick up the phone to dial 911 and get the dispatch in the city where the pbx is 20 miles away.   Not very useful and downright dangerous.  

Can you help eliminate this by using least cost routing tables and dialing schemes that recognize the fact that the extension is in another city and will need to dial a different number? Yes you can however,  most of the time the only way you can dial straight into a 911 center is to be dialing from a phone line in the coverage area of that particular 911 center.  All other incoming numbers will appear to the 911 diapatcher as an ouside call and therefore not an emergency.  

If the Center is large and or extremely busy those numbers may not even ring in the 911 center itself and at best will be answered by non dispatch staff and then forwarded into the 911 center at worst not be answered at all. Neither of those situations are the quickest way of getting the help you need.

When you use the traditional phone system there are taxes, regulations and laws in place to ensure that this happens correctly.  In this reguard VoiP is completly unregulated (and I would like it to stay that way)  It is up to us as providers/implementers/users to ensure these types of problems do not exist.  Unfortunately all it will take is a tragedy that could have been prevented by a properly designed/configured system and here come the regulations restrictions .....

As an independent consultant implementing network and communications solutions (Including VoIP) I am not sure where the blame would lie if some one was hurt or god forbid killed because of an improperly configured system such as I have described above.  At the very least  I would have a very hard time living with that thought.

VoIP can be a very powerful tool and as such it brings with it great responsibilities.

Just a few thoughts to consider.
LOL... is that the longest EE post in history???!

Some good points made though :)
MercantilumAuthor Commented:
Thanks Hofftek, interesting study. Still need more "success" or "failure" stories about VoIP (and stats if possible).
Mercantilum, you may find this interesting:
MercantilumAuthor Commented:
Thanks - looks like a ad a bit but anyway, interesting.
Yes maybe. But was mainly pointing out "the numbers" section that I thought you were mainly after!
Just a little thanks for the `points :-))
There are a couple of UK based VoIP providers that are seem to be directed at the smaller business, i.e., less than 1000.
If anyone is taking up a smaller VoIP (eg SOHO) then please contact me. I have tried and
The latter seems to have encompassed my wish list even to POTS fallback in event of VoIP failure.

Question has a verified solution.

Are you are experiencing a similar issue? Get a personalized answer when you ask a related question.

Have a better answer? Share it in a comment.

Join & Write a Comment

Featured Post

Improve Your Query Performance Tuning

In this FREE six-day email course, you'll learn from Janis Griffin, Database Performance Evangelist. She'll teach 12 steps that you can use to optimize your queries as much as possible and see measurable results in your work. Get started today!

  • 12
  • 7
  • 5
  • +4
Tackle projects and never again get stuck behind a technical roadblock.
Join Now