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Solutions for Small Business VoIP?
MajorBlunder asks: "I'm part of the IT department of a small but prospering software company. We have recently filled the capacity of the POTS PBX phone system we currently have installed. We are currently looking into switching over to a VoIP phone system. We have a sizable IT staff in proportion to the rest of the company, so we'd like to be able to maintain the hardware/software in house as much as possible. I wanted to ask the Slashdot readership what experiences they have had with switching over to from POTS to VoIP. Any recomendations for full end to end solutions would be appreciated, and recomendations of things to avoid would be great."
I have a small printing shop that switched 6 months ago. Our first thing was to make sure your bandwidth settings were set to the highest value. This can be set on the Vonage website and I last I looked there were 3 choices. I have seen new lines default to the lowest setting which is total crap. I have 3 lines on a cable modem connection and have never had call quality issues. I have had just about every other issue with ringing and connect delays, voicemail, caller id, etc. Most of the time you pick up and say Hello and the other person doesnt hear anything cause the call has not properly connected yet. But it saves me hundreds/month and the minor issues I have learned to live with. --
Go to the digium web site, pay them a thousand dollars, and let them install asterisk for you. Either that look around for a local asterisk provider. If you live in a metropolitan area you should be able to find a few without any problems.
evil is as evil does
... at least for us (a small business). Once you add in all of the per-line charges, the hardware, the setup fees, the broadband, and the fact that if you want to use DSL, you still have to buy at least one phone line from the phone company. Plus, of course, the reliability of broadband still isn't nearly at the level of hard telephone lines. After taking this into consideration, unfortunately, going through the local Ma Bell monopoly was still the cheapest and most reliable option for us (a business needing 3-5 phone lines).
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I work for a small firm, 100 people or so across 3 offices which are relatively close, about to add another 20-40 people. We are in a similiar position, because our old PBX system won't handle that many users without some upgrades, which we don't want to do because it is reaching the end of its lifecycle. We did a little looking around, and suprisingly the Cisco Call Manager Express was the best priced solution for us. The only way we could beat their price was going with an IP PBX system instead of a VOIP solution. They were running a promo, so there was a 39% discount from the list price on all hardware. Unfortunately, the owners decided to hold off on the upgrade and bandaid our system until late next year because we will be moving into a new building and merging two of the offices. We couldn't get a quote from Avaya, their rep never called us back, and both 3com dealers we spoke with had recently quit selling 3com. I can tell you not to go with Nortel, their solution was over 1.5x that of the Cisco solution.
The cool thing about these phones is each phone gets its own real phone number as well as internal extension. We are located in California and when we have trade shows in Florida we take one of these phones and plug it into any ethernet jack. The phone auto-configures itself and you get the same phone number and extension and you can call other people in the office on speaker as if you were in the next cubicle. Pretty rad. Hope this helps.
I've switched to using http://asterisk.org/ along with http://www.broadvoice.com/rates_compare.html. I think you'll find this Wiki to be a very useful resource: http://voip-info.org/
g ory&category=hardware or any other Analog Telephone Adapter (ATA), or you could use Softphones installed on employee PCs such as X-Lite (free), or similar.
The plan I'm using is BYOD-Lite which costs me only $6 a month and there was no activation fee, since I had my own VOIP equipment in the form of an Asterisk PBX installed on Linux. From what I can tell, they are one of the few providers who allow the use of customer supplied VoIP hardware/software, in my case Asterisk.
Something you'll have to research is what technology you want to use for hooking up individual phones to Asterisk. One possibility would be to use hardware from Digium: http://www.digium.com/index.php?menu=product_cate
Good Luck!
http://www.gloryhoundz.com/
last three years. We now have over 250 phones installed at 4 locations(including a call center). We started switching to Asterisk three years ago and grew the system to the point where everythign is Asterisk and we do all inter-office calls over VOIP(IAX trunks). The cost savings in licensing costs alone more than justifies 2 full-time IT staffers salaries.
/ Florell_astricon_2005.html
If you have some time to get comfortable with it, you will be very happy with the control you have over the system and the tremendous choice in phone hardware you can use with Asterisk. And if your company is anything like ours, they will love the cost savings.
Here's a link to a case study presentation I gave at Astricon 2005 last month:
http://astguiclient.sourceforge.net/astricon_2005
Your network is a factor here as well. Do you know how much traffic you have on the network currently? Can your routers do prioritization on different traffic types, either IP Type of Service or tcp/udp port? You want to have that understood to make sure the quality is good, so VoIP doesn't affect your usual traffic and vice versa.
You can also get switches/modules nowadays that have Power over Ethernet (PoE). So of the two RJ-45 connections (you have the physical cabling for this, right?) in a cube, one connects their PC and the other connects the VoIP appliance/phone back to the PoE port. The phone gets it's power from the ethernet cable. If those switches and the rest of your key servers and network are on UPS, the phones still work when the power goes out.
Good luck.
Where it starts getting tricky is how to connect your LAN-phones to the outside world. You can use POTS lines, or a BRI or PRI, or a T1, but that all requires additional hardware from Digium. You can get VOIP service from many cable companies and CallVantage and Vonage and such but beware! If the VoIP service requires you to use their hardware adapters, you STILL need additional hardware. You might save a little money, but other than that there is no advantage for POTS if you have to use their adapters. Plus, what a kludge that is. Your incoming call goes digial(in)--> analog(adapter)--> digital(PBX)--> analog(phone)--> digital(PBX)--> analog(adapter)--> digital(out) JUST in your PBX! If you can get/can afford the bandwidth, a 100% digital solution requires minimal hardware investment (only the phones and the PBX server). There still don't seem to be that many providers, though. But I have had pretty good luck with a couple. Broadvoice has a BYOD (bring your own device) line of rate plans that are compatible with Asterisk, though you can only have 2 simultaneous lines per account. Teliax has a flat-rate plan with up to 4 simultaneous calls, and you can have an unlimited number of simultaneous calls (subject to bandwidth constraints) using the Pay-As-You-Go plan. The other nice thing about Teliax is that it supports audio codecs other than the standard 64kbps(per incoming and outgoing channel) that Broadvoice supports. Using more efficient codecs will allow you to pack more simultaneous calls in the same amount of bandwidth.
Oh, and use a high-quality router that supports QTos packet prioritization.
If you don't know where you are going, you will wind up somewhere else.
I set up a small voip system in our office in NJ (3 lines) using broadvoice paired with asterisk - and while the service (most notably broadvoice tech support) leaves some things to be desired - our phone system is much better in terms of its feature set than it was on our POTS pbx. That said, most of the reliability issues we've encountered were the fault of our service provider, and we're generally quite happy with the switch.
The website i found myself constantly referring to in terms of making phone, software, hardware and other choices - as well as finding out the quirks and perks of each and mountains of setup info is the voip wiki.
Cheers, and good luck - you may need some in the process.
The secret to creativity is knowing how to hide your sources. - Albert Einstein
I won't argue since I have an obvious bias, but Asterisk and CCM aren't really comparable. Using CCM for 400 users wouldn't be cost effective, which is why CME exists. And yes, Callmanager is about a thousand times more complex than Asterisk, and it does a hell of a lot more as well. A lot of those features probably don't matter to a lot of folks, but Callmanager runs installs with tens (and hundreds) of thousands of phones. A bit different running, say, all the phones for a major bank or credit card processing house than running 400 phones in a small or medium sized business.
Different strokes for different folks, but you'd be stupid to dismiss either option out of hand.
Think before you leap because the potential of VOIP is tantalizing, believe me I know, I got sucked in and, to be honest, in many ways I regret it.
I'm a home user/home worker, none of my calls are that important but the quality definitely isn't there. We humans have a great capacity to blind ourselves to minor inconveniences, such as having to alter our conversational style to accommodate slightly unsychronised conversations or drops of several seconds in which the other person can't hear us but, ultimately, these things wear you down and change your relationship with your phone - you can no longer trust your phone but, like the flaws in a new lover, you excuse these things because you're so enamoured with the promise, the potential to route around the bastarding telephone monopolies that have held us all hostage for so long.
I should mention that I'm a UK user and, obviously, that places an extra burden on a US-based service. I signed up to Broadvoice because they had the best thought out plans and their support is, well, it exists which is more than can be said for many of the others. On the whole, though, I absolutely cannot recommend them to UK users because they let me down badly with regard to 0800 (UK tollfree) and 0870 (UK region-free numbers) which, although they claim otherwise on their rates pages, they simply cannot connect to, not for any amount to money. This alone renders their service redundant because, in the UK, an increasing number of businesses only provide and 0800 and 0870 number. The best example of this is Apple's UK branch who no longer accept emails - I wanted to buy about £3000 worth of computers and emailed them with a query, received an automated reply telling me that the only way to contact them was via their 0800, with no regular number to use as an alternative. This may sound like a fairly marginal problem but you wouldn't believe the number of times I've ended up using a mobile, at 20p per minute, to wait on a "freephone" service queue. Apple, BTW, lost that sale along with the chance that I'll ever again suggest their systems to a client.
So, for home users looking to save a few quid, don't buy into the dream while it's still a dream; certainly don't replace your main phoneline.
For home workers attracted to the idea of contacting clients all over the World, ask yourself if you, as a client, would be happy dealing with a service provider who you can't hear properly or with whom conversations are arduous.
For executives eager to boost their corporate careers by manfully slashing millions from their company's telecoms bill, ask yourself if adding an extra stress to the every single employee who uses the phone might not be, in the long-term, a serious blow to the company as a whole - somehow added employee stress and customer frustration never makes it onto Powerpoint presentations, but it's smart to know what's annoying the Hell out of your rank and file.
I wanted VOIP to live up to the dream, I really did - all I'm saying is that, in my case, it didn't, be aware of that amidst all the hype.
What is the deal? All you have to do is link asterisk.org and you get modded up 4 informative? geeze, is that sarcasm in the mods???
;-)
OK REAL Voip in a nutshell. You can run voip INTRAoffice then go out to copper (PRI) yourself or you can find someone to do voip trunking. (ie Your voice travels to an offsite virtual PBX and they send it to the pstn) [I say REAL voip because I'm talking business class, not running skype over a dsl line for kids to talk.]
While trunking is the coolest way to do it, sadly, voip trunking is about where cell phones were in the late 80's. Useable but you had to be sorta dedicated to the task. But I'll give you an example.
One of my clients decided to let speakeasy do the trunking. I (then) wholeheartedly recommended Speakeasy. It was a nightmare.
The problem was that we were like their third business VOIP customer. The bigger problem was that they lied to us and told us they knew what they were doing. I've been a full time geek almost 20 years. --I have NEVER had a customer support nightmare as bad as speakeasy VOIP.-- The problem was they had nobody trained on the system and they just made shit up. Then when you asked them to do what they said they could do, they would claim they never said it. I got to the point where I put EVERYTHING in writing.
If they had just come clean and said "Hey, we're learning this, give us a break" I would have helped them... But they didn't. I finally left my "dedicated" support person and went into the regular support queue. I got the support person to admit they were so new at it and they were clueless. I went back to my "dedicated" support person and told him the gig was up and he just stammered.
****But the service was good*****
The fact they were lying sacks of shit not withstanding, by the time they delivered the product, it worked well.
The topology goes like this.
You have a Edgemarc router (I think it is edgewaternetworks.com, google is your friend) and you put everyone behind it. (Voip phones, workstations and even servers)
The thing about the edgemark is that it does the traffic shaping to give priority to voice. (With speakeasy...) Every phone off hook costs you 90K. So a 1.544 T1 gives you 16 phones off hook simultainiously. (not 24) The balance is allocated dynamically to data. (Many systems use 64K per line) Speakeasy can bond 2 T's to give you 3MB if you need more lines.
Behind the Edgemark, you put a standard issue 100MB switch for your network. Spekaeasy uses (used) Cisco phones which have 2 enet ports. You can daisy chain as many phones as you like and the LAST one can be a phone or a PC. We often wire each branch phone-phone-phone-workstation.
With a SIP phone (google SIP if it is new to you) you can bring the phone anywhere in the world and plug it into a ethernet jack and you have your extension with you. No long distance etc. People just dial your local number and you can dial interoffice extensions just like usual. -coolness-
This is a big advantage of outsourcing the virtual PBX. (or setting yours up to support WAN connections.) Sadly, while this feature is possible with Speakeasy phones, (no exaggeration...) they didn't have anyone on staff smart enough to figure out how to do it. They lied to me on several occasions and said they knew how. (but no I'm not still bitter
With most trunking systems, each phone gets its own phone number (google "DID" it stands for 'Direct Inbound Dial' or some such) this is cool because they can bring their phones or use a softphone on a laptop.
Why Voip?
To me the biggest reasons to go VOIP today are to avoid the cost of a PBX or avoid the cost of long distance. Speakeasy charges about 26 bucks a month per line but since you use a virtual PBX running on their system, you have no out of pocket for the PBX. Good VOIP phones cost no more than good regular phones so that is a draw IF you are starting new or replacing equipment. But regular PBXs ain't cheap.
If you
One aspect of a VOIP system you may want to consider is the potential for redundancy.
If you should happen to choose to go the Asterisk (open source) route, the Asterisk@Home distribution installs straight off a CD and can be backed up / restored through a web browser. This means that if you exclusively use IP connected components -- T1 or POTS gateways and IP connected phones -- then you only need to shove the Asterisk@Home install CD into another server should one fail and restore a recent backup -- voice mail, configuration and all.
In addition, you can get a much higher level of service (potentially) from a service contract with an Asterisk consulting firm than your traditional Nortel / Toshiba / Avaya vendors. For example, if your phone system itself should suffer a meltdown, it is easy (in a small to medium office) to swap it with a PC. If a switch or T1 gateway should bite the dust, they are generally inexpensive enough to keep a spare around. My experience with the "big heavy" vendors is that a service contract will get you up & running in a day or less -- while a asterisk solution could potentially recover from the same type of hardware failure within an hour.
I have to recommend against using a VOIP phone service however -- getting a T1 line from a good provider is likely to be cheaper and much more reliable.
A T1 is 1.5 Mbps. Using a reasonable quality codec like G.729ab means you can fit 85 to 100 simultaneous calls into a single T1. Certainly you could stick to G.711 a/u-Law codec and have slightly better quality than G.729ab, and even with signalling overhead (either H.323 or SIP), you could fit 22 simultaneous calls into a T1.
These numbers comes from a real, working system. It's right now passing 85 calls, and consuming 1.5 Mbps. This particular VoIP router is sitting on an E1 (2Mbps) and can pass a maximum of 120 calls.
Are T1 circuits in the U.S. still so expensive? Do carriers charge more for an unframed data circuit than a PRI phone circuit? (which sounds bassackwards, but it's the new unregulated America where anything can happen) Average price for an E1 in Europe is about US$150/month for a data circuit, and depending on the phone company at the other end, about US$250/month for PRI over E1.
the AC
Hemos is like...sci-fi fans;he thinks technology is cool, but he hasn't bothered to understand the science it's based on
Actually, Bandwidth In Mirror Will Be Larger Than It Appears (BIMWBLTIA)! And, when it gets right down to it, you don't care about bandwidth anyway; you only think you do.
1. Why do companies spend $500 a month for a 1.544Mbps T-1 when a 1.5Mbps DSL connection is only $29? BECAUSE YOU DON'T CARE ABOUT BANDWIDTH (you only think you do. more below.)
2. Why does your 64Kbps codec consume more than that when you actually look at it? BECAUSE OVERHEAD COULD DRIVE THROUGHPUT AS HIGH AS 3,500Kbps! (actually that's just a theoretical, non-real world extreme, _as is 64Kbps_, more below.)
Regarding #1. Bandwidth, schmandwidth. It's all about LATENCY. Which is better for voice, a 50Mbps pipe or a 56Kbps pipe? Answer: Cannot tell from info provided in question. If, in the 60th second of a minute-long call, I deliver 3000Mb of voice data, I've given you the promised 50Mbps bandwidth. Unfortunately, there were 59 seconds of silence followed by an auctioneer's delightful squirt of one minute's words delivered in one second! Far better if they had been delivered less dramatically, but spaced evenly, over that minute. VOICE IS DIFFERENT FROM DATA IN THIS WAY. Had that been a big file, it wouldn't have made any difference. For file-type data, you pay your provider for the bandwidth. For voice-type data, you need to find a provider who can guarantee you evenly-spaced, regular delivery: that is, low latency and jitter. A T-1 has low latency, jitter and pkt loss; a DSL pipe may have identical _bandwidth_ but comes with no guarantee as to what is really important for voice, latency-jitter-loss.
That 56Kbps pipe? If it were a plain old $20-a-month land line from the phone company, that skimpy bandwidth would be delivering your voice with an end-to-end delay (latency) of less than 150ms; compare that to the VOIP standard (again, nominal) of 450ms. Your land line is still the Gold Standard for voice quality. (And yes, I have experienced better-sounding voice over Skype; Pure Friendly Magic! Great proof that VOIP can exceed even Carrier Grade. Someday, Vladimir, someday all the workers will have Carrier Grade VOIP.)
Regarding #2. I know that XorNand mentions overhead and is obviously aware of the following, but let's be explicit: overhead is more than trivial. You will never, never, never, never deliver voice at 64Kbps with a 64Kbps codec. That is a fake number, the limit that VOIP might approach asymptotically. Worst case? Your voice, encoded at 64Kbps, consumes about 3.5Mbps of bandwidth. (Also a fake number; we make a deal with the Devil, i.e. Delay, to keep the bandwidth down.)
The phone company standard codec, G-711, samples your voice 8000 times per second and represents the volume of your voice in that sample as an 8-bit number: 8bits*8,000 samples --> 64,000bps. The phone company then drops your voice onto the wire (on say a T-1 line) 8 bits at a time; each sample drops as soon as it's encoded, eight thousand times a second. Because this wire goes straight to the Central Office (say), the Telco does not need to add an IP address: there's only one place for it to go, the other end of the wire. Because the wire has a clocking device at both ends (the CSU that terminates a T-1) the Telco does not need to attach an RTP Timestamp to your voice: the T-1 circuit does that too. Because the voice samples can't leapfrog eachother in the wire, or get lost, the Telco does not need to attach a TCP sequence number or acknowledgement; the CSUs know whether a sample is to be used as voice or data, and handle multiplexing, so there is no need for a TCP/UDP port number.
You can see where this is going, right? VOIP takes the same sample, and to deliver it attaches an RTP header for timing/sequencing/codec info, a UDP header for port number, an IP header for end-to-end addressing, and an Ethernet header to get you across your LAN. That 1-byte sample is now dozens of bytes long. It's as if to carry 8000 commuters to work you sent out 8000 trains, each with a string of locomotives to pull a single commuter down the rails.