Voip > blog
Voip Blog
MRCP and Text-To-Speech
Wednesday, April 30, 2008
I recently got a press release from a company that does text-to-speech services. The press release proclaimed that the solution supports Media Resource Control Protocol, or MRCP. In theory, the press release says, it should make it easy to integrate text-to-speech into the largest call center/contact center solutions.
What is MRCP? It is defined by RFC 4463 and is described as follows:
The Media Resource Control Protocol (MRCP) is designed to provide a mechanism for a client device requiring audio/video stream processing to control processing resources on the network. These media processing resources may be speech recognizers (a.k.a. Automatic-Speech-Recognition (ASR) engines), speech synthesizers (a.k.a. Text-To-Speech (TTS) engines), fax, signal detectors, etc. MRCP allows implementation of distributed Interactive Voice Response platforms, for example VoiceXML interpreters. The MRCP protocol defines the requests, responses, and events needed to control the media processing resources. The MRCP protocol defines the state machine for each resource and the required state transitions for each request and server-generated event.
In short, MRCP makes it possible to distribute the various functions necessary for audio processing in order to implement things like text-to-speech in a scalable way. This is certainly the kind of technology necessary in large call centers!
What is interesting about this protocol is how "transparent" this protocol is. Much like SIP, the lingua franca of voice over IP, and the HTTP protocol used for web pages, the control aspects of the protocol are communicated in plain text that is human readable.
MRCP is built with the assumption that VoIP protocols like SIP and RTSP will be used. In fact, the RFC explicitly states that some functions are not addressed by MRCP and that SIP or RTSP should be used instead.
Even though this has been assigned an Internet RFC, the IETF cautions that the RFC is not a candidate for any level of Internet Standard. That being said, it will be interesting to see what other situations where MRCP will crop up. There are a lot of interesting possibilities here.
Phones Sound Different Worldwide
Tuesday, April 29, 2008
Image via WikipediaIf you've ever been to a foreign country and used a telephone, you'll know the phone just different--the moment you pick it up. Everything from the dial tone to the ring back tone to a busy signal and other things frequently sound different. Why? Custom, maybe.
What's interesting about this from a voice over IP perspective is where these tones are generated. On a typical landline, the central office generates all of these tones. You have no control over what those tones are on your handset.
In the case of an IP telephone or an analog terminal adapter, all of the tones you hear are generated on the device itself. The device knows, based on various telephony conditions, and how it's programmed, what tones it should play when.
The cool thing about this is that you can actually make an IP telephone or analog telephone adapter sound like any kind of landline you want. Let's say you're in Egypt and you have a voice over IP service there with analog terminal adapter. Depending on the device, it's possible to make that analog terminal adapter sound like a traditional Egyptian landline. Or, if you're a UK transplant into the States and you want your telephone to sound just like it does at home, you can do it.
Linksys devices allow complete control over all of the various tones that are played. Many other IP telephones and ATAs do not allow this kind of tinkering and may instead provide a simpler method for setting all the call progress tones to a specific country setting.
If you're trying to configure your Linksys ATA or IP phone, a good tool is the Linksys Localization Wizard at Voxilla, which I originally developed for them. Another useful site is World PSTN Tone Database at 3am Systems, which documents the tones used in many countries. It provides configuration information in both Sipura/Linksys format and Zaptel/Asterisk format.
Scan Your Network For SIP Devices
Monday, April 28, 2008
If you know anything about network security, you probably know what port scanners are. They are automated programs used to determine what nodes are on the network and what each node is running. Each program and computer operating systems operates a little differently than all the others. How each node responds can determine--pretty accurately--what operating system it is running and what services are available on it.
I'm sure the home user is looking at this and thinking--how can a network administrator not know what's on their network? A network administrator, short of restricting physical access to each network port, has no way of knowing what's plugged in. And if the network is wireless? Well you might as well forget knowing.
Port scanners are useful tools and identify a fair bit of information about the hosts. However, in some cases, it's not enough. IP telephones and analog telephone adapters don't relay enough information to determine what kind of device it is.
Enter svmap, part of the SIPviscious tool suite. svmap is designed explicitly to scan for SIP devices and PBX servers on a target network. This can be useful as a sort of inventory tool. Because it is specifically targeted for SIP over UDP, it's very fast.
svmap can perform the following functions:
- Identify SIP devices and PBX servers on default and non-default UDP ports
- Scans both single hosts and large ranges of networks
- Allows you to specify previous scan results as input, allowing you to focus on known SIP hosts.
- Employs different scanning methods (makes use of REGISTER instead of OPTIONS request)
- Can make all phones on a network ring at the same time (using INVITE as the method)
- Resumes previous scans
It's quite a useful tool for administrators of large voice over IP implementations.
Let's Go Down To The Party Line
Sunday, April 27, 2008
The folks at Equals have come up with a neat application for Facebook: The Party Line. Here's how it works: you add the application to your Facebook profile, add up to five of your friends to a given party line, then when you activate the party line--either through Facebook or by calling 877 428 9963--all of the other friends on your party line are called.
Equals is using Voxeo's Prophecy Platform for the development and delivery of this new application. The Prophecy Platform provides support for standards such as VoiceXML, which makes it easy to bring a web interface to the telephone. It also supports Voxeo's CCXML (Call Control eXtensible Markup Language) engine, which made it easy for Equals to deploy robust conferencing and IP telephony capabilities using SIP, but without all the complex low-level coding.
Compared to typical conferencing solutions, this is fairly straightforward. There is no software. No downloads of any kind. PIN codes--typical for conference calling services--are not required at all. You either click on a button in Facebook or call 877 4 BUZZME (428 9963). The other participants are automatically called. The call--and the service--are free!
If you'd like to see how it works, here's a video demonstration:
Facebook users can immediately install “Party Line” at www.equals.com/partyline
While the service mentions mobile phones, I see no reason you could not also do this with landlines or VoIP services like voip.com, so long as the line transmits proper Caller ID. The service clearly uses Caller ID to authenticate inbound callers. Outbound calls could be made to any phone.
Are Bundles Cheaper?
Saturday, April 26, 2008
Chances are, you've been approached by either your local exchange carrier or your cable company about "bundling" your services and saving. In other words, combining video, voice, and data (a.k.a. the "triple play") onto one bill. In a few cases, you can even combine mobile phone service, thus giving you the "quadruple" or "four" play. The theory goes, you'll save money because all those services will be on one bill.
But will you save money? Do the math for yourself. I'll use myself as an example. Comcast is my local cable company and they have approached me about doing voice, video, and data for $99. In theory, that's $33 per service and it sounds reasonable, right?
Right now, my TV service is roughly $15 a month through Comcast, data service being about $55. Theoretically, this means that for another $29 a month, I can have phone service. Is this a good deal? No.
On one hand, I can get Voice over IP phone service from voip.com for $20 a month. It's cheaper. Also, if I were to change to their bundle, I would have to upgrade my cable TV service--I don't want the extra channels, thanks--and I'd have to downgrade my data service to a slower speed. Any savings I'd get by "bundling" would be wiped out by upgrading my data package to their highest tier.
There's another reason I would never go this route: I also need to have DSL Internet service at my house. Yes, for various reasons, I have both types of Internet service. In order to get DSL service through my local exchange carrier, I must have a landline with voice service.Since I'm stuck using my local phone company's phone service anyway, my phone company has offered me a similar deal: bundle phone, Internet, and video service--to be provided over a satellite--for a similar price point. They do bundle my DSL and phone service, thus I do save $5 a month for that.
By adding video to this bundle, I would not be saving enough bundling my video service to offset the difference between what I'd have to pay for the bundle and what I pay now for video service. And, of course, to make matters worse, if I ditch video service on my cable service, I'd have to pay an extra $10 a month to Comcast.
I will admit, my particular situation is unique. Your particular situation is probably unique as well. The bottom line: look at all the options, including voice service from voip.com. Do the math for yourself.
How VoIP Providers Get Numbers In Your Area
Friday, April 25, 2008
One of the advantages of using a voice over IP telephony provider such as voip.com is that you can get local numbers outside of your local area. In other words, if you live in San Francisco, you can get a local number on New York. Or Seattle. In fact, you can even have more than one number assigned to you, and they all ring the same telephone. Neat, eh?
You might wonder how an internet telephony service provider like voip.com manages to get local telephone numbers in your area. It boils down to two options: be (or become) a local exchange carrier (LEC), or partner with someone who is.
A LEC is someone who is capable of providing you local telephony service. There are two classes of LEC: the "incumbent" LEC (ILEC) and the "competitive" LEC (CLEC). ILECs are companies like AT&T, Verizon, and Qwest that are the dominant provider of traditional telephony with a specific area. CLECs are companies like Covad, XO, Level 3, and others that can also provide you local telephony service. Your local cable company, assuming they offer telephony services, might be a CLEC also.
This might raise the question: are internet telephony providers like voip.com CLECs? Generally not. You must apply to the various state regulatory agencies in order to become a CLEC.
Most internet telephony service providers get their telephone numbers from CLECs that sell services to VoIP providers. Because not all CLECs offer services in all areas, multiple providers are often used. There are also companies like Voxbone and DIDX that give you access to a wide range of telephone numbers.
There are some exchanges where it is simply not possible to get a local telephone number for an Internet telephony service provider. This is because there is no LEC that is willing to sell service. Note that in these cases, it is often possible to get a number in a neighboring exchange. For example, when I lived in Port Orchard, WA, some VoIP providers couldn't give me a number in Port Orchard, but could give me one in nearby Bremerton or Silverdale.
Of course, if you can't get a local number where you live, or even if you can, you can always opt for a local number someplace else. Unlike your local exchange carrier, telephone numbers aren't bound to your location.
How Expensive Was Telephony Before The Internet?
Thursday, April 24, 2008
A recent episode of South Park tackled the issue of the Internet and what would happen if it just disappeared. The town, discovering that their Internet doesn't work at home, mobs the local Starbucks looking for Internet access. The Starbucks employees turn them away because they don't have any. People are trying to find out what happened to the Internet, but can't because the Internet is down! One of the mob asks "what did we used to do to get news before the Internet?"
It's hard to think of anything in modern life that hasn't been touched or changed in some way by the Internet. There is no doubt that life as we know it would suddenly be a different place. Telephony is no exception.
The Internet did not start coming into the mass mindset until the mid 1990s, though I got my first taste of it in 1990 as part of a high school project. It wasn't anything like today's Internet, of course, being mostly text based. The World Wide Web didn't show up until I was in college. Once people caught wind of NCSA Mosaic and Netscape Navigator, the game was on!
I do remember a pre-Internet world. And while I could talk about a lot of aspects of this pre-Internet world, let's talk about it in terms of telephony. Telephony is a rather mature industry, having been around for over a century. Innovations, such that they were, were slow. And calls? They were expensive. How expensive? Here's an AT&T ad from 1984:
$10 an hour for a weekend or evening state-to-state long distance calling? For $19.95 a month, or $199 a year, you can get a phone line and unlimited calling within the U.S. and Canada through voip.com. Or how about the ad from 1986 that proclaims 64 cents a minute calling to the UK?
Two decades later, thanks to the Internet and worldwide deregulation in telephony, calls from the United States to the UK--using voip.com, at least--can cost as little as 1.2 cents a minute! Call charges vary by destination and whether or not you are calling a mobile phone, but even voip.com's most expensive rate for calling a UK number is a far cry from 64 cents a minute!
So to answer the original question of this post--what did we do before the Internet--at least with respect to telephony, we spent way too much money!
Ringer Equivalency Numbers And You
Wednesday, April 23, 2008
One of the things you've never probably heard of is something called REN, or Ringer Equivalency Number. In short, it is the amount of electrical load that a telephone ringer puts on a line. Modern telephones all have a REN value printed on the bottom of the telephone.
A REN value of 1 represents the loading effect of a single "traditional" telephone, for example a Western Electric Model 500 desk telephone. Most telephone these days have a REN far less than that, though some have more. My cordless telephone has a REN of .08.
Why is this even an issue? There is a difference between what an analog telephone adapter is able to support in terms of REN and what a typical landline supports. A Linksys SPA-2000 supports a REN of 3, whereas a typical landline supports a REN of 5.
Why does this matter? If you're distributing VoIP throughout your home and you have a lot of old telephones, not having enough REN will prevent your telephones from ringing, or cause them to ring improperly. This is because there is not enough ring voltage being sent to the phones!
There are a couple of choices to resolve this problem: reduce the load by unplugging or swapping phones to ones with lower REN values, or increase the REN output. How do you do that? Using a device like the Ring Voltage Booster II from Mike Sandman Enterprises.
The Ring Voltage Booster II takes whatever ring voltage it gets and boosts it to an acceptable level. You plug one end into your analog telephone adapter, the other into your telephones. The Ring Voltage Booster II will take a ring voltage as low as 30VAC RMS and boost it to 90VAC at 20 cycles. It also supports a REN of 7.5, meaning you can plug in 7 of those old Western Electric phones and it will ring them all!
Latency and VoIP Call Quality
Tuesday, April 22, 2008
Back when I was a kid in the 1980s, there was a time when I lived in California and my mom lived in Hawaii. In those days, calls between Hawaii and the mainland went over the satellite. This meant there was, at minimum, a half-second delay--complete with a bit of echo.
A couple of decades later, that problem is but a distant memory. Fiber optics runs between the mainland and the Hawaiian islands, making calls sound like they're right next door. The echo and delay are gone.
What I experienced on those calls to Hawaii as a kid is something you might experience on a VoIP call--latency. In short, it's the amount of time it takes your voice to reach the other end of the connection. What's reasonable? 150 milliseconds. Anything over 300 milliseconds and the voice quality deteriorates.
This is one reason why VoIP on dialup is typically not supported. Dialup involves round-trip packet times of at least 300-350 milliseconds, and that's before it even gets to the Internet! When you add the round-trip time to a site on the Internet, the round-trip times can start approaching what I experienced with the calls traveling over satellite.
In order to achieve latency figures around 150 milliseconds, the following things must occur: the sender's voice has to be encoded into IP, sent out over the network, arrive at the service provider, be sent to the receiving party, received by the receiving party, decoded and played for the receiving party.
While some of the latency can be controlled by the providers--namely stuff that happens within their network--a large chunk of the latency occurs between you and the provider over your public Internet connection. Variations in the latency and packet loss can play a huge role in how your VoiP calls sound.
Your VoIP Device May Be Hacked!
Monday, April 21, 2008
If you configure your own VoIP device, you might become a victim of having your VoIP devices configuration changed without your knowledge! This could lead to all kinds of untold information leakage or worse!
The problem isn't specific to VoIP devices, it's anything that has a web interface. The problem is with something called Cross-Site Request Forgery (CSRF). What is it? Quoting from the Cross-Site Request Forgery FAQ:
Cross Site Request Forgery (also known as XSRF, CSRF, and Cross Site Reference Forgery) works by exploiting the trust that a site has for the user. Site tasks are usually linked to specific urls (Example: http://site/stocks?buy=100&stock=ebay) allowing specific actions to be performed when requested. If a user is logged into the site and an attacker tricks their browser into making a request to one of these task urls, then the task is performed and logged as the logged in user. Typically an attacker will embed malicious HTML or JavaScript code into an email or website to request a specific ‘task url’ which executes without the users knowledge, either directly or by utilizing a Cross-site Scripting Flaw. Injection via light markup languages such as BBCode is also entirely possible. These sorts of attacks are fairly difficult to detect potentially leaving a user debating with the website/company as to whether or not the stocks bought the day before was initiated by the user after the price plummeted.
What does this mean? It means if you're like the typical home user, your VoIP device has a web interface that is either configured with no password at all or a default password that is well-known. That makes it possible that, under the right set of circumstances, your VoIP device may become reconfigured without your knowledge!
What can you do? The easiest thing you can do is to pick a good, non-default password for your web interface. This will prevent any of these attacks from working.
If you’re a Firefox user, another thing you can download is a copy of NoScript. NoScript disabled JavaScript for web sites you don’t explicitly trust. In addition, NoScript has a number of XSS-related checks in it to thwart CSRF-related attacks on well-known websites.
E911 and VoIP: Location, Location, Location
Sunday, April 20, 2008
As you probably know by now, VoIP services like voip.com must, by law, provide E911 service to you as part of the normal service. That means when you dial 911 from your telephone connected to our service, 911 service will operate more or less as it would from a landline with your local exchange carrier, connecting you with the appropriate public safety answering point (PSAP) and communicating your location information to them.
At least that's how it's supposed to work. However, there are a lot of moving parts that make this difficult. The first issue is: where are you?
One thing that E911 communicates to the PSAP is your physical location. This is fairly straightforward with a traditional local exchange carrier. They know what pair of wires you're coming in on. They know exactly where that pair of wires is located. They know precisely what PSAP(s) it should use to relay your call.
In the case of VoIP providers, there are no physical wires. All calls come in over IP. How does a VoIP provider know where you're calling from? Fairly simple: you have to tell them. As part of your residential service on voip.com, you must divulge the physical location where your service will be used. Other services require similar disclosures.
Because of the difficulties and complexities of connecting into the 911 infrastructure, voip.com and other providers outsource the E911 parts of their operations to third parties. These third parties handle figuring out which PSAP to communicate your 911 call to and what data to communicate to the PSAP.
Unlike a traditional landline situation, where the wires are basically in the ground and can't easily be moved, an analog telephone adapter is relatively straightforward to move from one place to another. While that makes it convenient for the customer, it can wreak havoc when you dial 911. It is therefore very important that you keep your location information up-to-date, else 911 may send help to the wrong address.
What Is A Phone, Anyway?
Saturday, April 19, 2008
Back when I was a kid, I pretty much took for granted that a telephone looked something like this:
However, what we consider a telephone has changed quite a bit. According to Wikipedia:
The telephone (from the Greek words tele (τηλέ) = far and phone (φωνή) = voice) is a telecommunications device that is used to transmit and receive sound (most commonly speech), usually two people conversing but occasionally three or more. It is one of the most common household appliances in the world today. Most telephones operate through transmission of electric signals over a complex telephone network which allows almost any phone user to communicate with almost anyone.
Or to quote Merriam-Webster's Dictionary:
an instrument for reproducing sounds at a distance; specifically : one in which sound is converted into electrical impulses for transmission (as by wire)
With these definitions in mind, let's see if these conventional definitions hold up with objects in today's world.
First, let's look at the cordless phone we use at home. Yes, it does part of it's job wirelessly, but it still transmits and receives sound from a third party.
How about a mobile phone, such as my Nokia N95? While it has a ton of other features, I make and receive calls to the same telephone network my cordless handset uses. Furthermore, calls can go over WiFi using SIP to connect with a third party VoIP service.
How about a computer or a device like a Nokia Internet Tablet? It can also be a phone. If you have a headset and VoIP software such as Skype or the voip.com softphone, you can make and receive calls to the same telephone network as your landline phone, with the added benefit that in many cases, it's cheaper.
So what is a phone? Does it really matter as long as we're all connected and can all communicate on the same telephone network?
Image from Alejandro The Great
Distributing VoIP Throughout Your Home, Part 3
Friday, April 18, 2008
If you've read the previous two articles, you've found your Network Interface Device (NID) or demarc and you've managed to get yourself disconnected from the PSTN. The next step is to figure out how the wiring is done in your house.
There are two common methods, and unfortunately, they're often both used within the same house. The first method is what they call a "star" or "home run" method. Wires run from a central point--sometimes at the demarc, sometimes to a different location--to the locations where the phone jacks are. This is the preferred method.
The other method, used in older homes, is called the series or daisy-chain method. One line runs from the demarc into your house. Each telephone jack is simply daisy-chained into the next one until you get to the end of the chain. Note that the star method also employs the series method to a certain extent, but the daisy chain is limited to a specific area of the house.
Regardless of the method, as long as all of the wiring connects to a single place, and as long as you are disconnected from the PSTN network, you should be able to plug in your analog telephone adapter to any available telephone jack. Congratulations, your voice over IP service is on every phone in the house.
On the other hand, you can avoid the house wiring issues entirely by switching to one of those cordless systems with multiple handsets. This is what I ended up doing in my new house. In addition to eliminating wiring from the equation, you can also benefit from features like handset-to-handset paging.
These last three articles give you a rough overview of the process of hooking all your analog telephone handsets to a voice over IP provider. However, there are intricacies and details that I'm sure I left out. If you are at all serious about doing this, I highly recommend the detailed tutorial by Michigan Telephone.
Distributing VoIP Throughout Your Home, Part 2
Thursday, April 17, 2008
In our last piece, you found your Network Interface Device (NID) or demarc. Now I'm going to explain what you can do with your NID.
The NID boxes attached to most recently-built homes make it possible to completely disconnect all your home wiring from the telephone network by simply unplugging a single phone cord. This is useful for two reasons. First, if you are having problems with your telephone line (e.g. noise), you can easily determine whether or not the problem is inside your home wiring--typically your responsibility--or the problem is outside your home, which is the phone company's responsibility. You can plug in your own telephone at the demarc point and quickly make this determination.
Note that older NIDs or demarcs also provide this capability, but not with a standard RJ-11 jack. You'd either have to use a butt set, or destroy a RJ-11 cord, strip the wires, and hook up the wires directly to the terminals. I'd read more about telephone wiring on the Michigan Telephone site that goes into this in a lot more detail.
The other, more relevant reason you would want to disconnect your house from the PSTN network is that you have cut the cord with your PSTN provider and you want all the phones in your house to use a Voice over IP service. Even on lines where the PSTN provider is not providing dial tone, voltage is being sent down the wire. That voltage is enough to light up some phones, but it's also enough to fry analog telephone adapters from a service provider. This makes it critical for you to disconnect your home from the PSTN before you plug an analog terminal adapter into it.
Once you've done that, can you plug your analog telephone adapter in and expect it to go to all the phones in your house? Not quite yet. In the next piece, we'll go over the different ways your phone lines might be connected in your home.
Distributing VoIP Throughout Your Home, Part 1
Wednesday, April 16, 2008
If you have service from voip.com or one of the many other, similar service providers, chances are, you've thought about making it so all the analog telephones in your household can take advantage of the service. It certainly is possible to do, but it requires knowing a bit about the wiring in your house.
In most homes, all phone jacks appear to be hooked together. That allows you to pick up the phone in any room in your house and have it work properly, or put an answering machine, etc. The theory goes, you should be able to plug in your analog telephone adapter provided by a Voice over IP service and it should work, right? Before you do that, read on!
The first thing you need to know is where your Network Interface Device (NID) is. Now this sounds like a computer-related thing, but it's not. It's where your home interfaces with the public telephone network. Other names for this are the demarcation point, minimum point of entry, or simply demarc.
In most recently-built single family homes, the demarc is a little grey box attached to the side of your home. It is relatively easy to get into this box with either a flathead or phillips-head screwdriver, depending on who made it. In older single-family dwellings, it might be a fist-sized box that contains several bolts with wires screwed in.
In multi-family dwellings (e.g. apartments), it may be somewhere not easily accessible. The building owner has to provide an easement to the telephone company to allow access these connection points. If it's not obvious from walking around the building, you'll have to check with the building owner to find out where it is.
Have you located your demarc? Great. That's the first step. In the next article, I'll tell you what you need to do at your demarc.
VoIP Devices Are Not Fun To Configure
Tuesday, April 15, 2008
One of the ways I first got known in the VoIP industry was figuring out the arcane art of configuring Linksys (then Sipura) devices. The SPA-2000, the SPA-3000, and a whole host of others. Just to give you an idea of what you're up against when configuring one of these devices, let me show you just the Ext 1 tab of an old SPA 941 I just pulled out of storage to take some snapshots for this piece. Click on the image to see it up close and personal:
In order to get things working right, quite often, you have to jump into the advanced settings, which I can't even get in a single screen shot!
Of course, that's not the entire configuration. There are several other tabs in the interface to configure other arcane features on the device that may or may not be needed for compatibility with a particular Internet telephony service provider, IP PBX, SIP Proxy, and so on. It's enough to make your eyes spin!
This reason alone is why, on the whole, VoIP providers want to manage the configuration of these devices as much as possible. These devices are complex to configure. One wrong setting can cause untold issues with your VoIP service.
Those providers that do offer bring your own device support--voip.com being one that does--typically provide you the necessary information with the disclaimer that they won't support your device if it doesn't work as expected. It's most certainly for advanced users only.
If it's at all possible to avoid having to configure your VoIP device, I'd recommend letting the provider handle it, particularly if you are not inclined to dug into the configuration of these little beasties. Otherwise, go to town!
Symbio DECT VoIP Phone By Thomson
Monday, April 14, 2008
The folks at Thomson commissioned the folks at EliumStudio to design a Cordless DECT VoIP Phone and Internet Radio. It is quite an innovative looking handset, and it has some innovative technical features as well.
What's an Internet Radio doing on a telephone handset? Not sure, but they support a list of stations by location or desired musical style.
Why does a telephone have an RSS reader? Not entirely sure about that, either. My mobile phone has an RSS reader, but it also has a full web browser. The feature on the Symbio is called Myinfokiosk. The spec sheet or the brochure don't have a whole lot of information on this feature, but my guess is that it's fairly limited.
The phone gets a good six hours on a charge and communicates between the base and the handset using an HD codec (G.722). This should improve the quality of your VoIP calls substantially, or at least the DECT portion of the call won't make the call sound any worse.
Other features include:
- Phone usable with analog or VoIP service (subject to availability)
- Caller ID (1)
- Hands Free (a.k.a. Speakerphone)
- 200 Name Phonebook
- Last 50 incoming calls
- Multi-line support (1)(2)
- SMS Text Messaging (1)(2)
- Phonebook synchronization (2)
- Myinfokiosk (RSS reader) (2)
- Color screen with auto-orientation
- Built-in Stereo Speakers (3 watt)
- Radio Alarm
- Custom MP3 or WAV Ringtones
- Station name and other info such as artist/song’s name (station-dependent feature)
(1) Operator-dependent feature (2) Not all models support this feature
Pricing and availability on this handset is not available anywhere that I can find. Wish I could get my hands on one to check it out!
Flashphone: If You Can See A Web Page, You Can Use It
Sunday, April 13, 2008
If you've tried to use voice over IP service from behind a firewall--particularly a corporate one--you might find yourself having problems. I've tried Gizmo5, SightSpeed, and many other SIP-based clients from my corporate laptop, and no dice. The reason? The firewalls simply don't forward most traffic to the Internet. However, you can get out to the Internet through an HTTP Proxy.
Skype has been the only thing that's worked in just about any environment I throw it in. It even traverses my HTTP Proxy. But what do you do when your corporate computing policies prohibit you from installing applications on your PC?
Enter Flashphone. Flashphone utilizes Adobe Flash within your web browser to allow you to make and receive calls over VoIP. This is nice because 98% of web browsers out there support Flash. This means you won't have to install software.
Flash even works on mobile phones. However, Flashphone is only supported on Windows Mobile if you load the appropriate version of Adobe Flash--provided on the Flashphone site.
The other thing that Flashphone does that ensures it will work in most all situations is that it talks only in HTTP packets. This means it will traverse pretty much any firewall or web proxy that permits HTTP without any special configuration necessary.
Flashphone tunnels HTTP packets to their backend servers where they will speak SIP. They will even allow you to program in your own voice over IP service configuration to allow you to make inexpensive long distance calls.
The idea is pretty neat. They're not charging anything for use of the service. I do wonder where these guys are going to make money with this. Licensing deals, maybe? What say you?
It's Not Just Voice
Saturday, April 12, 2008
VoIP originally started as being Voice over IP. Predominately, it still is. However, the V in VoIP also stands--increasingly--for video. The technology to transmit video in realtime at any acceptable framerate has been around, but it has been proprietary, expensive, and not widely deployed.
For some reason, we've had this love affair with doing two-way video. Or at least someone has. AT&T was promising video calling was just around the corner since the mid 1960s. They promised it again in 1993 in one of their "You Will" ads:
In this decade, we're starting to see a couple of important trends, which makes video over IP reasonable:
- A wide deployment of cameras. People aren't necessarily buying webcams in large numbers, but we're starting to see cameras embedded into laptops and monitors. The simple fact it's there makes video calling possible.
- The bandwidth is starting to be there to do video adequately.
- There are a number of high-quality solutions out there. SightSpeed makes a great video tool. Skype also has a high-quality video option that--on the right equipment and bandwidth--looks absolutely fantastic.
Going away from one-on-one communication to media generation and consumption, there are now plenty of ways to both consume and generate video content over an IP network. Sites like Hulu, the U.S. television network sites, and even content creators themselves are distributing video over the Internet.
In terms of generating content, the king of user-generated content, YouTube lets you record your videos directly into their site straight off your webcam. You also have sites like Seesmic that are nothing but threaded video conversations. If you're on the go, you can record videos with Qik or Flixwagon and stream them live to the Internet complete with text chat messages straight to your phone while you're taking video.
While video may not be anywhere near the majority when it comes to one-on-one calling, it's there. I heard Jonathan Christensen from Skype say at a panel at the Spring 2008 VON.x Conference that there was a video component in 30% of their calls. 30%? That's nothing to sneeze at.
I think it will be a while before video over IP hits the mainstream. My barometer for that is: is my wife using it without my help. Currently, she's not, so we've got some time yet.
People Listening In On VoIP Calls? Give Me A Break!
Friday, April 11, 2008
Every so often, I come across a panic piece by someone blogger who thinks that their VoIP calls are insecure because their traffic can be sniffed and decoded rather easily. If only people would implement encryption, they say, this problem would be resolved.
While on the face of it, I tend to agree with these people, I am going to put forth another viewpoint. One that is written across The Hitchhiker's Guide To The Galaxy in large, friendly letters: DON'T PANIC.
First, let's talk about SIP--the lingua franca of VoIP--and what information is actually sent:
- Call Information Goes In The Clear: Yes with SIP, whom you are and whom you're calling does go in the clear. Yes, you can use Transport Layer Security (TLS) to encrypt this information.
- The Voice Data Goes In The Clear: What you're saying also goes in the clear as well. This can be mitigated by using SRTP to encrypt the data portion of the call.
That's quite a lot of information. A scary amount, in fact. But let's look at your typical analog phone line. If someone were to covertly listen in on your phone line, they'd get the same information. Outbound calling would be communicated via DTMF tones. Inbound calls are communicated via a "burst" of data during the ring cycle. That plus the contents of your VoIP conversation. Yup, they pwn you.
Let's look at what it takes to actually pull off this kind of sniffing attack against a VoIP call. With a landline phone, you essentially have to have physical access to the line. Depending on the neighborhood--or the office--this could either be easy or hard. However, the demarc outside your house is always quite accessible. Office building demarcs are a little tougher to get into.
To do this same kind of attack in the VoIP world, you have to be at a location where the call is traveling through, either at the telephone service provider or the ISP. Since the path a call might take over the Internet isn't guaranteed, there are only a couple of points where it is practical to intercept a VoIP call: on the client or proxy's premises, or at the ISP used by either endpoint.
Guess what? Nothing's really all that different. Both VoIP and PSTN calls can be intercepted--at essentially the same locations, too. Why are people freaking out?
Mobile Phone Roaming Costs: Out Of Control
Thursday, April 10, 2008
I am thankful that I work for an employer that, when they do ask me to travel outside the United States, they pay my mobile phone bill, no questions asked. Not everyone's in that situation. In fact, I'm going to be indirectly in that situation when my wife goes to Italy next year.
International roaming rates are expensive. The U.S. carriers charge a minimum of $0.99 per minute to make or receive a call while you're abroad. If there's long distance involved, heaven help you. It's somewhat less egregious roaming in Canada--$0.69 per minute on AT&T--but only somewhat.
There was a time where this was a problem within the United States. Back in the early days of mobile telephony in the United States, if you went out of your local calling area--and the calling areas were relatively small in those days--expect to be hit with roaming charges--making or receiving calls. Assuming it even worked.
In the late 1990s, AT&T Wireless--not the current iteration of this company--changed the game with their Digital One Rate plans. 450 nationwide anytime minutes for $60--anywhere in the United States. It was a game-changer. Pretty soon, all the carriers went with national rate plans. Roaming charges--within the borders of the United States, anyway, were a thing of the past.
The European Union has been doing stuff to curtail roaming charges within the EU. While this is great, it only applies to people who have service with a carrier within the EU. Everyone else--including us Americans--will continue to be charged exorbitant rates.
What's a world traveler to do? Assuming you already have a compatible GSM phone, there are essentially three choices for dealing with this:
- Paying the exorbitant roaming rates, which if you're only going to be in a country for a day or so, might be the easiest option.
- Get a local SIM card in the country you're in. This may be difficult to set up beforehand, but it might be something you could do once you land at your destination. Either that or have a local in that country send you the appropriate SIM card.
- Get one of these "global" SIMs from companies like MaxRoam, United Mobile, and Universal Mobile. These companies are virtual network operators that aggregate access to a number of different GSM networks worldwide. They also use VoIP and callbacks to reduce the overall cost of the call and give you inbound numbers in many different countries.
Of course, the problem with #2 and #3 is that you have to give your contacts a different phone number or simply forward your cell phone to the new number. If you don't have a GSM handset, you've also have to acquire one of those.
Note that you can't simply use one of these global SIMs in the U.S. because they are passing exorbitant roaming charges they pay onto you. However, that seems to be a short-term problem.
In short, roaming internationally sucks. There are ways to lessen the pain, but it's still more expensive than it needs to be.
Ain't Nothing Like The Real Thing
Wednesday, April 09, 2008
I've owned a pinball machine for a number of years. For the past few years, the pinball machine has been broken and taking up space in my office. Recently, I've had someone come out and get it working. After a few visits by the repairman and some part replacements, I have myself a working pinball machine. It still needs a few repairs, but that's beside the point.
The machine in question is an Eight Ball Deluxe Limited Edition. I paid for an emulator for this pinball machine over a decade ago, long before I bought the real thing. At the time, it was pretty good. It rendered the table nicely and the game play was authentic enough.
Playing a rendition on a computer and playing the real pinball machine are completely different experiences. The ball does crazy stuff like bounce off the glass and over rollovers. You can nudge the machine and change how the ball bounces so that it goes you way. It has a feel to it that a computer just can't emulate.
So how does this in any way relate to VoIP? Glad you asked. Voice over IP has to turn your voice into ones and zeros. It does this through a series of rapid sampling of your voice and then encoding it into ones and zeros using something called a codec.
Codec is a combination of two words: COmpressor and DECompresser. The audio must be compressed and sent off to the remote end, which must decompress it. There are various codecs out there, some of which optimize for quality (at the expense of bandwidth) or optimize for bandwidth (at the expense of quality).
Codec usage is one thing that can affect quality. Packet loss and latency plays the largest role. If some packets get lost, one party might sound like it's underwater, really garbled, or may disappear entirely.
There are a lot of moving parts involved in making your voice over IP call sound almost as good as being there. Video can help bridge that gap somewhat, but at the end of the day, it's not always enough to call--or video--it in. Sometimes, you've got to have the real thing--that person, face to face, in the same place. That's a problem VoIP can't solve.
Grandstream Goes PBX With The GXE 502x Products
Tuesday, April 08, 2008
I've always been impressed at the ambition of Grandstream's IP telephones. I have a Budgetone 100 and a GXP2000 sitting in my VoIP gear box in my office. They were decent phones for their time, and were fairly functional. However, I always found myself wanting more in terms of overall product stability and build quality from these two phones.
Recently, I've received word they've taken it a step farther and built their own IP PBX: the GXE 502x line of products. There are two models: the lower-end GXE 5024 and the higher-end GXE 5028. The main differences are the number of ports--the GXE 5024 supports 4 analog extensions whereas the GXE 5028 supports 8, the number of conference rooms (2 versus 4), and the amount of flash memory to store things like voicemail.
Both products will support up to 100 registered VoIP extensions with 50 simultaneous VoIP calls supported, can connect to two analog telephone lines for making and receiving calls, and support Power-over-Ethernet.
The product isn't shipping as of yet, but is currently in beta test. I haven't decided if I want to sign up and get the product to test it. Of course, I have a PBX here from a different vendor that I still need to test with.
One error that I see in the manual up-front is that Grandstream mixes up the use of FXS and FXO. Anyone who have ever taken an analog telephone line and plugged it into the FXS port of an analog telephone adapter can tell you that getting this wrong is a great way to wreck your equipment. FXS ports are for analog telephones, FXO ports are for analog phone lines that connect to a local telephone company.
This could be an interesting product. Hopefully the build quality, both in hardware and software, will be improved over my past experience.
How Much Data Does VoIP Use?
Monday, April 07, 2008
I thought about this after reading about the onerous terms and conditions that AT&T and Verizon Wireless have on their mobile data service. Basically, if you go over 5 gigabytes of data in a month, you may end up paying $0.50 per megabyte overage--in other words, $500 a gigabyte.
With that in mind, let's look at what a typical VoIP call would take. In this example, I used Skype, but I assume even a SIP-based call would have a similar profile. I did a 20 second call to our favorite echo123 user, i.e. the Skype Call Testing Service.
In that period of time, Skype generated 385,855 bytes of network traffic, or about 377 kilobytes of data. For a minute of VoIP conversation, that's just a hair over a megabit of data.
Yes, I realize my math may be somewhat off here as I include call setup and tear down packets. Then again, keep in mind Skype does a lot of things that are hard to account for. Are we really account for just voice traffic here?
For the sake of argument, I did the same experiment with Gizmo5, which uses industry-standard SIP for communication. I got roughly 357 kilobytes of data for a minute of call. In this case, I was able to focus only on the voice stream itself--not the call set up, etc.
This is hardly an apples to apples comparison, and you could argue that I could have gotten my numbers more precise. But for the purposes of ballpark comparisons, it's safe to say that as long as you don't talk for hours on end over a mobile Internet connection, you're probably safe with the occasional Skype or Gizmo5 call.
That is, assuming you have an "unlimited" plan. If you pay per megabyte, all bets are off.
How "Green" Is Video Over IP?
Sunday, April 06, 2008
I've had a couple of PR firms play the environmental card when pimping their services to me. I suppose being "green" is the next big thing--one I am not going to argue with. However, I always tend to be skeptical of any such claims.
Any form of communication that isn't face to face could be construed as environmentally friendly on some level. Unless you can reach the person under people power, you're likely having to burn some fossil fuels to reach them. Not to mention all the time and such that it will take.
What these video services do--things like Skype Video, SightSpeed, Gizmo5, Yugma, and others--is make it possible to communicate more information in real-time than is possible with simply a phone call. In the case of Skype, it's possible to get a higher quality audio call than a normal telephone would give. In the case of Yugma, it's sharing your desktop so you can see what the other person is seeing.
Anything that limits the amount of fossil fuels burned is certainly a good thing in my book. Using these tools in the right circumstances could certainly eliminate a trip in a gas-guzzling car (or airplane).
But one thing I've discovered over the years is that these kinds of tools--along with many others--make it easier to communicate, but it increases the desire to actually meet in person. This means you're likely back to some gas-guzzling alternatives.
On the whole, however, I agree with the assertion these video over IP tools are green, though I would argue their"green" benefits are probably not as substantial as the PR folks are saying.
What do you think about this? Are video over IP services helping in some small way to give us a greener planet? Is this just PR speak? Let me know what you think!
How Will Text To Speech Voice Sound Over VoIP?
Friday, April 04, 2008
You may have seen this video floating around the Intertubes:
While it might sound real, I can certainly tell it's not real. It's a text-to-speech system by the folks at IVONA. It is certainly better than a lot of the text-to-speech stuff I've heard over the years, but I wouldn't have a hard time telling it apart from a human.
One obvious application for this text to speed technology is interactive voice response systems. You know, those systems that you run into when you call a corporation's 800 number. Even voip.com has one of these systems answering their 800 number.
Right now, what happens is that a person has to record the prompts. They may have an in-house person speak the prompts or they hire a professional such as Allison Smith to do it. Either way, it costs time and or money.
Imagine an IP-based PBX where you could simply type in what you want the PBX to say. It would speak it for you and it would sound close enough to real for most people. No more paying for a professional or spending hours getting your voice prompts right. Just type it in, and you're done.
With this hypothetical IP PBX, calls would come in over a number of methods--including IP. Depending on what codecs are used for that particular call, the voice may sound perfectly natural, or it may sound like crap.
The thing is, to transmit your voice over an IP protocol, your voice must be sampled. The codecs often leave something to be desired--either making the voice more metallic-sounding or worse. If the voice isn't perfect, putting it through a VoIP call is going to amplify any imperfections.
To give a real-life example of this, when I call home on my cell phone and talk to my young daughter, I have a difficult time understanding her on the phone, even though I can understand her fine in person. While a mobile phone doesn't necessarily imply VoIP, the same principle of codecs applies since an analog voice must be converted into something digital so it can be transmitted over the mobile networks. The process of encoding and decoding the voice--sometimes multiple times--degrades the voice quality.
How is this stuff going to sound over VoIP, hard to tell. If anyone can do any real-world testing of this, leave some feedback.
New Sheriff In Town
Thursday, April 03, 2008
Welcome to the Voip Blog hosted by voip.com. You may have noticed that things have been a bit quiet over the past several months. I've been hired to change that.
Chances are, if you read blogs about voice over IP, you've come across my work either on phoneboy.com or on The VoIP Weblog or maybe have seen old posts on the Voxilla forums where I've helped people make their VoIP gear work with a variety of providers. I also use a number of VoIP-related tools on Macs, PCs, Linux, mobile phones, and so on.
I started in the industry as a systems administrator, and have spent the last 10 years or so doing technical support for Enterprise-level customers. I used to get really deep into the technical details--and I still do as a result of my job. However, I have become increasingly in favor of technologies that work and are reliable.
While I still like lots of features and functions in my technology, at the end of the day, it needs to be simple and, most importantly, needs to work. I treat VoIP no differently in this regard. I've fiddled with my share of solutions that require way too much care and feeding, yet provide very little value.
Since the blog is on the voip.com website, I have to point out things that voip.com is doing from time to time. It's only fair, after all. When I'm not doing that--which will be most of the time--I will let you know what's going on related to the world of voice and video over IP. More importantly, I'll let you know what works and where the value is, if it has it.