August 31, 2005

VOIP-Problem solved

VOIP Testing

Success! The major issues I have been having since this post have been resolved. My VOIP server ( asteriskwin32.exe) is now running as a service and the biggest changes were in the sipdefault.cnf. the changes are listed here for my Cisco 7960 and 7940 phones. Seems like a human error problem but I really didn't know that these setting were important or needed to be changed.


# Outbound Proxy Support

outbound_proxy: "" Had to add this line and put an address in it

# RTP Call Statistics (SIP BYE/200 OK message exchange)
call_stats: 1 ; 0-Disabled (default), 1-EnabledAnd because I was crossing timezones...
# Time Server (There are multiple values and configurations refer to Admin Guide for Specifics)
sntp_server: "69.25.96.11" ; SNTP Server IP Address
sntp_mode: directedbroadcast ; unicast, multicast, anycast, or directedbroadcast (default)
time_zone: CST ; Time Zone Phone is in
dst_offset: 1 ; Offset from Phone's time when DST is in effect
dst_start_month: April ; Month in which DST starts
dst_start_day: "" ; Day of month in which DST starts
dst_start_day_of_week: Sun ; Day of week in which DST starts
dst_start_week_of_month: 1 ; Week of month in which DST starts
dst_start_time: 02 ; Time of day in which DST starts
dst_stop_month: Oct ; Month in which DST stops
dst_stop_day: "" ; Day of month in which DST stops
dst_stop_day_of_week: Sunday ; Day of week in which DST stops
dst_stop_week_of_month: 8 ; Week of month in which DST stops 8=last week of month
dst_stop_time: 2 ; Time of day in which DST stops
dst_auto_adjust: 1 ; Enable(1-Default)/Disable(0) DST automatic adjustment
time_format_24hr: 1 ; Enable(1 - 24Hr Default)/Disable(0 - 12Hr)

All in all, looking back on this. The setting do make sense but it was very frustrating trying to figure out which ones would make the difference. Now all the phones are working great and Broadvoice is( without fail) serving all of our needs. Three-way calling is a wonderful feature that I have now had a chance to use quite a bit because of how easy the Cisco 7960 and 7940 phones are to use. Now if I could just get a Vonage account to work...

Posted by Sean at 02:30 PM | Comments (0) | TrackBack

August 23, 2005

VOIP Takes Off

VOIP in every home?

More and more people like you and me...
According to analysis done by the TeleGeography research group, VoIP usage is on the rise as more and more people abandon their traditional telephone companies for broadband telephone service also known as VoIP (Voice over Internet Protocol).

According to the analysis VoIP usage has toped 2.7 million in the second half of 2005 compared to just 440,000 just a year ago.

The revenue for VoIP is also on the increase and it is expected to top 3 billion USD by the end of 2007.

This raises a very good question. What is the ratio of minutes of VOIP versus the minutes of plain old telephone service. My take on this is simple. The people that have VOIP use it mainly as a toy/only for certain types of calls i.e long distance and the rest of the time they are still using there plain old phone or cellular phone. On this point I could be wrong. What are your thoughts?

Posted by Sean at 09:50 AM | Comments (0) | TrackBack

Broadvoice and Asterisk

VOIP Testing

I am in need of some help... I have a asterisk server hosted out on the net and it has started a new issue. The problem seems to stem from the fact that the clients are all behind firewalls and the server is not sending phone calls to them. The service I use is broadvoice and that part seems to be working alright but I can not for the life of me receive a VOIP phone call through the system. The testing environment I was using before involved a locally hosted Asterisk server and so the big change was taking it outside my local network. I use Cisco 7960 and 7940 phones and have multiple phones set up to ring for various VOIP calls but at this point none ring and that is the issue. Any suggestions would be helpful at this point. I will keep testing and let you know if I find a solution or cause for the problem. Please check back as this is really driving me nuts and MUST be solved soon. Thank you

Posted by Sean at 09:21 AM | Comments (0) | TrackBack

August 03, 2005

Asterisk or Cisco or The cable company

Asterisk with a Cisco phone Revisited...
After many, many tests of calling locally, long distance and international I have found that the biggest problem I am having is with one phone inparticular which is behind a cable modem and NAT. It really doesn't make that much sense.
Here is the scenario:
Conference call scheduled for 8:00PM
I am at home and need to get onto the call. Cell phone dies. Plug in a Cisco 7960 phone point it at a tftp server and log into the Asterisk box at the office. So far so good...
I call in and get into it just fine 15-20 minutes later phone goes dead and hangs up.(no it wasn't unplugged)
I call back and am joined in again. All is well for 25-30 minutes and same thing. I call back in and again join the conference call. It is just about my turn to speak and it happened again. To make this story a little shorter, it happened twice more at very random times.
The coordinator for the conference call and I were on a first name basis by the time I got done(LOL). He said I was the only person having any issues.
I was puzzled...
When I got off of the call and tried calling somebody else. No problem. three, four, and five calls later and no dropping out, no issues whatsoever. Ran some test on the network last night and it seems that the cable company is losing more packets than usual every two or three hours for about an hour... It seems scary to me that the Asterisk PBX, and the Cisco VoIP phone are so finicky with dropped packets. Time to investigate. I am hoping this is a configuration error that I have created, otherwise, this will always be a random problem that might affect parts of town like the good old days of a large phone cable being inadvertantly cut by someone digging a hole somewhere. Anyway, I have a call in to the cable broadband people and am hoping the can plug the leak...

Posted by Sean at 03:21 PM | Comments (0) | TrackBack

August 02, 2005

Why Voice over IP Is on Hold

This is an interesting take on why VOIP isn't the standard for corporate america and the adopting of the technology has been so slow.
Vincent Ryan, www.NewsFactor.com

Voice over Internet Protocol (VoIP) technology, which allows enterprises to transmit voice calls over data networks, was supposed to make copper telephone networks obsolete and save companies millions of dollars in communication costs. But those ambitious goals have yet to be realized, and vendors are trying to unravel the mystery behind the technology's slow acceptance.
Call quality, unstable technology, and the perceived oddity of migrating an enterprise's voice communications to its data network used to be the main reasons cited for VoIP's lackluster adoption rate. No more.
"You don't hear a lot of talk anymore about voice quality or QoS (quality of service)," Frost & Sullivan VoIP analyst Jon Arnold told NewsFactor. "There's enough confidence and faith in the technology. A lot of the gap has been bridged, although it's not perfect."
So, why are enterprises still hesitant to deploy VoIP on a wide scale -- and why are service providers still twiddling their thumbs?
Education Needed
The slow pace of VoIP deployment is connected not only to the economic downturn, but also to a lack of customer education or acceptance, according to Ralph Santitoro, director of network architecture at Nortel Networks (NYSE: NT - news).
To remedy that shortfall and educate enterprise customers, he said, Nortel now offers a network assessment program in which Nortel business partners "crawl through" a company's Ethernet network to determine its fitness for IP telephony.
Cisco Systems (Nasdaq: CSCO - news) also is trying to teach customers about the business benefits of voice over IP. "Customers aren't aware they can get QoS in a corporate network," said Hank Lambert, director of product marketing in Cisco's enterprise voice and video group. "When they talk about voice over IP, they confuse [it] with voice over Internet. That's a very different application of voice over IP."
Currently, most corporate applications send voice calls through a company's data network and tie in to the standard copper-wire phone system at some point, avoiding the open Internet altogether.
VPNs Lead the Field
Currently, most enterprises adopting VoIP are focusing on voice virtual private network (VPN) applications, said Bob VanSickle, vice president of the Americas at Vocaltec, the company that introduced the first PC-to-PC voice client back in 1995.
With this technology, companies with numerous remote locations can use small gateway appliances that interface with their PBX (private branch exchange -- a private telephone switching system) to place calls that bypass the phone companies' legacy networks, creating a secure voice connection. And one large retail chain is using its frame relay links, which are idle during the day, to do unlimited calling between stores, VanSickle said.
He noted that return on investment for such VoIP projects usually can be achieved in less than six months. "You're not spending money to hear a lot of garbles, clicks and buzzes," he said. "Interoffice communications are also lower risk."
Of course, VoIP is better suited to some applications than others. For example, some service providers that offer IP voice clearly state in their ads that they do not support 911 emergency services. "That might slow the adoption for certain applications," Santitoro noted. Cisco, for one, has addressed this problem by introducing an emergency responder server that gathers information about the location of an IP phone.
Static About Security
Unlike VPN-based voice over IP, the adoption curve will be steeper for IP PBX devices that allow users to place calls over the open Internet, VanSickle said. Slower acceptance of this technology will be due partly to technical problems -- packet jitter, missing and dropped packets, and compression hurdles -- and partly to security concerns.
Sending voice packets across firewalls, for example, is a potential nightmare for some IT managers. "When you packetize everything, the networks don't know how to distinguish very well," Arnold said. In other words, identifying which packets are or are not acceptable in terms of network security is a difficult task.
VoIP handsets also have their share of security vulnerabilities. Theoretically, hackers wanting to eavesdrop would not need to establish a physical connection to tap a VoIP phone line because the phone is connected to computers over the open Internet. VoIP phones also have IP addresses, which means anyone with a browser could access the phone's Web page and modify its features and options.
But tapping an IP phone call is a lot harder than it sounds, VanSickle said, because packets follow multiple geographic paths and pass through different gateways to reach their destination. "Where do you place the tap, and how do you reassemble the packets?" he said.
Integration Obstacles
Security issues aside, interoperability, or internetworking, is also key to promoting widespread adoption of VoIP over the open Internet. For example, in today's technology market, one vendor's flavor of session initiation protocol (SIP) -- a signaling protocol for Internet conferencing, telephony and instant messaging (news - web sites) -- may not be identical to another vendor's.
"Although the equipment meshes 90 percent of the time, there will always be glitches," Arnold said. "Until you can throw everything into the pot and have it all work, no carrier in their right mind is going to have a large-scale deployment."
Although these obstacles to adoption are not insurmountable, overcoming them may take time.
Inert on IP
Service providers are the final piece of the VoIP puzzle, and they have been slow to deploy the technology and QoS-level agreements, Lambert said. But there are positive signs for VoIP on the horizon: Domestic RBOCs (regional Bell operating companies, such as BellSouth (NYSE: BLS - news), Ameritech and Nynex, among others) will need VoIP technologies in order to offer local phone service to customers outside their home regions, according to Arnold.
"If RBOCs push further down the road, they're going to need IP packet technologies to do it," he noted. "They're not going to build out their copper networks. That could serve as a big stimulus" to sales of VoIP services and equipment.
Even so, the RBOCs currently are dragging their heels on VoIP trials, putting the small, innovative equipment vendors running on limited capital resources in jeopardy. "There's nothing pushing [the RBOCs]," Arnold said. "Everyone is hoping they are going to do it all."
But there is an "X" factor that could change the RBOCs' tune: Cable companies, such as Cox Communications, are rolling out IP telephony services over HFC cable networks. If cable operators begin to make inroads into homes and businesses, the RBOCs may be forced to pick up the pace.

Posted by Sean at 08:29 AM | Comments (0) | TrackBack

August 01, 2005

Cisco 7960 and VOIP

The Cisco phones are an interesting beast. I have a few around that I get to play with. Some running SIP and some running the the Cisco Call Manager thing. I like some more of the features of the Cisco Call manager but the SIP functionality seems to outweight any benifit in the long run. The most important/fun feature of either is the XML services. Currently I have a phone that reads three different XML documents from my site. One is a XML from weather.com giving me the days weather, The second one is a dynamic telephone directory (for those of us that do not like typing something twice) and most importantly a RSS reader that allows a blog to streamed over the Asterisk connected phone.
The first two just get a mention but the third is where the fun begins. It took many tries but the phone now has a selection under the globe button that links me to a rss feed. Now I know why these phones cost a few extra dollars. I am expected to release a version of the XML services code soon after a few more revisions and perfections in the formatting...

Posted by Sean at 12:51 PM | Comments (0) | TrackBack

Asterisk -In Windows Part 2

Windows for Asterisk... What a concept and in part 2 of this I am going to bear some of the inner working of what DID work for me.

This is a continuation of Asterisk - In Windows Posted Yesterday.

Sip.conf (Broadvoice and Cisco configs.)
The big change is right here!!!
The first does NOT work and what I read on their site.
;register => 303XXXXXXX@sip.broadvoice.com:itsasecret:303XXXXXXX@sip.broadvoice.com

And This one DOES work!!!
register => 303XXXXXXX:itsasecret@sip.broadvoice.com
An issue I can not seem to explain and will do more research on as it is completely different than any of my UNIX servers. And the rest below is pretty plain jane...

[sip.broadvoice.com]
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=303XXXXXXX
secret=itsasecret
username=303xxxxxxx
insecure=very
context=from-broadvoice
authname=303xxxxxxx
dtmfmode=inband
dtmf=inband
canreinvite=no
nat=yes
disallow=all
allow=gsm
allow=ulaw
allow=alaw

(The cisco 7960 and 7940 phones)
[3000]
type=friend
username=3000
secret=itsasecret
host=dynamic
callerid= <3000>
nat=no

and six more of these all identical

The extensions.conf
[from-broadvoice]
Audio problem fix as reported in my last post(not exactly a great way to fix it) causes the audio to be passed immediatly as soon as the asterisk server gets a sip notification of the call.
exten => s,1,Dial(SIP/3001&SIP/3002,60,tm)
Here is the way it should have worked.(Any insight on this would be appreciated.)
;exten => s,1,Dial(SIP/3001&SIP/3002,90,t)
exten => s,2,Voicemail(3001)
exten => s,3,hangup

As I said all is going well and the AMD machine is not working very hard at all to keep up. Now I just need to figure out what the big difference is for the audio problem... As I said. Insight from you would be greatly appreciated as this is about as odd as I have ever seen Asterisk get.

Posted by Sean at 09:58 AM | Comments (0) | TrackBack

July 31, 2005

Asterisk -In Windows

This is a followup of sorts as I venture into another fun challege/test of the VOIP system of choice for the monetarily challenged. Here is the line up

Windows XP Pro
Asterisk Win32 from asteriskwin32.com
AMD 64 3200
1 Gb DDR2 RAM
650 GB SATA storage
MSI NVIDIA 6600 PCI Express
A8N-SLI Deluxe Motherboard
- NVIDIA nForce4 SLI
- PCI Express Architecture
- SATA 3Gb/s
- Dual RAID
- Dual Gigabit LAN & AI NET2
- NV Firewall
- AI NOS™
- AI Audio (8-channel Audio)

My goal is to get all of this up and running with asterisk and see what the performance ramifications of running it under CYGwin are. I will use the majority of the configurations from the other ASterisk servers I have with the main difference being the Windows thing... I will post update this port in a few hours and explain the results...
Update 1:
Server is up and running with only a few minor issues. Issue 1 was broadvoice. The setup is not as they have explained in the help files they have on there site or any of the true unix implementations I have done. Issue 2 was the way it is handling the Cisco 7960 SIP phones. Currently, in incoming calls only, I can hear the other person when they call but they cannot here me. In outbound calls it is fine. The workaround I have found it to use a m switch in the extensions.cfg so that the incoming caller hears music instead of ringing. Audio is then passed throughout the entire call.
Overall the server seems to run very well. The calls are routed efficiently and the broadvoice service again is shining as a very easy way to establish calls to non VOIP equipment or phones.
I will post again the details of the change to the first issue. Currently I have modified the registration information for the broadvoice service to create teh connection/registration to them. Once I am sure it is solid I will post the sip.cfg file that succeeds.
So far Asterisk for windows is a success and the system can still be used for other tasks easily. Memory usage is low and will 10 VOIP to VOIP calls going on with translation to another codec the processor doesn't break a sweat. I am very pleased so far. Tomorrows post will bring the guts out of this config so that you can duplicate it at home without having to have a seperate box for it.

Posted by Sean at 05:26 PM | Comments (0) | TrackBack

July 29, 2005

SIP 101 - Session Initiation Protocol Explained

For all of us that use it but didn't understand exactly what it was doing... This is an excellent primer on SIP. Now if I could just get my hands on one of those SIP WiFi phones...

Session Initiation Protocol or SIP refers specifically to a language that various computers can communicate to one another in so that they can complete voice calls. It has become vitally important in recent years as it plays a central role in VoIP or Voice Over Internet Protocol. VoIP Is the rapidly growing technology which has millions of Americans throwing out their local and long-distance telephone bills and replacing them with free calls made over the internet.

While Session Initiation Protocol sounds like technobabble, it helps if you can imagine SIP as the common language that new generation operators use to complete calls over the internet. With SIP, however, the operators are no longer hundreds of people in a room somewhere connecting one call to another but simply your computer device connecting to the telephone or computer device of the person you want to talk to. The fact that there is no need for real operators, or even a central board to complete calls through, explains part of why SIP is so revolutionary.

SIP was intended to give ordinary callers like you and me all the familiar functions and features of what we expect from a phone call, such as a dial tone, a ringing sound, etc. So while all the communication from our end seemed exactly the same as before, SIP makes phone calls by communicating directly with the other person’s telephone device. Unlike traditional telephony, which was based on a cog and wheel approach in which the call you placed goes through a central location and then is routed to the person you are trying to call, SIP is based on internet protocol. This means that there is no need for a central cog to run calls through, but rather calls can be made directly from person to person.

The fact that the technology is based on internet protocol (IP) rather than a traditional cog and wheel also means that placing and receiving calls are no longer inhibited by location. To conceive of this more easily it is best to think of something like your e-mail. You can take your laptop and access your e-mail from your home, just as easily as you can plug that laptop into the internet at access your e-mail from anywhere. In the same way, you can plug your SIP phone into any access point in the internet and call a person who can be located anywhere in the world. Similarly you can receive phone calls from anywhere in the world no matter where you are, simply be plugging your SIP phone into the internet.

As you can probably imagine, this ability has some pretty remarkable applications. For companies or businessmen that work out of the office, moving your office phone number is as simple as picking up the phone and carrying it with you. There is nothing else to it. For those often staying in hotels for travel or business, this means always having the direct office line with you wherever and whenever you want. No need to forward calls or even to ever pay for long distance or hotel phone access again!

The SIP technology is already revolutionizing the way in which humans communicate. In recent years, literally millions of Americans have tossed aside their traditional land based phone lines and opted for the freedom and cheapness of VoIP. Empowered with SIP technology it is uniquely able to provide you with virtually free calling, anywhere in the world, anytime in the world, without having to forward calls, change your number, or rely on others to check important messages. It is truly a technology for the future of business as well as the future of communication.

Posted by Sean at 08:14 AM | Comments (0) | TrackBack

July 28, 2005

Broadvoice & Asterisk Exceeding my VOIP needs

VOIP is starting to change my world. With Asterisk at the helm and a Cisco 7940 at the house and a 7960 on my desk at work at between my wife and myself we now have a Bat phone of sorts... She pickes it up and hits a key and mine rings...I pick up the phone and push a button and I reach her. Then if either of us need to call long distance or international... It is as simple as just dialing the coutry code and the number...Voila. And I heard that Broadvoice was a bummer of a service. I have had to call them a few times and that has been a pain, however, I have found a secret number that gets an answer every time! And the person on the other end has completely diagnosed/solved my problems. As far as I am concerned Broadvoice is much better than most of the other providers in that they have there BYOD policy which meets the needs of the advanced techno-geeks out there... I am included in that bunch. Anyway, So far Broadvoice will continue to be my provider....

Posted by Sean at 01:58 PM | Comments (1) | TrackBack

Empirix and Digium (Asterisk) Partner

I found this interesting... VOIP a sign of the times...

Empirix and Digium (of Asterisk fame) have announced that Empirix has been named as "a premier participant in the Digium Partner Program", as well as Digium's sole partner for VoIP testing. Digium, Inc., the leader in open source PBX solutions, utilizes the Empirix Hammer FX and Hammer Call Analyzer to test its current open source products, including the Asterisk Business Edition product.

"We're delighted to have Empirix as our test system partner. Hammer systems are central to the comprehensive test program that ensures Asterisk's reliability, performance, and interoperability with key hardware, software, and protocols, "said Jim Webster, director of software technologies for Digium. "We use Hammer to test Digium hardware for full compatibility with Asterisk Business Edition, as well as several select models of our open source VoIP devices. Our test bed systems are also subjected to extreme stress conditions, using Hammer test equipment, to simulate hundreds of complex VoIP calls in various real-world combinations and configurations."

"Empirix has worked closely with Digium to develop and implement test methodologies that meet the challenges of open source system development," said Duane Sword, vice president of product marketing for Empirix. "We look forward to extending that relationship through the Digium Partner Program, and to working with Digium on future product testing."

Posted by Sean at 01:42 PM | Comments (0)

VOIP

Another successful day of VOIP. Using my Cisco 7960 G phone at my house with a server located at work I am able to use four digit extensions through two firewalls and I am not having any problems. I expected this to be a huge hurdle and yet I have managed to pass through it without much fuss. The configuration is pretty basic. I am using Broadvoice as a carrier for all long distance and past my internal netowrk phone calls and a windows XP machine for the internal calls. Asterisk seems to shine here. The configuration was easy to implement.
Sip.cfg
added the broadvoice config
Exten.cfg
Added the dialing plan for the internal and external calling
and voila....
I have been calling many friends that otherwise I would have been paying LD for. I have even been able to call a friend in Ireland for next to nothing.... More to come...

Posted by Sean at 05:00 AM | Comments (0)

Asterisk Up and running

Well, I have gotten asterisk up and running... Made my first VOIP international call today! Incredible how easy it was to do. I downloaded the software and set it up with 2 cisco 7960 and and 2 cisco 7940 voip phones. It was almost too easy, it was really that simple. There is more to come as I figure out the config details. Summary of configuration:
4 Cisco VOIP phones
1 Asterisk server running Windows Media center edition
1 broadband voip provider
configured over 4 states and so far 200+ test calls made...
Testing is going great.
More to come...See tomorrow.

Posted by Sean at 02:10 AM | Comments (0)

July 27, 2005

VOIP -Explained Part 3

You pick up the receiver, which sends a signal to the ATA.
The ATA receives the signal and sends a dial tone. This lets you know that you have a connection to the Internet.
You dial the phone number of the party you wish to talk to. The tones are converted by the ATA into digital data and temporarily stored.
VoIP Terms
The central call processor is a piece of hardware running a specialized database/mapping program called a soft switch. See the "Soft Switches" section to learn more.
The phone number data is sent in the form of a request to your VoIP company's call processor. The call processor checks it to ensure that it is in a valid format.
The call processor determines to whom to map the phone number. In mapping, the phone number is translated to an IP address (more on this later). The soft switch connects the two devices on either end of the call. On the other end, a signal is sent to your friend's ATA, telling it to ask the connected phone to ring.
Once your friend picks up the phone, a session is established between your computer and your friend's computer. This means that each system knows to expect packets of data from the other system. In the middle, the normal Internet infrastructure handles the call as if it were e-mail or a Web page. Each system must use the same protocol to communicate. The systems implement two channels, one for each direction, as part of the session.
You talk for a period of time. During the conversation, your system and your friend's system transmit packets back and forth when there is data to be sent. The ATAs at each end translate these packets as they are received and convert them to the analog audio signal that you hear. Your ATA also keeps the circuit open between itself and your analog phone while it forwards packets to and from the IP host at the other end.
You finish talking and hang up the receiver.
When you hang up, the circuit is closed between your phone and the ATA.
The ATA sends a signal to the soft switch connecting the call, terminating the session. Probably one of the most compelling advantages of packet switching is that data networks already understand the technology. By migrating to this technology, telephone networks immediately gain the ability to communicate the way computers do.
It will still be at least a decade before communications companies can make the full switch over to VoIP. As with all emerging technologies, there are certain hurdles that have to be overcome.

Posted by Sean at 03:58 AM | Comments (0)

VOIP and E911

I just got an email from Broadvoice stating that I was NOT to call 911 from any of my VOIP phones useing there service. Seems like they are having a little trouble getting their E911 working. Anyway it brought me to this read...

The FCC announced a 30 day extension of its e911 rules. The FCC has stated that it will not initiate any action until August 30, 2005 against any provider of VoIP service. (This pertains to the requirement that the provider obtain acknowledgement by every customer.) The email states that it is conditional that the VoIP provider file a detailed report with the Commission by August 10, 2005.

Posted by Sean at 03:52 AM | Comments (0) | TrackBack

July 26, 2005

VOIP -Explained Part 2

Try it Yourself
If you're interested in trying VoIP, then you should check out some of the free VoIP software available on the Net. You should be able to download and set it up in about three to five minutes. Get a friend to download the software, too, and you can start tinkering with VoIP to get a feel for how it works. One place to look is http://www.skype.com.

But chances are good you are already making VoIP calls any time you place a long-distance call. Phone companies use VoIP to streamline their networks. By routing thousands of phone calls through a circuit switch and into an IP gateway, they can seriously reduce the bandwidth they're using for the long haul. Once the call is received by a gateway on the other side of the call, it is decompressed, reassembled and routed to a local circuit switch.

Although it will take some time, you can be sure that eventually all of the current circuit-switched networks will be replaced with packet-switching technology (more on packet switching and circuit switching later). IP telephony just makes sense, in terms of both economics and infrastructure requirements. More and more businesses are installing VoIP systems, and the technology will continue to grow in popularity as it makes its way into our homes.

VoIP Features
The Forrester Research Group predicts that nearly 5 million U.S. households will have VoIP phone service by the end of 2006. Perhaps the biggest draws to VoIP for the home users that are making the switch are price and flexibility.

Flexibility
With VoIP, you can make a call from anywhere you have broadband connectivity. Since the IP phones or ATAs broadcast their info over the Internet, they can be administered by the provider anywhere there is a connection. So business travelers can take their phones or ATAs with them on trips and always have access to their home phone. Another alternative is the softphone. A softphone is client software that loads the VoIP service onto your desktop or laptop. The Vonage softphone has an interface on your screen that looks like a traditional telephone. As long as you have a headset/microphone, you can place calls from your laptop anywhere in the broadband-connected world.

Price Most VoIP companies are offering minute-rate plans structured like cell phone bills for as little as $30 per month. On the higher end, some offer unlimited plans for $79. With the elimination of unregulated charges and the suite of free features that are included with these plans, it can be quite a savings.

Most VoIP companies provide the features that normal phone companies charge extra for when they are added to your service plan. VoIP includes:
Caller ID
Call waiting
Call transfer
Repeat dial
Return call
Three-way calling

There are also advanced call-filtering options available from some carriers. These features use caller ID information to allow you make a choice about how calls from a particular number are handled. You can:
Forward the call to a particular number
Send the call directly to voicemail
Give the caller a busy signal
Play a "not-in-service" message
Send the caller to a funny rejection hotline

With many VoIP services, you can also check voicemail via the Web or attach messages to an e-mail that is sent to your computer or handheld. Not all VoIP services offer all of the features above. Prices and services vary, so if you're interested, it's best to do a little shopping. Now that we've looked at VoIP in a general sense, let's look more closely at the components that make the system work. In order to understand how VoIP really works and why it's an improvement over the traditional phone system, it helps to first understand how a traditional phone system works.

The VoIP Phone System: Packet Switching
Data networks do not use circuit switching. Your Internet connection would be a lot slower if it maintained a constant connection to the Web page you were viewing at any given time. Instead, data networks simply send and retrieve data as you need it. And, instead of routing the data over a dedicated line, the data packets flow through a chaotic network along thousands of possible paths. This is called packet switching. While circuit switching keeps the connection open and constant, packet switching opens a brief connection -- just long enough to send a small chunk of data, called a packet, from one system to another. It works like this:
The sending computer chops data into small packets, with an address on each one telling the network devices where to send them.
Inside of each packet is a payload. The payload is a piece of the e-mail, a music file or whatever type of file is being transmitted inside the packet.
The sending computer sends the packet to a nearby router and forgets about it. The nearby router send the packet to another router that is closer to the recipient computer. That router sends the packet along to another, even closer router, and so on.
When the receiving computer finally gets the packets (which may have all taken completely different paths to get there), it uses instructions contained within the packets to reassemble the data into its original state.
Packet switching is very efficient. It lets the network route the packets along the least congested and cheapest lines. It also frees up the two computers communicating with each other so that they can accept information from other computers, as well.

The Advantage
VoIP technology uses the Internet's packet-switching capabilities to provide phone service. VoIP has several advantages over circuit switching. For example, packet switching allows several telephone calls to occupy the amount of space occupied by only one in a circuit-switched network. Using PSTN, that 10-minute phone call we talked about earlier consumed 10 full minutes of transmission time at a cost of 128 Kbps. With VoIP, that same call may have occupied only 3.5 minutes of transmission time at a cost of 64 Kbps, leaving another 64 Kbps free for that 3.5 minutes, plus an additional 128 Kbps for the remaining 6.5 minutes. Based on this simple estimate, another three or four calls could easily fit into the space used by a single call under the conventional system. And this example doesn't even factor in the use of data compression, which further reduces the size of each call.
Let's say that you and your friend both have service through a VoIP provider. You both have your analog phones hooked up to the service-provided ATAs. Let's take another look at that typical telephone call, but this time using VoIP over a packet-switched network:

Posted by Sean at 03:58 AM | Comments (0)

July 25, 2005

VOIP Explained -Part 1

If you've never heard of VoIP, get ready to change the way you think about long-distance phone calls. VoIP, or Voice over Internet Protocol, is a method for taking analog audio signals, like the kind you hear when you talk on the phone, and turning them into digital data that can be transmitted over the Internet.
How is this useful? VoIP can turn a standard Internet connection into a way to place free phone calls. The practical upshot of this is that by using some of the free VoIP software that is available to make Internet phone calls, you are bypassing the phone company (and its charges) entirely.

VoIP is a revolutionary technology that has the potential to completely rework the world's phone systems. VoIP providers like Vonage have already been around for a little while and are growing steadily. Major carriers like AT&T are already setting up VoIP calling plans in several markets around the United States, and the FCC is looking seriously at the potential ramifications of VoIP service.

Above all else, VoIP is basically a clever "reinvention of the wheel." In this article, HowStuffWorks will show you the principles behind VoIP, its applications and the potential of this emerging technology, which will more than likely one day replace the traditional phone system entirely.

The interesting thing about VoIP is that there is not just one way to place a call. There are three different "flavors" of VoIP service in common use today: ATA - The simplest and most common way is through the use of a device called an ATA (analog telephone adaptor). The ATA allows you to connect a standard phone to your computer or your Internet connection for use with VoIP. The ATA is an analog-to-digital converter. It takes the analog signal from your traditional phone and converts it into digital data for transmission over the Internet. Providers like Vonage and AT&T CallVantage are bundling ATAs free with their service. You simply crack the ATA out of the box, plug the cable from your phone that would normally go in the wall socket into the ATA, and you're ready to make VoIP calls. Some ATAs may ship with additional software that is loaded onto the host computer to configure it; but in any case, it is a very straightforward setup.

IP Phones - These specialized phones look just like normal phones with a handset, cradle and buttons. But instead of having the standard RJ-11 phone connectors, IP phones have an RJ-45 Ethernet connector. IP phones connect directly to your router and have all the hardware and software necessary right onboard to handle the IP call. Soon, Wi-Fi IP phones will be available, allowing subscribing callers to make VoIP calls from any Wi-Fi hot spot.

Computer-to-computer - This is certainly the easiest way to use VoIP. You don't even have to pay for long-distance calls. There are several companies offering free or very low-cost software that you can use for this type of VoIP. All you need is the software, a microphone, speakers, a sound card and an Internet connection, preferably a fast one like you would get through a cable or DSL modem. Except for your normal monthly ISP fee, there is usually no charge for computer-to-computer calls, no matter the distance.

Part 2 Coming soon

Posted by Sean at 03:56 AM | Comments (0)

July 10, 2005

Asterisk for Windows? Absolutely

Now the playing field has truly been leveled... I found this to be a good read:

Microsoft Windows™ users who want to deploy the popular Asterisk open source PBX system finally have a solution — sort of.

N2Net, a provider of mission critical communication hosting, announce the immediate availability of AstWind, a package allowing users of several Windows platform to run Asterisk, but only as a Voice over IP application.

"AstWind allows Windows users to safely install and test the Asterisk PBX with the click of a button." said Gregory Boehnlein, Vice President of N2Net. "Prospective users can test the software and become familiar with it before deploying a full Linux system. Developers can maintain multiple environments for testing and PBX Administrators can create disaster recovery plans to backup existing installations."

Most VoIP only features are available on the Windows version, although there is currently no support for any hardware interfaces. The AstWind package contains a full development environment and the Asterisk PBX pre-installed on a Debian GNU Linux filesystem.

Through the coLinux networking subsystem, Asterisk has access to the network and can participate fully as a unique server.

Cooperative Linux (short-named coLinux) is the first working free and open source method for optimally running Linux on Microsoft Windows natively. It is a port of the Linux kernel that allows it to run cooperatively alongside another operating system on a single machine. For instance, it allows one to freely run Linux on Windows 2000/XP, without using a commercial PC virtualization software such as VMware, in a way which is much more optimal than using any general purpose PC virtualization software.

"The creation of AstWind was remarkably easy, thanks to the amazing work of the coLinux project which allowed us to setup a Debian GNU Linux environment on Windows with minimal overhead," Boehnlein said. "The entire process took less than an hour. For obvious reasons, I do not advocate using AstWind for mission critical installations, but it does open some interesting doors for hobbyists and developers. With AstWind, users can experiment with the Asterisk PBX in a no-risk manner."

The AstWind installer can be downloaded as a single executable file from N2Net's high performance FTP cluster located at ftp://ftp.nacs.net/asterisk/astwind.

Posted by Sean at 01:29 PM | Comments (0) | TrackBack

October 14, 2004

VOIP Testing

Here's some news about a VoIP testing tool. VoIP testing tools are becoming more important as VoIP grows, so enjoy this bit of news, I did:

Long Beach, CA - October 12, 2004
Psytechnics, the global leader in voice and video quality assessment software, has teamed with NetTest, a leading worldwide provider of monitoring, management and testing solutions for advanced and converged networks, to bring industry standard voice quality measurement capabilities to NetTest's service assurance tools for VoIP networks.

NetTest will integrate Psytechnics' psyVoIP software for measuring the voice quality of live customer calls into the MasterQuest VoIP solution. The MasterQuest VoIP solution is part of NetTest's OSS solution family based on MasterQuest, designed specifically for telecommunication network carriers and service providers to optimize their business performance and increase revenues

MasterQuest VoIP is a complete service assurance solution for IP networks, combining troubleshooting with key performance indicators to support the management of suppliers' service level agreements. The incorporation of psyVoIP will enable not only real-time assessment of the end-user experience, but also rapid diagnosis of customer-affecting network problems, allowing efficient prioritization of network maintenance.

"As demand for VoIP grows, managing voice quality and understanding end-user experience is a top concern among service providers. The decision to integrate real-time voice quality monitoring and diagnostic tools into our VoIP solution was obvious, and Psytechnics is second to none when it comes to voice quality," said Mr. Weber, vice president and head of the Systems Business Unit at NetTest. "Psytechnics has a proven track record, having developed the ITU standard for voice quality assessment, and its technology is a perfect fit for the current market demand."

"When monitoring the quality of a VoIP network, service providers need both a consolidated view of overall network performance and constant awareness of individual subscriber experience," said John Winchester, CEO of Psytechnics "The pairing of psyVoIP and MasterQuest allows NetTest to offer a complete monitoring and service assurance system that unifies a network and service-centric reporting, with customer-centric reporting."

psyVoIP is a passive voice quality measurement solution that monitors live customer calls and rates the perceived quality on a MOS (mean opinion score) scale of 1 to 5, with 5 being the highest. In addition to quality metrics, psyVoIP provides a range of diagnostics information so customer-affecting problems are identified and rectified as quickly as possible. psyVoIP's voice quality assessment capabilities are based on Psytechnics' PESQ, the ITU-T P.862 standard for objective speech quality assessment. Psytechnics is also involved in the current development of an ITU standard (P.VQT) for measuring quality in VoIP networks.

For additional information about NetTest and MasterQuest, please visit www.nettest.com. For further details about Psytechnics and psyVoIP, please visit www.psytechnics.com.

Posted by Sean at 06:23 PM | Comments (0) | TrackBack

April 26, 2004

Testing shows VoIP a big winner

I found this to be a great article as well as a lauching pad for my own investigation

By Joel Snyder, Network World, 04/26/04

I've just spent the last two weeks doing interoperability testing of VoIP equipment for NetWorld+Interop. You can get the full results next month in Las Vegas at the show or in the May 10 issue of Network World, but here are some quick observations to whet your appetite:

H.323 is dead. Oh, man, is it dead! In past years, we've struggled to get H.323 devices to interoperate. They don't do it well and, what's worse, debugging is a total pain. Not so with Session Initiation Protocol (SIP)-controlled telephony. We had incredibly good basic interoperability in just minutes between SIP phones. When we wanted to debug problems, having all the control messages show up in plain text made it easier than H.323 ever was. For debugging, we used ClearSight Networks' VoIP analyzer as well as WildPackets' EtherPeek NX, but rarely needed the power and advanced features of either tool.

If you want to do a single-vendor VoIP telephony deployment, you don't care what protocol is underneath it all. Go ahead and get whatever makes sense from Avaya, Cisco, Nortel or your favorite vendor. But if you want to go for massive interoperability, mixing and matching vendors, phones and equipment, then SIP is the only way to go.

Not all phones are created equal. In the network world, we've become accustomed to treating equipment as a commodity. You can argue the fine points forever, but when it comes to most companies, it often doesn't matter what brand of network interface card, switch or server you buy. Not so with phones. We saw tremendous difference in the voice quality and performance across different products. Managing jitter across the network, encoding and decoding speech, and just sounding good or lousy - devices were all across the map. There is definitely a human factor involved with phones that's going to be unfamiliar to most IT people.

We found lots of variance in configurability and flexibility. You can tell the maturity of a product by how many knobs it has on it. Newly released, bargain-basement devices let you get on the network and little else. The battle-scarred veterans have 50 or 100 settings to tune the device for optimum performance in your network. The nice thing about SIP was that even without tuning, we had great interoperability results.

Getting started is easy. You might not want to run your company on a piece of freeware that you drop on a Linux box (or maybe you do), but you can sure get started with SIP that way. I've struggled to make open source and commercial H.323 work, but I've never seen anything as easy as the open source product SIP Express Router from iptel.org, with Digium's Asterisk a close second. Our team had both products installed and routing calls before lunchtime. Throw in a couple of inexpensive phones, such as the $75 Grandstream BudgeTone, and you're doing basic IP telephony for almost nothing.

So what are you waiting for? It's time to learn about IP telephony!

Posted by Sean at 11:35 PM | Comments (0)