Cheap voip call to India can use with intervoip as of date is the cheapest provider of international call to India. intervoip.com provides the rate at about 1.5 cents (with VAT, 1.7 cents) per minute on InterVOIP. This is by far, to the best of advicevoip knowledge is the cheapest international calling rate to India.
Explained in previous postsm, you can also make voip calls via Access Numbers although these numbers are only available in about 9 countries which doesn’t include USA and Canada. Intervoip offers PC to Phone and Phone to Phone connections and fully support SIP. You can either use Mobile VOIP, SIP Phone, VOIP ATA or IP phone.
Rebtel voip phone company
Cheap international calls from your mobile phone
Not computers, downloads, or phone cards necessary
Just dial and talk internationally at local rates.
Online status of contacts
No SkypeIn required
No PC required
Free VoIP call to india trial period!
Voip Provider Opinion:
Rebtel probably works off voip servers thus no need for voip dialers to be installed. No download and installation means no adware or spyware risk. Thus no requirement of a PC when you are on the move. You can even use your mobile phone. Rates are free for most countries. Simply no negatives not to try. Definitely recommended.
USA: 1.8 c/min
UK: 1.9 c/min
India: 5.6 c/min
Poland: 1.9 c/min
China: 1.9 c/min
Enjoy to VoIP Phone Calls
Saturday, June 28, 2008
World Phone VOIP Service VoIP calls in India
VoIP calls in India,World Phone Internet Services provides Internet bandwidth, Internet Telephony (VoIP - Voice over Internet Protocol) and its related services to the corporate and domestic users in India.
World Phone account provides you with complete portability. SO you can access World Phone from anywhere in the world using World Phone’s softphone or a phone adapter. To access World Phone’s softphone, log into your account and launch webphone. The softphone automatically installs, which runs only on Windows platform. TO use the softphone, your computer must be running in order to receive calls. If your computer is not on or you are unavailable, the call will be directed to your Voicemail.
Use a phone adapter, just plug it into any high-speed Internet connection and place calls. If you are using a device that works without a computer such as an IP phone or a telephone adapter, calls will be directed to that device; unanswered calls will be directed to your Voicemail.
World Phone service you can call any phone number that you can call with a traditional phone. Calls to regular phones are charged on a per minute basis while those to other World Phone service users are free.
Individuals and corporations with moderate to large long distance or international call usage can enjoy significant savings, depending on call volume up to 50%.
World Phone provides calls to USA, Canada, UK, Australia, Austria, Belgium, China, Chile, Denmark, France, Germany, Hong Kong, Italy, Israel, Netherlands, Norway, Poland, Singapore, Spain, Sweden, Switzerland, Taiwan and Thailand for less than one Rupee per minute. World Phone also provides unlimited calls to some selected countries for approximately 1500 rupees per month, useful for software companies and call centers.
The World Phone service routes your incoming and outgoing voice calls right alongside the data being sent to or from your computer. Thus, you can make and receive calls while you use your computer to access the Internet. World Phone uses advanced audio compression techniques to minimize the data traffic caused by voice calls and maximize the bandwidth available for your other Internet traffic.
If you have a router with firewall installed to connect to the Internet or if you using the service in a corporate environment, World Phone provides a NAT traversal feature that allows you to use the softphone in such environments.
How you think about VoIP with World Phone Internet Services provides?
World Phone account provides you with complete portability. SO you can access World Phone from anywhere in the world using World Phone’s softphone or a phone adapter. To access World Phone’s softphone, log into your account and launch webphone. The softphone automatically installs, which runs only on Windows platform. TO use the softphone, your computer must be running in order to receive calls. If your computer is not on or you are unavailable, the call will be directed to your Voicemail.
Use a phone adapter, just plug it into any high-speed Internet connection and place calls. If you are using a device that works without a computer such as an IP phone or a telephone adapter, calls will be directed to that device; unanswered calls will be directed to your Voicemail.
World Phone service you can call any phone number that you can call with a traditional phone. Calls to regular phones are charged on a per minute basis while those to other World Phone service users are free.
Individuals and corporations with moderate to large long distance or international call usage can enjoy significant savings, depending on call volume up to 50%.
World Phone provides calls to USA, Canada, UK, Australia, Austria, Belgium, China, Chile, Denmark, France, Germany, Hong Kong, Italy, Israel, Netherlands, Norway, Poland, Singapore, Spain, Sweden, Switzerland, Taiwan and Thailand for less than one Rupee per minute. World Phone also provides unlimited calls to some selected countries for approximately 1500 rupees per month, useful for software companies and call centers.
The World Phone service routes your incoming and outgoing voice calls right alongside the data being sent to or from your computer. Thus, you can make and receive calls while you use your computer to access the Internet. World Phone uses advanced audio compression techniques to minimize the data traffic caused by voice calls and maximize the bandwidth available for your other Internet traffic.
If you have a router with firewall installed to connect to the Internet or if you using the service in a corporate environment, World Phone provides a NAT traversal feature that allows you to use the softphone in such environments.
How you think about VoIP with World Phone Internet Services provides?
Linksys SPA2100 ATA Review
Linksys SPA2100 ATA like to be good job.I did a lot of research on which one to buy and decided on a Linksys/Sipura model as they seemed to be reliable and the majority of people had positive things to say about them. I chose the Linksys SPA2100 for two reasons: A) because it had a WAN and a LAN port, which meant simply putting it between my DSL router and my PC, and B) because it had two FXS (phone) ports, so I could set one up with a permanent service (I use Callcentric, who have been very helpful in my learning phase) and then use the second one to play with, changing settings and providers to my hearts content while knowing I could still make and receive calls on the first port (as it turned out, the novelty of that wore off and I always keep Voipuser on the second one).
The Feature Set Clinched It
One other thing that the Linksys SPA2100 had in its favour was the QoS bit – it can tell the router it wants a higher priority than less important stuff, such as email, so the speech quality is not affected unnecessarily. I would have thought that this would have been a primary requirement for any ATA, however there are a lot of them (including some SPAs) that do not have QoS.
When my Linksys SPA2100 first arrived, I was pleased to see it even included a cable to connect to the DSL router, so it was ready to plug in and play with, which I did.
In Use
I managed to get the thing working adequately with Callcentric in a very short time, however it took weeks to get it to function properly with Voipuser (one way speech, then calls to the 0844 number telling me I was unavailable, despite the web site telling me I was logged in).
My problems were partly caused by the Linksys SPA2100 being so configurable that I didn’t know what I was doing, and partly because I am a male and therefore played around and didn’t bother reading instructions. Fortunately for me, someone else must have previously experienced the same issues and they shared their findings (I still haven’t read the Linksys SPA2100’s manual, although I did get as far as downloading it in a moment of despair).
Now that I have a fully functioning Linksys SPA2100 I am thrilled with it. The speech quality, whilst not being as good as PSTN calls, exceeded my expectations. The codec used obviously makes a great deal of difference to speech quality, although more often than not it uses the G711u codec, despite me telling it that G729a is my preferred one (G711u has got to be the best quality one it has, but I hate to think how much bandwidth it eats).
The echo cancellation seems to be as good as the ones the used in the PSTN international gateways here, and the only time I have experienced echo was when the far end user tried to put me on speakerphone (which I hate anyway).
Advanced Features
Having got the speech basics working, I set about making the Linksys SPA2100 do other things I consider important, namely making itself appear to fit in with my telecommunications ideals. To this end, I taught it how to make proper tones (it defaults to US tones – ewwww) and proper ringing cadences. This is where the ultra configurability that initially caused me problems now became my friend – my SPA now goes “ring ring” when someone calls me. It was worth teaching it the tones too, because although Voipuser seems to pass the ringing tones from the B party back to me (I like that), Callcentric does not; the ringing tone I get when making a Callcentric call actually comes from the Linksys SPA2100 (this is not the SPA's fault) and the speech path is only established once the B party answers (giving a short delay between losing ringing tone and being connected to the callee).
Other PSTN-like things the Linksys SPA2100 can do include CPC (sometimes called Clear Forwarding) and Answer Reversal, which means it can interface with analogue PABXs and answerphones and work as you would expect. The dialing plan is user configurable and can do useful things such as number translation, which is handy if you don’t want to keep dialing **275*448 in front of a Voipuser number when calling from other VSPs.
I like my Linksys SPA2100 so much that not only would I recommend it to people, but I also bought a second one as a gift for my father. Having learnt the hard way, I first configured it via the link I mentioned above, then entered my own account details into it (hey, I had to make sure it was working before I gave it away!), made test VoIP calls (in and out), made PSTN calls in and out, and it worked. Completely set up in less than an hour, including changing the tones and ringing.
The Linksys SPA2100 was also very cost effective at $NZ138 (approx £46).
Conclusion
I have worked in the telecommunications field for over two decades (including a few years in international) and I have high expectations from a phone. The Linksys SPA2100 gave me more features than I expected at a higher quality than I expected. There is nothing that I feel is missing from it, although some people might like an FXO (central office) port, but then those people would not have bought this model. I specifically did not want an FXO port as I like to consciously choose whether to make a VoIP call or a PSTN call – I don’t want to make a call assuming it’s free only to find out it’s been routed via the PSTN at exorbitant rates. Also, I have it running through a small PABX, so the only difference to me is which access code I dial (one for each of my PSTN line, SPA port 1, and SPA port2).
Enjoy Linksys SPA2100 ATA with VoIP !
The Feature Set Clinched It
One other thing that the Linksys SPA2100 had in its favour was the QoS bit – it can tell the router it wants a higher priority than less important stuff, such as email, so the speech quality is not affected unnecessarily. I would have thought that this would have been a primary requirement for any ATA, however there are a lot of them (including some SPAs) that do not have QoS.
When my Linksys SPA2100 first arrived, I was pleased to see it even included a cable to connect to the DSL router, so it was ready to plug in and play with, which I did.
In Use
I managed to get the thing working adequately with Callcentric in a very short time, however it took weeks to get it to function properly with Voipuser (one way speech, then calls to the 0844 number telling me I was unavailable, despite the web site telling me I was logged in).
My problems were partly caused by the Linksys SPA2100 being so configurable that I didn’t know what I was doing, and partly because I am a male and therefore played around and didn’t bother reading instructions. Fortunately for me, someone else must have previously experienced the same issues and they shared their findings (I still haven’t read the Linksys SPA2100’s manual, although I did get as far as downloading it in a moment of despair).
Now that I have a fully functioning Linksys SPA2100 I am thrilled with it. The speech quality, whilst not being as good as PSTN calls, exceeded my expectations. The codec used obviously makes a great deal of difference to speech quality, although more often than not it uses the G711u codec, despite me telling it that G729a is my preferred one (G711u has got to be the best quality one it has, but I hate to think how much bandwidth it eats).
The echo cancellation seems to be as good as the ones the used in the PSTN international gateways here, and the only time I have experienced echo was when the far end user tried to put me on speakerphone (which I hate anyway).
Advanced Features
Having got the speech basics working, I set about making the Linksys SPA2100 do other things I consider important, namely making itself appear to fit in with my telecommunications ideals. To this end, I taught it how to make proper tones (it defaults to US tones – ewwww) and proper ringing cadences. This is where the ultra configurability that initially caused me problems now became my friend – my SPA now goes “ring ring” when someone calls me. It was worth teaching it the tones too, because although Voipuser seems to pass the ringing tones from the B party back to me (I like that), Callcentric does not; the ringing tone I get when making a Callcentric call actually comes from the Linksys SPA2100 (this is not the SPA's fault) and the speech path is only established once the B party answers (giving a short delay between losing ringing tone and being connected to the callee).
Other PSTN-like things the Linksys SPA2100 can do include CPC (sometimes called Clear Forwarding) and Answer Reversal, which means it can interface with analogue PABXs and answerphones and work as you would expect. The dialing plan is user configurable and can do useful things such as number translation, which is handy if you don’t want to keep dialing **275*448 in front of a Voipuser number when calling from other VSPs.
I like my Linksys SPA2100 so much that not only would I recommend it to people, but I also bought a second one as a gift for my father. Having learnt the hard way, I first configured it via the link I mentioned above, then entered my own account details into it (hey, I had to make sure it was working before I gave it away!), made test VoIP calls (in and out), made PSTN calls in and out, and it worked. Completely set up in less than an hour, including changing the tones and ringing.
The Linksys SPA2100 was also very cost effective at $NZ138 (approx £46).
Conclusion
I have worked in the telecommunications field for over two decades (including a few years in international) and I have high expectations from a phone. The Linksys SPA2100 gave me more features than I expected at a higher quality than I expected. There is nothing that I feel is missing from it, although some people might like an FXO (central office) port, but then those people would not have bought this model. I specifically did not want an FXO port as I like to consciously choose whether to make a VoIP call or a PSTN call – I don’t want to make a call assuming it’s free only to find out it’s been routed via the PSTN at exorbitant rates. Also, I have it running through a small PABX, so the only difference to me is which access code I dial (one for each of my PSTN line, SPA port 1, and SPA port2).
Enjoy Linksys SPA2100 ATA with VoIP !
Friday, June 27, 2008
Linksys SPA-921 Review
Linksys SPA-921 is cheapest model from Linksys after the Linksys SPA-90.
How good about Linksys SPA-921.
Introduction
The Linksys SPA-921 built by Sipura/Linksys is the second cheapest model from Linksys after the Linksys SPA-901, which doesn't have an LCD screen. The SPA-921 currently sells for $110 at www.telephonyware.com from which I bought it.
Installation
Hardware-wise Linksys SPA-921 , it's pretty simple: Just plug the network cable into a switch/hub connected to the Net, and plug the 5V transformer.
Software-wise, it's as rich/complicated as the 3102 VoIP gateway, which means multiple web pages that only make sense if you're already used to VoIP (and even then.) Hit the Admin Guide PDF for more information.
Basically, you need to create a new VoIP account, enter login/password and the server IP address, add a STUN server if you're behind a NAT router, and you should get a dialtone.
Usage
Like other phones, I think it has too many buttons, but it's simple enough for basic usage.
Besides a few buttons that are set to only one feature (Voicemail, Menu, Hold, Mute, Headset, Speakerphone, Volume), it relies on four buttons under the LCD screen along with a four-directional round button to navigate through menus. As an alternative, you can configure the phone through its embedded web interface.
Conclusion
It's a nice, affordable, entry-level phone.
There are just a couple of things I don't like:
- Contrary to what the documentation says (at least, as of May 2007), it cannot download a phonebook from a server
- Like the Linksys SPA-921, the phonebook only holds 100 entries
- The LCD screen cannot be raised, hence the need for the desk stand that comes with the unit. The GrandStream GXP-2000 is better in this respect
- The screen is a bit small. The GXP-2000 is also better for this
- Unlike the SPA9x2 phones, the interface is only available in English
How good about Linksys SPA-921.
Introduction
The Linksys SPA-921 built by Sipura/Linksys is the second cheapest model from Linksys after the Linksys SPA-901, which doesn't have an LCD screen. The SPA-921 currently sells for $110 at www.telephonyware.com from which I bought it.
Installation
Hardware-wise Linksys SPA-921 , it's pretty simple: Just plug the network cable into a switch/hub connected to the Net, and plug the 5V transformer.
Software-wise, it's as rich/complicated as the 3102 VoIP gateway, which means multiple web pages that only make sense if you're already used to VoIP (and even then.) Hit the Admin Guide PDF for more information.
Basically, you need to create a new VoIP account, enter login/password and the server IP address, add a STUN server if you're behind a NAT router, and you should get a dialtone.
Usage
Like other phones, I think it has too many buttons, but it's simple enough for basic usage.
Besides a few buttons that are set to only one feature (Voicemail, Menu, Hold, Mute, Headset, Speakerphone, Volume), it relies on four buttons under the LCD screen along with a four-directional round button to navigate through menus. As an alternative, you can configure the phone through its embedded web interface.
Conclusion
It's a nice, affordable, entry-level phone.
There are just a couple of things I don't like:
- Contrary to what the documentation says (at least, as of May 2007), it cannot download a phonebook from a server
- Like the Linksys SPA-921, the phonebook only holds 100 entries
- The LCD screen cannot be raised, hence the need for the desk stand that comes with the unit. The GrandStream GXP-2000 is better in this respect
- The screen is a bit small. The GXP-2000 is also better for this
- Unlike the SPA9x2 phones, the interface is only available in English
Thursday, June 26, 2008
VoIP calls to Pakistan for only $0.298 per minute!-Call Union
VoIP calls to Pakistan for only $0.298 per minute!-Call Union
Cheap VoIP calls to Pakistan that only $0.298 per minute! by Call Union.
On Call Union VoIP service
Call Union(callunion.com) is the VoIP (Voice-Over-Internet) calling service that lets you call from anywhere to anywhere at a fraction of what the telephone company charges.
Using VoIP your telephone call is carried over the internet instead of traditional telephone lines using the SIP VoIP protocol.
Use Call Union to call land lines and mobiles in the Pakistan at Call Union competitive tariff of $0.298 US dollars per minute. Compare our prices to see that Call Union is the leader in high quality low cost Internet telephony! (Please check Call Union price page for the latest updates.)
In addition, you can make free calls to other internet telephone users, and discounted calls to normal phones anywhere.
Use your broadband internet connection instead of, or in addition to, your normal telephone line.
Make calls using Call Union free PC-Phone or using a normal telephone connected to the internet using a VoIP Adapter.
Benefit from free call features, such as caller ID, call waiting and call forwarding, and voicemail-to-email.
There is no contract to sign, no hidden fees or taxes, and you can modify or cancel the service at any time.
AUS and NZ £ 1.4p/min.
China $ 3.0¢/min.
France / Germany £ 1.4p/min.
Moscow / St. Petersburg $ 2.4¢/min.
South Africa £ 4.7p/min.
United Kingdom £ 1.2p/min.
United States $ 2.4¢/min.
How do you think about VoIP calls to Pakistan for only $0.298 per minute?
Thank for good answer for think about VoIP !
Cheap VoIP calls to Pakistan that only $0.298 per minute! by Call Union.
On Call Union VoIP service
Call Union(callunion.com) is the VoIP (Voice-Over-Internet) calling service that lets you call from anywhere to anywhere at a fraction of what the telephone company charges.
Using VoIP your telephone call is carried over the internet instead of traditional telephone lines using the SIP VoIP protocol.
Use Call Union to call land lines and mobiles in the Pakistan at Call Union competitive tariff of $0.298 US dollars per minute. Compare our prices to see that Call Union is the leader in high quality low cost Internet telephony! (Please check Call Union price page for the latest updates.)
In addition, you can make free calls to other internet telephone users, and discounted calls to normal phones anywhere.
Use your broadband internet connection instead of, or in addition to, your normal telephone line.
Make calls using Call Union free PC-Phone or using a normal telephone connected to the internet using a VoIP Adapter.
Benefit from free call features, such as caller ID, call waiting and call forwarding, and voicemail-to-email.
There is no contract to sign, no hidden fees or taxes, and you can modify or cancel the service at any time.
AUS and NZ £ 1.4p/min.
China $ 3.0¢/min.
France / Germany £ 1.4p/min.
Moscow / St. Petersburg $ 2.4¢/min.
South Africa £ 4.7p/min.
United Kingdom £ 1.2p/min.
United States $ 2.4¢/min.
How do you think about VoIP calls to Pakistan for only $0.298 per minute?
Thank for good answer for think about VoIP !
Wednesday, June 25, 2008
Free VoIP Calls To Pakistan-LowRateVoip
How about free VoIP calls to Pakistan?
The time of VOIP service provider lanch Free VoIP calls to Pakistan.
LowRateVoip.com is a product of popular VOIP service provider BETAMAX, LowRateVOIP is offering free calls to countries generally not offered by other free VOIP providers like free calls to Pakistan (both landlines and mobile), however you need to top-up the minimum call credit for taking advantage of the free call offer, Registered users get max 200 minutes per week of free calls, measured over the last 7 days and per unique IP address. Unused free minutes cannot be taken to the following week(s). If limit is exceeded the normal rates apply. During your Freedays you can call all destinations listed as "Free" for free. When you have run out of Freedays, the normal rates apply. You can get extra Freedays by buying credit.
Below is the list of free call destinations:
Andorra,Australia,Austria,Bahamas (+mobile)
Belgium,Bermuda (mobile),Brunei Darussalam (+mobile),Bulgaria,Canada,Chile,China (mobile)
Croatia,Cyprus (+mobile),Czech Republic,Denmark,Estonia,Finland,France,Germany,Gibraltar,Greece,Guadeloupe,Guam,Hong Kong (+mobile),Hungary,Iceland,Ireland,Israel,Italy,Japan,Jordan,Laos,Latvia,Liechtenstein,Lithuania,Luxembourg,Macao (+mobile),Martinique,Monaco,Netherlands,New Zealand,Norway,Pakistan (+mobile),Panama,Peru,Poland,Portugal,Puerto Rico (+mobile),Reunion,Romania,Russian Federation,San Marino,Singapore (+mobile),Slovenia,South Korea (+mobile),Spain,Sweden,Switzerland,Taiwan,Thailand (+mobile),United Kingdom,United States (+mobile),Us Virgin Islands,Venezuela
Visit LowRateVoip.Com for free calls to Pakistan and Jordon.
How you think about Free VoIP Calls To Pakistan-LowRateVoip?
Enjoy VoIP !
The time of VOIP service provider lanch Free VoIP calls to Pakistan.
LowRateVoip.com is a product of popular VOIP service provider BETAMAX, LowRateVOIP is offering free calls to countries generally not offered by other free VOIP providers like free calls to Pakistan (both landlines and mobile), however you need to top-up the minimum call credit for taking advantage of the free call offer, Registered users get max 200 minutes per week of free calls, measured over the last 7 days and per unique IP address. Unused free minutes cannot be taken to the following week(s). If limit is exceeded the normal rates apply. During your Freedays you can call all destinations listed as "Free" for free. When you have run out of Freedays, the normal rates apply. You can get extra Freedays by buying credit.
Below is the list of free call destinations:
Andorra,Australia,Austria,Bahamas (+mobile)
Belgium,Bermuda (mobile),Brunei Darussalam (+mobile),Bulgaria,Canada,Chile,China (mobile)
Croatia,Cyprus (+mobile),Czech Republic,Denmark,Estonia,Finland,France,Germany,Gibraltar,Greece,Guadeloupe,Guam,Hong Kong (+mobile),Hungary,Iceland,Ireland,Israel,Italy,Japan,Jordan,Laos,Latvia,Liechtenstein,Lithuania,Luxembourg,Macao (+mobile),Martinique,Monaco,Netherlands,New Zealand,Norway,Pakistan (+mobile),Panama,Peru,Poland,Portugal,Puerto Rico (+mobile),Reunion,Romania,Russian Federation,San Marino,Singapore (+mobile),Slovenia,South Korea (+mobile),Spain,Sweden,Switzerland,Taiwan,Thailand (+mobile),United Kingdom,United States (+mobile),Us Virgin Islands,Venezuela
Visit LowRateVoip.Com for free calls to Pakistan and Jordon.
How you think about Free VoIP Calls To Pakistan-LowRateVoip?
Enjoy VoIP !
Cheap VoIP calls to India Unlimited
Most VoIP company make for cheap Voip to IndiaUnlimited. Unlimited voip calls to India is almost impossible atleast in current scenarios because of the high termination charges voip company have to pay. So please do not look for these fake offers which will either loot you or will make it unsustainable for company in long run. Best is to search for lowest possible per minutes rates which are around 4 to 7 (US)cents/min currently.
So another list VoIP providers of best rates(per min ) for cheap calls to India follows
Joiphone.com offer rates in range of 7 to 9cents depending on type of number(mobile or Landline).
Voicestick.com rates from 6 to 8cents depending on plan and type of number(mobile or Landline).
CallEasy.com* offer rate of 2 euro cents to both mobile and landline(Sip calls not available).
VoipWise.com* offer rate of 2.5 euro cents per min for sip calls to India.
Smsdiscount.com* offer rate of 3 euro cents flat per min.
Nonoh.net* same 3euro cents per minute flat rate.
Enjoy Cheap VoIP Calls To India,But All the operators marked are Betamax providers, beware there rates change too frequently.
Enjoy VoIP !
So another list VoIP providers of best rates(per min ) for cheap calls to India follows
Joiphone.com offer rates in range of 7 to 9cents depending on type of number(mobile or Landline).
Voicestick.com rates from 6 to 8cents depending on plan and type of number(mobile or Landline).
CallEasy.com* offer rate of 2 euro cents to both mobile and landline(Sip calls not available).
VoipWise.com* offer rate of 2.5 euro cents per min for sip calls to India.
Smsdiscount.com* offer rate of 3 euro cents flat per min.
Nonoh.net* same 3euro cents per minute flat rate.
Enjoy Cheap VoIP Calls To India,But All the operators marked are Betamax providers, beware there rates change too frequently.
Enjoy VoIP !
Tuesday, June 24, 2008
Second look and invites for AT and T's improved Pogo web browser
AT&T both intrigued and confused us back in April when it announced Pogo, a new Mozilla-based web browser (of all things) that they're testing in a very private beta. It's notable for some new ideas about how web browsers should present information, and we found some of them interesting when we gave Pogo a thorough testing. But the remarkably heavy system requirements needed to come down out of the clouds for Pogo to attract much of an audience. Today, AT&T told us that a new private beta is ready to roll, and the system requirements have been significantly reduced—the company gave us 500 invites to share with Ars Technica readers so you could see for yourself.
Not much has changed about Pogo's UI or functionality since we first reviewed it, so we aren't going to rehash much of our previous coverage. In a nutshell, Pogo is AT&T's first step into an increasingly competitive web browsing market, one that's heavy on visual flourishes. Really heavy. Pogo uses baroque 3D scrolling visualizations to present categories of thumbnailed bookmarks and your browsing history, as well as a Dock to present open pages in thumbnailed "cells" (tabs, in normal speak). There's even an Opera-esque Springboard that presents a grid of your favorite sites for easy access.
Back in April, though, Pogo's heavy emphasis on a visual UI (one that sometimes eschews usability) came with a price in terms of performance that was surprisingly high. The minimum system requirements included a 1.6GHz processor, 2GB of RAM, and a video card with at least 256MB of VRAM. This obviously meant that we couldn't take Pogo for a spin in any Windows virtualization apps, but it turned out that some respectable PCs with very capable video cards couldn't even handle it. When we brought out the big hardware guns, Pogo still brought the machine to its knees with barely three tabs cells open. At that point, it was clear why Pogo was available only in a very private beta.
Now Pogo is back with some significant performance enhancements and system requirements that have been cut virtually in half. While there aren't any whiz-bang new features, Pogo can now run with a 1.0GHz processor, 1.0GB of RAM, and a video card with 128MB of VRAM. Since Pogo was arguably doing fine in the whiz-bang department already, we figured it was worth another spin.
This time around, we installed Pogo beta 1.1 on a Vista Home Premium install running native on a quad-core 2.6GHz Mac Pro with 3GB of RAM and a GeForce 7300 GT video card sporting 256MB of VRAM. We hit the ground running by opting to have Pogo import our del.icio.us bookmarks—all 1,680 of them—which took well over 10 minutes. Initially we assumed that Pogo was preemptively caching a thumbnail for every bookmark it imported as it went. Unfortunately, we were asked to create said cache after the import process' "all done!" ding woke us up.
Pogo's attempt to render lists of bookmarks as old 'n busted
Again, we won't rehash all the different visuals that Pogo brings to things like bookmarks and browsing history. While our test machine this time around is actually a bit lighter-weight than the Opteron 256 with two 3GHz CPUs (as well as 4GB of RAM, and an NVIDIA 8800 GT video card with 512MB of VRAM) that we maxed out our testing on last time, we're happy to report that Pogo's performance has improved quite a bit. While just three open pages rendered Pogo useless back in April, Pogo remained acceptably responsive with as many as 6-10 pages open.
That said, our Quad Mac Pro test machine is still well above Pogo's new minimum system requirements, which makes us wonder just how well such a UI-intensive browser will perform on most people's computers, which are probably closer to Pogo's minimum system requirements than to the "recommended" stats.
On a more general level, after our second time around the block with Pogo, we wound up questioning how useful the visually rich UI actually is. Playing with Pogo again and putting its notable features through the ringer left us wondering if Pogo's developers watched Minority Report one too many times.
But don't take our word for it—the Pogo team wants to give Ars Technica readers 500 invites so you can take it for a spin yourselves. Just head over to the Pogo site and enter "r3STRPgV" as your invite code, but make sure your machine meets the default system requirements (we know: it feels weird having to warn you guys about system requirements). If you fire Pogo up though, definitely let us know what you think about it and the specs of the machine you're running it on.
Not much has changed about Pogo's UI or functionality since we first reviewed it, so we aren't going to rehash much of our previous coverage. In a nutshell, Pogo is AT&T's first step into an increasingly competitive web browsing market, one that's heavy on visual flourishes. Really heavy. Pogo uses baroque 3D scrolling visualizations to present categories of thumbnailed bookmarks and your browsing history, as well as a Dock to present open pages in thumbnailed "cells" (tabs, in normal speak). There's even an Opera-esque Springboard that presents a grid of your favorite sites for easy access.
Back in April, though, Pogo's heavy emphasis on a visual UI (one that sometimes eschews usability) came with a price in terms of performance that was surprisingly high. The minimum system requirements included a 1.6GHz processor, 2GB of RAM, and a video card with at least 256MB of VRAM. This obviously meant that we couldn't take Pogo for a spin in any Windows virtualization apps, but it turned out that some respectable PCs with very capable video cards couldn't even handle it. When we brought out the big hardware guns, Pogo still brought the machine to its knees with barely three tabs cells open. At that point, it was clear why Pogo was available only in a very private beta.
Now Pogo is back with some significant performance enhancements and system requirements that have been cut virtually in half. While there aren't any whiz-bang new features, Pogo can now run with a 1.0GHz processor, 1.0GB of RAM, and a video card with 128MB of VRAM. Since Pogo was arguably doing fine in the whiz-bang department already, we figured it was worth another spin.
This time around, we installed Pogo beta 1.1 on a Vista Home Premium install running native on a quad-core 2.6GHz Mac Pro with 3GB of RAM and a GeForce 7300 GT video card sporting 256MB of VRAM. We hit the ground running by opting to have Pogo import our del.icio.us bookmarks—all 1,680 of them—which took well over 10 minutes. Initially we assumed that Pogo was preemptively caching a thumbnail for every bookmark it imported as it went. Unfortunately, we were asked to create said cache after the import process' "all done!" ding woke us up.
Pogo's attempt to render lists of bookmarks as old 'n busted
Again, we won't rehash all the different visuals that Pogo brings to things like bookmarks and browsing history. While our test machine this time around is actually a bit lighter-weight than the Opteron 256 with two 3GHz CPUs (as well as 4GB of RAM, and an NVIDIA 8800 GT video card with 512MB of VRAM) that we maxed out our testing on last time, we're happy to report that Pogo's performance has improved quite a bit. While just three open pages rendered Pogo useless back in April, Pogo remained acceptably responsive with as many as 6-10 pages open.
That said, our Quad Mac Pro test machine is still well above Pogo's new minimum system requirements, which makes us wonder just how well such a UI-intensive browser will perform on most people's computers, which are probably closer to Pogo's minimum system requirements than to the "recommended" stats.
On a more general level, after our second time around the block with Pogo, we wound up questioning how useful the visually rich UI actually is. Playing with Pogo again and putting its notable features through the ringer left us wondering if Pogo's developers watched Minority Report one too many times.
But don't take our word for it—the Pogo team wants to give Ars Technica readers 500 invites so you can take it for a spin yourselves. Just head over to the Pogo site and enter "r3STRPgV" as your invite code, but make sure your machine meets the default system requirements (we know: it feels weird having to warn you guys about system requirements). If you fire Pogo up though, definitely let us know what you think about it and the specs of the machine you're running it on.
Monday, June 23, 2008
Tokyo Telco Bets Big on VoIP
telecom services venture Japan Communications is expected to lease wireless spectrum from NTT DoCoMo (DCM) for a new type of cell-phone service. The Tokyo company is likely to be the first operator in Japan to offer mobile handsets that can make calls on-the-go using voice-over-Internet protocol technology.
JCI's service and other mobile VoIP services like it have the potential to change drastically the economics of cellular services. Analysts say that VoIP should mean lower rates for subscribers, especially for long-distance calls. That's good news for users in Japan, where basic monthly rates average around $60 and are among the highest in the world.
JCI officials have a loftier goal: Simplify communications by routing mobile-phone calls through the same digital channels we now use to browse the Net. Microsoft (MSFT) and Cisco (CSCO) have been pushing this concept, known as "unified communications," in recent years, but hurdles remain even for these tech giants.
JCI's service and other mobile VoIP services like it have the potential to change drastically the economics of cellular services. Analysts say that VoIP should mean lower rates for subscribers, especially for long-distance calls. That's good news for users in Japan, where basic monthly rates average around $60 and are among the highest in the world.
JCI officials have a loftier goal: Simplify communications by routing mobile-phone calls through the same digital channels we now use to browse the Net. Microsoft (MSFT) and Cisco (CSCO) have been pushing this concept, known as "unified communications," in recent years, but hurdles remain even for these tech giants.
Sunday, June 22, 2008
VoIP providers to get full 911 access with bill's passage
VoIP has had its ups and downs when it comes to accessing 911 emergency services, and some incumbent telcos haven't made things easier for VoIP providers like Vonage. Finally, the US Senate has passed legislation that requires 911 network operators to allow VoIP customers to get through, no matter what service they're calling from.
After a woman blamed Vonage for not connecting her to 911 and the resulting death of her baby in 2005, the FCC gave VoIP providers a 120-day ultimatum to implement 911 service. At first the companies had trouble meeting the deadline, but today VoIP providers boast that more than 97 percent of their customers have E911 (Enhanced 911) service. In February this year, the Senate passed legislation that requires all VoIP providers to supply their customers with E911 service, which properly transmits a callback number and address to 911 dispatch centers.
Despite this legislation and the VoIP industry's reports of E911 access for nearly all customers, we still see claims from VoIP providers that the incumbent telcos who operate the proprietary 911 system have been deliberately blocking VoIP access in order to stifle competition.
Those days may soon be over with the Senate's passage of the New and Emerging Technologies 911 Improvement Act. With the House having passed the legislation as well, it's now expected to be signed into law.
In addition to leveling the 911 access, the legislation gives dispatch centers liability protection when handling VoIP calls. An amendment added by Sen. Ted Stevens (R-AK), the bill will also require the US government to develop next-generation 911 capabilities and extend the range of these networks to the rural areas that are not yet endowed with E911.
After a woman blamed Vonage for not connecting her to 911 and the resulting death of her baby in 2005, the FCC gave VoIP providers a 120-day ultimatum to implement 911 service. At first the companies had trouble meeting the deadline, but today VoIP providers boast that more than 97 percent of their customers have E911 (Enhanced 911) service. In February this year, the Senate passed legislation that requires all VoIP providers to supply their customers with E911 service, which properly transmits a callback number and address to 911 dispatch centers.
Despite this legislation and the VoIP industry's reports of E911 access for nearly all customers, we still see claims from VoIP providers that the incumbent telcos who operate the proprietary 911 system have been deliberately blocking VoIP access in order to stifle competition.
Those days may soon be over with the Senate's passage of the New and Emerging Technologies 911 Improvement Act. With the House having passed the legislation as well, it's now expected to be signed into law.
In addition to leveling the 911 access, the legislation gives dispatch centers liability protection when handling VoIP calls. An amendment added by Sen. Ted Stevens (R-AK), the bill will also require the US government to develop next-generation 911 capabilities and extend the range of these networks to the rural areas that are not yet endowed with E911.
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