LIBJINGLE in IM/P2P jungle? GoogleTalk explains what it is!

In one of my previous posts VOIP IP Telephony: A week in VOIP and IP Telephony, I mentioned about LIBJINGLE and I was excited about it. Apparently, there were many others, according to googletalk blog itself. The search link it gives lists my site in page two!. That is fine. This will be a new era in IM P2P and VOIP, IP Telephony.
As much the excitement, there were some confusion as well about the LIBJINGLE library. So Mike Jazayeri, the Product Manager has taken the task up and explains the scenario's and capabilities as far as GoogleTalk is concerned. I think it is a good read and here are the topics that Mike decided to talk about;
1. Voice calls between other PC IM/VOIP clients such as Gaim, Adium, Psi, etc. And Google Talk.
2. Voice calls between mobile devices and Google Talk!
3. Peer-to-peer applications.

I am sure there are more ways than the above to use this library. I am for one playing with my Jabber server and Asterisk server. I will let you know how it goes.

VON 2006 by PulverMedia coming to San Jose

VON is the industry event for VoIP, now in its 10th year of delivering for C-level attendees around the globe.
Mark your calendar NOW to attend Spring 2006 VON, set to be the biggest and best yet.
Spring 2006 VON Conference & Expo will be taking place March 14-17 at the San Jose Convention Center in San Jose, CA.
You can check out the schedule here and if you want to catch the early bird registration this is the place to go.
I will be seeing you there.

A week in VOIP and IP Telephony

Microsoft enters VOIP arena with MCI
Microsoft has announced plans to release its first beta version of Live Messenger with VoIP.
Microsoft has staked a claim in the US VoIP market by announcing its software users can soon make Internet calls to standard phones thanks to a new partnership with MCI. Although rival products from the likes of Yahoo and Skype offer extra features, given Microsoft's broad market reach elsewhere and its alliance with MCI, this should not affect the company's foray into VoIP in the longer term.
Jingle All The Way
Today, two major advances have been made in the openness of our voice capabilities. This morning, the Jabber Software Foundation (JSF) introduced two new proposed extensions to XMPP, known as Jingle and Jingle Audio. These enhancements describe how to write software compatible with Google Talk's voice features and have been introduced into the JSF's standards process where they'll be reviewed and improved by the XMPP community. To make implementing these extensions even easier, we've released a library we call "libjingle."
Libjingle is the very same code Google Talk uses to negotiate, establish, and maintain peer-to-peer voice sessions, packaged as a library for other developers to use in their own projects. By incorporating Libjingle into your project, you enable its users to voice chat with other users of the Google Talk service.

Jingle bells continued... Google opens up google talk
Google has released this source code as part of our ongoing commitment to promoting consumer choice and interoperability in Internet-based real-time-communications. The Google source code is made available under a Berkeley-style license, which means you are free to incorporate it into commercial and non-commercial software and distribute it.
You can use any or all of these components.
* base - Low-level portable utility functions.
* p2p - The p2p stack,including base p2p functionality and client hooks intoXMPP.
* session - Phone call signaling.
* third_party - Non-Google components required for some functionality.
* xmllite - XML parser.
* xmpp - XMPP engine.

All Vonage VoIP Users Now Get E911 Service
Leading VoIP provider Vonage said Wednesday that its entire customer base has access to enhanced emergency 911 services.

“Today, any Vonage customer in the U.S. who dials 911, will get help when they need it most,” read the announcement.

Providing E911 E911 capability has held up the deployment of many VoIP providers and Vonage was no exception. The Federal Communications Commission extended the deadline for compliance to Nov. 28, but then extended it beyond that date without naming a specific date for compliance.

To Skype or not to Skype? Debate goes on and All agree that Skype is NOT enterprise ready, as of now!

Network World has conducted some tests regarding viability and usability of Skype use in the enterprise. I have spoken about this before and received some bad press from the Skype VOIP product users. I was not against the Skype nor any other IP Telephony or VOIP application. I was against the behavior of Skype in a network. Skype uses an unbridled use of network, draining available Internet bandwidth. It might be ok for a user with broadband connection. But think of a T1 connected office network with 10 or 20 Skype users!. It might be even worse if some of them have public IP addresses. (This is rare in an office network since all of them are behind some kind of firewall).

If you are a Skype user or a network Admin worried about Skype's network usage or related security, I think Network World's debate on whether Skype is ready for use in the enterprise is a good read. I am sure all will be able to see some light in the VOIP , Network or security tunnel created by Skype and you will agree that the peer-to-per communications software is not ready, for enterprise.
Start with Tim Greene's Beware of Skype. He says "Corporate IT time is better spent now keeping Skype off corporate networks , difficult as it may be, than trying to make it safe for business use." But not if you read my previous article VOIP IP Telephony: How to kill a Skype� / remove Skype� installs from your network! the task is much easier than all these people think.
Then you can go to other article/s which I enjoyed reading. Hope you do too.
I will be writing more on this soon.

Yahoo undercuts Skype, and others on call rates

Yahoo, which had Voice over it's messenger service (PC to PC) for a while now, did not have ability to call-out to land like skypeout service. But things seems to be changing. All the main stream media is reporting this as it is the beginning of the world.
Yahoo has it's eyes set on many directions now, It is getting tasty now with del.icio.us purchase.
Hope it can cope with the additions like google does. I use services from both, Yahoo and Google. Both have their goods and bads but it seems I happened to be more at Google. I really don't like to be attached to one service so I hope yahoo to be the other service to keep me away from Google. I have been dealing with MSN as well but it need to do more to pull me over to them.

Phone in a USB Stick called PhoneDrive! maybe we call it Skypedrive!!

I saw this voip gadget on the IOCELL website, a korean company specializing in USB memory products. After digging further into the information, I found that Memsen is selling the same product here in the US of A. They call it SkypeDrive but the details are not clear at the moment.
PhoneDrive is the world's first USB flash drive for VoIP transmission incorporated according to a press release from the US counterpart (I might be wrong) of the Korean IOCELL,

"This is an intelligent product and service that will change the way people view USB drives and is the key to allow people to thrive in a user-friendly and convenient digital environment. Using the USB drive equipped with Phonedrive technology enables people to carry, store and launch their own phone calls from any PC in any place through the Internet. The Memsen Phonedrive also enables users to place calls to land and cellphones for a nominal per minute fee. Most importantly, The Memsen Phonedrive has a built-in security system that automatically deletes any personal information stored in the USB drive by a remote server if lost and stolen. You can even make calls to the Phonedrive!"
"The Phonedrive is flexible and compatible with other VOIP service providers and calling card minutes providers. Memsen is currently in talks with distributors of USB drives from other countries in the world to ensure multi-vendor interoperability and compatibility. The Memsen Phonedrive will change the way individuals and business entities use and access information from USB Flash drives."

Cingular HSDPA Phone announced! See and talk at the same time!

HSDPA High Speed Downlink Packet Access
Cingular's HSDPA test network in Atlanta, using Lucent equipment. The
tests reached sustained data rates over 3 Mbps. Initial HSDPA devices
will suppCingular and Lucent today announced the first successful data calls onort 3.6 Mbps peak speeds. As faster devices are released, the network can ultimately deliver theoretical peak data speeds of up to 14.4 Mbps. HSDPA is an enhanced version of UMTS/WCDMA 3G technology.

The trial network was first announced in May. Cingular has already contracted with Ericsson, Lucent, and Siemens to provide HSDPA network equipment that should be launched most major markets by the end of 2006.
Read more at Lucent.
Read the Cingular News release

Google Talk have blog now!

I wrote earlier about google talk here and I found today google blog that there is a blog page dedicated to google-talk opened.
There is not much yet on the blog but you can be the first to read first article, go there from here!
What is google talk? From the google talk site;
"Google Talk is a simple and free way to talk with and send instant messages to your friends. Like Gmail, Google Talk uses Google's innovative technologies to help people communicate more effectively and efficiently. Think of it as Google's approach to communications.

Google Talk is easy and intuitive to use. All you need to make free calls is an Internet connection, a microphone, and a speaker. After you download Google Talk, sign in with your Gmail username and password. Invite your friends to download Google Talk, and once they do, you'll be able to talk or IM with them instantly."

Lots to Talk about!

OpenSER version 1.0.0 Is released

OpenSER, a fork of SER, Has released the Version 1.0.0. For those who does not know what OpenSER and SER is, Read the end of the article. Let me write about OpenSer Version 1.0.0 first.
New in OpenSER modules

acc module
[*] call leg accounting support

- proper accounting information can be stored when the server deals with multiple redirects
[*] accounting of failed transactions based on flags

- failed transactions can be stored based on specific flag
[*] usage of pseudo-variables format in parameters

- any pseudo-variable can be now stored as adjacent information
[*] the module is compiled by default with database support

avp_radius module
[*] enhancement to support loading AVPs having integer ID or value

avpops module
[*] formatted printing with pseudo-variables support - avp_printf()

- this allow string concatenation of avps, strings and pseudo-variables
[*] perl/sed-like substitution on AVP with string value - avp_subst()

- this allow substitution or extraction of parts from an AVP
[*] support for arithmetic operations with AVPs

- integer operations with avps can be done (add, sub, mul, div, mod)
[*] support for bit-wise operations with AVPs

- bitwise operations can be done with AVPs (and, or, xor, not)

- bitwise checks can be performed via avp_check (and, or, xor)
[*] more operators for avp_check()

- new operators: ne, le, ge + bitwise operators can be used
[*] cast function with avp_copy()

- convert the value from int to str and vice-versa
[*] uri parameter for db-related function can be taken from an AVP

- along with R-URI, From and To, the URI can be now loaded from an AVP
[*] new function to check if an AVP exists -- is_avp_set()

- useful function to test which AVP exists when loading all AVPs for an user
[*] hexadecimal format for integer values

- avp_write(), avp_check() and avp_op() allow hexa integer values as parameter - to ease bitwise operation handling
[*] avp_write() and avp_pushto() can access and set the value of 'dst_uri' field (outbound proxy address)
[*] avp_write() allow $hdr(name) to be coherent with avp_printf()

- old format $hdr[name] is still valid
dispatcher module
[*] possibility to change host:port in r-uri (ds_select_domain())
[*] round robin distribution per process (alg=4)

group module
[*] uri parameter for is_user_in() can be taken from an AVP

lcr module
[*] added support for gateway prefixes

maxfwd module
[*] MAX-FORWARDS cannot exceed 256 (as per RFC3261)

nathelper module
[*] possibility to replace origin IP in SDP (o= line)

- SIP devices which check o= line can be now used with nathelper
[*] nat ping with OPTIONS requests (stateless)

- possibility to interwork with NAT boxes which close the pinhole when no traffic goes from behind the nat

- made stateless to reduce memory consumption
[*] possibility to set nat pinging method per user

- you can set per user what type of natping to send (four bytes or OPTIONS)
postgres module
[*] transaction rollback for failed queries

- postgres module encapsulates every query into transaction. If the query failed, the transaction was not finished, thus a new database connection was created on the next query
rr module
[*] enhancements which opens the road for a dialog awareness support:

- RR API - exported functions:
** add_rr_param()
** check_route_param()
** is_direction()
** get_route_param()
** register_rrcb()
[*] added callbacks - can be registered callbacks to be executed when local Route is found and processed.

textops module
[*] pseudo-variables support in subst(), subst_uri() and subst_user()

- parts of sip message can be substituted by dynamic values
[*] function to check the request's method using ID

- faster comparison of request/reply's method

- you can test the method against a set of values (is_method("ACK|BYE"))
tm module
[*] usage of pseudo-variables format in parameters

- any pseudo-variable can be sent via fifo or unix socket to external applications
[*] support for delayed CANCEL

- canceled transactions will be marked to be able to cancel delayed replies
[*] new function t_check_trans()

- checks if the request belongs to a transaction
[*] new function t_was_canceled()

- returns true if the transaction was canceled from the UAC side

- request forwarding functions return false if the transaction was already canceled
[*] pending callbacks

- support to register TM callbacks prior the transaction is created
[*] t_flush_flags() - flush to Transaction (UAS side) only the global flags

uac module
[*] annonymization finalized

- full ability to replace and restore the From and To headers in subsequent requests
uac_redirect module
[*] - special module to handle redirect replies on server

- redirects via 3xx replies can be handled on server

- address filters can be set to allow/deny redirects
xlog module
[*] use pseudo-variables in xdb() and xlog()
[*] if the first parameter of xlog() is ommited, the message is printed to L_ERR level


additional tools
[*] postgressql.sh - script to create Postgres database structure
[*] sc.dbtext - script to manage dbtext database structure


The ChangeLog file keeps track of all important changes:

They also have listed the differences between OpenSER 1.0.0 and SER 0.9.
New in OpenSER vs. SER-0.9.0

NOTE: The next list presents what OpenSER brings new than SER 0.9.0. Several are backported from CVS head and the ones marked with (NEW) are newly added.

NOTE: The structures for 'usrloc' and 'aliases' tables has changed to store the incoming interface details. This solves the issues when dealing with NAT-ed clients and offers better support for SIP replication - requests, replies and NAT pings are sent to UA using same interface where the requests from de UA were received, so the NAT will not drop them.

Now here is info on both the SIP servers;
OpenSER is a project spawned from FhG FOKUS SIP Express Router (SER). The reason for this new venture is the lack of progressing and contributions to the SER project from the other SER team members as well as the reticience to new contributions from project's community members. We want to accelerate the integration of public contributions to the SER project.

OpenSER promotes a new management policy (OPEN) -for new code acceptance and code-through propagation- and development approach -design and architecture. We have decided to bring more dynamics into SIP world by creating this new project that can benefit of TLS and so many other contributions. We welcome your contributions to the success of this project.

The project is managed by three of SER developers. You all like SER so we will do our best to maintain our exiting work and help the users of our solutions. We will therefore maintain close contacts with our former colleagues and SER project.

Hear Me? See Me? Yes! on Skype Beta "Skype 2.0"

Skype, which is a part of eBay now, is to offer video calls with the latest version of its internet telephony (VOIP or IP Telephony) software. Well Where do go from here? Users rejoice and Network Admins? More reasons for Skypekiller?. I don't really know because I did not test it yet.
But we do report all things VOIP and IP telephony related. Since there is a volume of Skype users large enough to fill a small country, Skype is a part of VOIP and here we go! Skype conference!!
"At Skype we want to make talking over the Internet the most natural, simple thing for people to do all over the world. With the release of our new software, it's never been easier for people to talk to one another for free, and now they can see each other with video as well" said Skype chief exec Niklas ZennstrÃm.

Skype has teamed up with Logitech and Creative who make webcams and headsets needed to use the Skype service. There are other systems and phones that also can be used with Skype.Onee or two listed here.
I think Sony can say good bye to it's IVY, software released together with Glowpoint, a IP Telephony provider, Lest that too might have rootkits accompanying it. IVY league is ok, but here we consider IVY to be a *&%^$.
Here are some info from current Beta's change log;
# feature: Skype video (webcam)
# feature: contact grouping
# feature: quickfilter in contact list and history (enable from Options->Advanced)
# feature: new language - Portuguese (Portugal)
# feature: new sound events for chat user join, leave and incoming message to existing chat
# feature: chats and conference calls shown in history
# feature: expandable "My Panel" (mood, events, services integrated)
# feature: add contact directly from main window
# feature: show file transfers in history
# feature: quickfilter in history
# feature: delete single or multiple history items
# feature: compact chat participant list
# feature: save user's last auth request
# feature: API commands GET/SET PCSPEAKER
# feature: API commands BTN_RELEASED PAGEUP/PAGEDOWN
# feature: API command SET VIDEO_IN
# change: auth requests new design
# change: changed group selection hotkeys to Ctrl-PgUp/PgDn
# change: call tab visual layout changed
# change: tray icon connecting animation
# change: changed "Free Internet Telephony" to "The whole world can talk for free" in file description
# change: import contacts error dialog displayed when there is nowhere to import contacts from
# change: file transfer in DND mode open file transfer dialog in minimized mode
# change: new installer and uninstaller icons
# change: history limited to 30 days for all events
You can read more and download from, Skype Site.
Community discission here

USB VOIP phone With voice recorder

Here is a phone that Skype users will love. You can talk to who ever you want and if you are not sure if you can make the voip connection soon again, you can record the conversation, in .wav or mp3 format.
How sweet, you can have voice prompts in your .....'s voice. Fill in the blank as you prefer.
Geemodo: USB VOIP phone With voice recorder (UP-111C-A)

A must read discussion on Small Business VOIP!

This discussion on /.(Slashdot) is a must read if you are thinking of implementing VOIP. The discussion goes from Asterisk, Cisco Call manager, Cisco Call manager express, Vonage, SBC, AT&T, (did I write that twice, I meant SBC is AT&T!) 3COM, various CODECS, and Line quality etc. It is simple VOIP or IP Telephony GEM. Most of all don't forget to read this comment, which is at the bottom, about asterisk@home.

Also pieces like "Setting the call quality to the highest setting means that the G.711 codec is used, which consumes 64k/s per conversation. That's generally not a problem with a home user who only has one call happening at a time, but it will easily overwhelm the standard small-business broadband connection which might only have 128-256kps upstream bandwidth. Setting the call quality lower is probably using the iLBC or the GSM codec. GSM is commonly used for cell conversations, iLBC is a variable rate codec designed for VoIP. They both consume far less bandwidth, but you're right, the call quality sucks."

Also links like this. Giving information as how to setup a call center, "We now have over 250 phones installed at 4 locations(including a call center). We started switching to Asterisk three years ago and grew the system to the point where everything is Asterisk and we do all inter-office calls over VOIP(IAX trunks). The cost savings in licensing costs alone more than justifies 2 full-time IT staffers salaries. "

Vonage may get blocked, from signing up new customers.

A slashdot article lead me to the news that Vonage might be in trouble regarding e911 compliance.
Federal Communications Commission, FCC gave Vonage and other companies that sell Internet-based phone service 120 days to comply with its order requiring enhanced 911, or E911, in all their service areas. I have reported on this earlier. Click here to see all articles if you want to see the issues leading to the current situation.
From the article: "The company -- which has more than 1 million subscribers -- said it was capable of transmitting a call back number and location for 100 percent of its subscribers, but that it still was waiting for cooperation from competitors that control the 911 network." May be they want that free, SBC said earlier that if providers are willing to pay the service will be available from them.
The deadline to show the government where E911 is available was Monday. House and Senate lawmakers had urged FCC Chairman Kevin Martin to give companies more time and more tools to speed deployment, but no extension was granted.

In its compliance report to the FCC, Vonage said only 26 percent of its customer base had full E911 services. The company — which has more than 1 million subscribers — said it was capable of transmitting a call back number and location for 100 percent of its subscribers, but that it still was waiting for cooperation from competitors that control the 911 network.

AT&T declined to comment on its compliance levels before filing its report with the FCC. Calls to the company on Tuesday were not immediately returned. AT&T offers Voice over Internet Protocol, or VoIP, to about 57,000 customers through its CallVantage service.

SunRocket, which has more than 50,000 subscribers nationwide, said it had equipped 96 percent of its customers with full 911 services.

The VON Coalition, an industry group, had estimated that overall about two-thirds of Internet phone users would have enhanced 911 by the deadline.

Citing public safety concerns, the FCC in May ordered companies selling Voice over Internet Protocol, or VoIP, to ensure that callers can reach an emergency dispatcher when they dial 911. The dispatchers also must be able to tell where callers are located and the numbers from which they are calling.
Read more at Yahoo and /.

Google wi-fi (goofi) approved by Mountainview City.

At a meeting on 15 November, the Mountain View City Council unanimously approved a plan that gives Google access to city-owned streetlight poles for the placement of wireless access points for a city-wide Wi-Fi network. In an e-mail statement, a Google spokeswoman said the company is "excited to begin work on this project" and is looking forward to providing Wi-Fi service to members of the local community.
"The City potentially could receive an annual payment of approximately $12,600 [adjusted annually for increases in the Consumer Price Index (CPI)] for the placement of Google equipment on City-owned light poles," according to the document. "All installation and maintenance costs will be borne by Google, and utility costs will be paid by the City and fully reimbursed by Google, which is estimated to be $3,000 to $4,000 per year."
Google hopes to use the Mountain View network as a proving ground to show officials in large metropolitan areas that the search giant can provide city-wide Wi-Fi access. About 72,200 people live in Mountain View, located about 40 miles (64 kilometers) from San Francisco. The company already provides Wi-Fi access in two local Mountain View businesses, and in Union Square, a popular outdoor public space in downtown San Francisco.
May be this will pave a way for vowifi. Will it be goofi woof!

Cisco plans to score in IP PBX on Digital Fairway!

With Cisco's last acquisition of Scientific Atlanta Everyone was looking for video over IP plug in their routers. I think finally Cisco thought there is something to be had in video arena. So next time when you buy a router, look for that video plug!
Hidden behind the Scientific Atlanta buy out is another Cisco Purchase, this time it is Canadian company, Digital Fairway.
Digital Fairway, a developer of IP telephony provisioning systems, intellectual property and other assets was purchased for $15.25 million in cash. Digital Fairway, which is based in Toronto, will become part of Cisco's Network Management Technology Group, joining the CDIC (Campus, Data Center and IP Communications) group led by Vice President Clive Foreman.
Digital Fairway's IP Telephony Provisioner will form the foundation for Cisco's IP Communications Provisioning Manager, a new enterprise-class IP communications provisioning product to be released in 2006. The IP Communications Provisioning Manager will enable rapid enterprise-wide IP telephony deployment by simplifying and automating operations through workflow-based unified provisioning. In addition to user and dial plan provisioning, it will also support change management using a database of record. Cisco will also incorporate the assets into its lifecycle services framework.

Online dting service adds VOIP services to the mix.

Had to misspel in order to get trackback to ping.(did not like dating, asian!)
Tom Keating at VOIP Blog is asking why Dating services based here in USA, slow to take on services like VOIP and Video Over IP. He also reports about the addition of VOIP service by Vivox, the largest online dating service in Asia.
Here is a part of press release by Vivox,
Vivox, a pioneer in context-specific communications services, today announced that it will power WorldFriends Networks' (WFN) new WorldFriends Phone service, which will allow WFN's half-million global users to communicate with each other via live text, voice or video chat. In doing so, WFN becomes the industry's first online dating service to provide its user community with an integrated Web-voice-video-Instant Messenger communications platform.

Owned and operated by Meta4 Group Limited, WFN is one of Asia's largest online dating sites. WFN caters to an international constituency by operating in 20 different countries and in four different languages - English, Chinese, Japanese and Korean. With more than 150 partner websites, WFN registers more than 1,000 new members every day making it the world's premier provider of private label multi-lingual online personals.

Read the rest and more at VOIP BLOG.....

Conference calls with Skype

For those who thinks I am against skype, here is a proof, that I am not. I am against the methods that skype uses. I don't like someone taking control of my PC.
Anyway I seem to be in the minority when it comes to Skype, eventhough there are better VOIP applications that are similar or better than Skype, there are more than enough users. So here we go.

Skype Conference Phone. (VOIP, IP Telephony hands free phone)
buffalo Corporation Nagoya city has some interesting products for computer users. I have seen many a gadgets, gizmos related and unrelated to voip or IP telephony come from this place.

The conference phone slated to released in the latter quarter of December 2005, is currently known as BSKP-CU201/BK.
It has smart microphone and a Speaker. It is said to capable of supporting 5-6 users simultaneously. Good for the Xmas Holidays, to say hello to grand ma and Grand pa together.


The system also has a microphone and headphone so that if you want to croon your better half in privacy, you can use those connectors.
There is no need to make changes to your current Skype software nor configuration. No external power source is necessary as it is powered by USB.

It seems you really don't need Skypekiller

As discussed here, About the application skype and skypekiller. Troubling? News have surfaced about the SKYPE company. It seems that Skype company killed it self! Well when VC's come in, money becomes more important than people. But it should not be hard to treat people better.
Andy at VOIP Watch has shortlist about post merger skype news.
Skyp e bay deal info is here.

Gurgle! With all the Google's Click-to-talk, Google talk and Free wi-fi You soon will be abale to gurgle.

Gurgle: Utter with a gurgling sound; "'Help,' the stabbing victim gurgeld" in this case the victim will be telecomm companies!
Google Talk, the voice-enabled IM client based on Jabber platform.
There was news that Google was experimenting with selling offline ads. Yannick Laclau, discovered something interesting in an article about AHS systems, the company featured in the Google experimental ads.
Also google-Wi-fi, GOOfi has taken off to a good start, Demand exceeds the supply.
This is what google say now;
The Google Secure Access client was originally intended to provide secure web access for users at a few local Mountain View, CA "hotspots" that Google established as part of a community outreach program. Adoption of this limited release technology has exceeded our expectations and we recently began restricting access to the few locations it was originally intended for. We are excited about the widespread enthusiasm for this technology and are currently investigating ways to make it more broadly available in the future.

“It’s a lot of exposure for cheap,” he said, adding that Google is “doing a ton of tracking on this. They’re using their own 1-800 numbers on this, and it forwards to our line.” The Internet addresses of the online versions of the ads also redirect traffic through Google servers.

With companies such as Ingenio and eStara promoting click-to-call technology and ecommerce sites looking to add click-to-call to help close sales, "click-to-call VoIP" (click on a web link and initiate a VoIP call) continues to make strides. Google, eBay, Yahoo and other major Web players have click-to-call directory assistance, not yet in wide use, as well as "click-to-close" (the sale) utilizing Web-initiated VoIP calls to aid in closing sales. May be for shopping carts that you find everywhere on the net. It is also a better way to verify the seller, in this internet age where verifying the buyer is the most important factor in eSALES.
One seller on the net explained the reason to use click-to-call this way;
"We know that more than 9 out of 10 industrial buyers go online first when looking for products and services. A good portion of our clients' business is initiated online, and completed offline. This is difficult to track for anyone".
I would say plus for clients like me, who still picks up the phone to order, after finding the product on the net.
Now that it is on /.(slashdot) may be more people will click and call.

AstPlanDesigner 0.1 download and have fun with Asterisk!

AstPlanDesigner 0.1 is a visual dial plan designer. It aids you in building a dial plan by drawing your plan. It is a good start for cumbersome extension.conf creation. Since it is written in Java, it will run anywhere.
If you an Asterisk voip/pbx user or developer, give it a go! It is free too.
Here is a screenshot!

Get your Cisco gear upto date, get Cisco IOS Software Release 12.4(4)T.

Cisco IOS Software Release 12.4(4)T, the second release in the Release 12.4T family, is now available. Release 12.4(4)T enhances threat protection against malicious worm and virus attacks, improves performance monitoring of VoIP networks, and extends support for secure concurrent services on the Cisco 1800 Series router.

Key features of Release 12.4(4)T include:

Cisco 1801, 1802, and 1803 Integrated Services Routers -- provide a cost-effective, single-box solution for small, medium, and branch office applications with secure concurrent services for broadband access


Cisco IOS Flexible Packet Matching -- next-generation Access Control List technology that provides a rapid first line of defense against malicious traffic at the entry point into the network


Application Firewall for Instant Messenger Traffic
Enforcement -- reduces exposure to potential vulnerabilities from instant messenger clients


Cisco IOS IP Service Level Agreements for VoIP with Real Time Protocol -- extends the productivity, OpEx, and availability benefits of Cisco IOS IP SLAs to VoIP networks


Hot Standby Router Protocol (HSRP) for IPv6 -- increases network availability by extending the fault tolerance and fast switchover capabilities of HSRP to IPv6


NetFlow Reliable Export via Stream Control Transport
Protocol -- ensures the integrity of accounting and billing information
Visit the URLs below to learn more about Cisco IOS Software Release 12.4(4)T and the entire Release 12.4T family.

Cisco IOS Software Release 12.4T homepage:

www.cisco.com/go/release124t/

Cisco IOS Software Release 12.4T New Features and Hardware Support, Product Bulletin No. 3001:

www.cisco.com/go/124tpb/

How to kill a Skype® / remove Skype® installs from your network!

How To Uninstall Modern Skype from WIndows 8

An updated version of this article is here

If you are looking to remove unauthorized installations of Skype® from your network?
Is your DSL router constantly hanging, when you or another at your home using Skype®? To make sure it is Skype®, turn off the computer with Skype®, just shutting down skype will not stop Skype®.
First you need to get skypekiller,

Then install it on a windows workstation or a server and make sure that a connection to your network is up and ready.

Now fireup SK and Select the default clean mode or tick the "Detect only" option if you just want to list Skype® installs.

Now is time to setup Filtering
You can apply filters to determine the type of target computers on which Skype® is to be removed. For example you can choose to run SkypeKiller only on your workstations.


Once filtering is setup, you need to Select Target Computers
Select target computers. You can browse to quickly add computers from your global catalog or Domain controllers.


Now that you have a selection of computers, it is time for Execution
Click on Execute, a progress window will display the operation.

Now is time to go and get your other work done or have cup of coffee. You can come back to check Result analysis.
SkypeKiller will indicate success or failure of the removal for each previously selected system.

Like any other network application, you also can schedule periodic runs of the SK.


To be sure, I would run a "detect only" first and also check on your bandwidthth usage using your monitoring tools. Again run the same tool, once the Clean step is done. You will be able to see just roughly how much bandwidth this particularer VOIP or IP Telephony application was using.

Look for updates here when SK updates its application, remember this is the first release.
If you are wondering whyReadad the following articles;
Skype explains why security evaluation omitted bug reports
Should businesses ban Skype?

SkypeKiller arrives in the VOIP scene!

After reporting about the hazards of skype's action on a corporate network,here and here, I found that a skypekiller has come to be. I found this news at Skype journal and of course they seems to not so happy about it. I have no qualms about it. If Skype cannot come clean with it's network usage, someone got to do it. Network administrators have to go from desk to desk to uninstall it as casual uninstallation will leave something behind that still use the network.
So as a Skype user, one does not have to be unhappy, if you are willing to share your network connection with countless others, in exchange for free calls, be my guest. But not me. There are countless other solutions that one could use to achieve to same goal.
Don't come breaking my network. There will be a step by step guide to remove Skype from your network will be here. It is not that difficult.

Asterisk 1.2.0 has been released!

The Open sourced complete PBX, Asterisk has reached another milestone. Now the version 1.2.0 has been released. I am a long time user of this complete VOIP PBX purely in software, and have been impressed with it ever since I first got it. I even met Mark at a trade show a few years back and I even like the guy too, not only his software. On the trade floor he took time to walk me through the inner working of the System.
According to Mark Spencer and Kevin P. Fleming, this release of Asterisk contains over 3,000 improvements on version 1.0, including hundreds of new features and applications.

If you are interested you can download it from ftp.digium.com FTP servers, as well as the Digium CVS servers (under the 'v1-2-0' tag).

They also said that they want to extend Their thanks to all the community members whose contributions have made this release possible; without their coding, support, testing and other involvement Core developers would not have achieved this milestone!

You can reach the site by clicking on the "Favorite VOIP PBX" under the links on right hand side column.

New VOIP Magazine Announced

News Source



I have been a long time subscriber of the INTERNET TELEPHONY magazine, Over the years, it continued to be published in printed form even many other similar printed magazines, disappeared for the news and magazine radar.
I like what they print, but it might not be for everyone. Then again if you are here reading this VOIP blog, go get a subscription, you can read it when your net is not available!
Technology Marketing Corporation (TMC) publishes four print magazines: Customer Interaction Solutions, Internet Telephony, SIP Magazine and IMS Magazine; as well as the digital publications, Speech-World, WiFI Telephony Magazine, VoIP Developer, IPTV Magazine and WiMAX Magazine, Has a New Magazine, SIP Magazine.

Rich Tehrani, TMC president and group editor-in-chief for SIP Magazine, said "The SIP market is one of the fastest growing anywhere and is an essential technology in the world of VoIP, IP Multimedia Subsystems (IMS), presence and VoIP peering/network interconnectivity. Everywhere you look it seems, SIP is there and is enabling tomorrow's applications today,"
"I've never seen a better time to launch a new publication," says Tehrani. "The SIP market is ready, and we're ready to lead it, just as we have done in the VoIP market since 1998 with INTERNET TELEPHONY magazine."

A free subscription to SIP Magazine is available here.

Federal Court Denies VOIP groups Motion to stay! on E911

Image from /.
The U.S. Court of Appeals for the District of Columbia denied a motion filed two weeks ago by a group of Internet telephone companies who claim the regulations are unreasonable.
The e911 requirements were reported here on this blog a few times in the past, read here, here, here and here.
In May, the FCC ordered providers of Internet-based phone calls, commonly called Voice over Internet Protocol, or VoIP, to certify that their customers will be able to reach an emergency dispatcher when they call 911. Dispatchers also must be able to identify the caller's phone number and location. Subsequently, date got pushed many a times and was set to Nov 28 with certain guidelines. Read this about the new FCC site Dedicated to e911 issues.

Jason Talley, president and chief executive of Overland Park, Kan.-based Nuvio Inc., which filed the motion for a stay on Nov. 1, said the FCC's decision ultimately reduces alternatives for customers and stifles innovation.

Three other companies that later joined Nuvio's suit, Lightyear Network Solutions LLC; McLean, Primus Telecommunications Group Inc.'s subsidiary Lingo Inc.; and i2 Telecom International Inc. will continue to pursue their appeal of the regulations.
Many other news sites carry the news, since VOIP and IP Telephony is now mainstream media items.

We all need to know about patents, Now we have a place to go

The way it is going, somebody is going to patent sneezing, it is already DOMAINated, and copy righted! So you are going to have to stop sneezing or cough up (no pun intended) some money to some Joe Gates somewhere.
I have been following this patent drama for a while and was happy to hear that ODSL (Open Source Development Labs) has came up with a reference library.
I think everyone, including people who patent your sneeze, have trouble figuring out all the new patent pledges and patent strategies. I do! And I am happy That ODSL started the patent commons reference Library. To protect Free and Open Source software and standards is progressing, and yourself!! We all should know all the terms and conditions. Patents applicability, Or simply a place to learn how complex the sloppy US patent procedures are.

A simple guide if you visit the site;
Contributors
Contains information about the people and companies that have made Commitments about software patents they hold. This database may be searched by Contributor name.

Commitments
This database is comprised of promises, pledges, covenants and other legal undertakings made by Contributors. Commitments may be searched by title, content, type, or Contributor. Learn more about Commitments here.

Patents
A collection of patent abstracts and links to the patents identified in some of the Commitments. The Patents database may searched by patent title, abstract, type of patent, patent number, or assignee.

Standards & Technology
Summaries and links to the Standards & Technology in support of which some Commitments are made. This database may be searched by content, developing organization, type of Standard & Technology, or Contributor.
Other Legal Solutions
Information about indemnification programs, litigation support funds, open source software licenses and other legal solutions that reduce the threat and potential impact of patent litigation. This database can be searched by Contributor or solution type.

All are welcome here, Companies, universities, nonprofits, and individuals can all contribute to The Commons through Commitments covering software patents.

Contact info is here, and the Press release below;


OSDL Launches Online Patent Commons Reference Library

For More Information:

OSDL
Jennifer Cloer for OSDL
Page One PR
Phone: +1 503.547.9451
E-mail : jennifer@pageonepr.com
OSDL hosts online patent commons library supported by industry leaders including CA, IBM, Intel, Novell, Red Hat, and Sun Microsystems to help protect open source software innovation

BEAVERTON, Ore., - November 15, 2005 - The Open Source Development Labs (OSDL), a global consortium dedicated to accelerating the adoption of Linux®, today announced the launch of its online patent commons reference library, the foundation of its Patent Commons Project. The Project’s goal is to provide greater confidence for developers and customers of all open source software.

The site, patentcommons.org , hosts searchable databases containing more than 500 patents pledged to date and more than a dozen technical standards supported by patent pledges and covenants. The library is freely available to developers, users and vendors, where they can quickly view information about patents and technology pledges benefiting open source software and standards.

''The OSDL Patent Commons Project is an important first step in helping customers, vendors and the development community understand the different commitments that have been made and how they work to reduce the chances of patent litigation,'' said Stuart Cohen, CEO of OSDL. ''The Project is focused on documenting the growing number of pledges and other legal solutions directed at the software patent issue, so that developers can innovate and collaborate as free as possible from litigation.''

The Patent Commons website will catalogue existing patent commitments from companies and individuals who wish to retain ownership of their patents, and will provide information about different types of pledges and covenants and how they work. In the coming months, the site will expand to include other legal solutions that benefit the open source community, including open source licenses, indemnification programs and information for organizations and individuals who wish to contribute to the commons.

The OSDL Patent Commons Project has already rallied the support of many industry leaders, including CA, IBM, Intel Novell, Red Hat, and Sun Microsystems. The Lab welcomes other IT vendors, corporations, organizations, government agencies and individuals to participate.

''CA is committed to fostering innovation in the Open Source community so that users can reap all the potential personal, social and business benefits that technology can offer,'' said Sam Greenblatt, senior vice president of technology at CA. ''We are supporting and participating in the Patents Common because we believe it will enable developers to fully and appropriately leverage each otherÂ’s innovations, while respecting partiesÂ’ intellectual property rights.''

''OSDL provides a natural point of entry to the Commons. We are confident that the Project will serve the needs of developers and customers by providing fair, objective and easily accessible information about the burgeoning Commons,'' said Jim Stallings, vice president, Intellectual Property & Standards, IBM.

''As a founding member of OSDL, Intel is committed to helping customers make informed decisions around their choices in computing platforms,'' said Richard Wirt, Vice President, Senior Fellow, and General Manager, Software and Solutions Group of Intel. ''OSDL is in a unique position to provide a trusted clearinghouse where enterprise customers and developers can find vendor-neutral information about open source software and intellectual property that can help them ensure that their decisions are based on the most complete and up-to-date information.''

''Customers want freedom of choice in making decisions about technology solutions,'' said David Patrick, vice president and general manager for Linux, Open Sources Platforms and Services at Novell. ''They should be able to make their purchase decisions based on technical merits, security, quality of service and value, not concerns over intellectual property ownership. The OSDL Patent Commons project will provide greater confidence to developers and customers that the open source solutions they are deploying are safe from patent challenges.''

''We are happy to see OSDL's Patent Commons online reference library go live,'' said Mark Webbink, Sr. Vice President, Red Hat. ''As the first open source vendor to make its patents available to the open source community, Red Hat views steps such as the one OSDL has taken with the Commons and Red Hat's creation of the Fedora Foundation as providing developers, vendors and end users with the ability to be confident and bold in their development efforts.''

''Sun applauds the work of the OSDL Patent Commons project and its library of patent pledges and non-assertion covenants,'' said Simon Phipps, Chief Open Source Officer, Sun Microsystems, Inc. ''As the largest commercial code contributor to the various open source communities, Sun is well aware of the many obstacles these communities face due to the uncertainties that surround today's software patents, which neither patent pools nor targeted pledges really solve. This project offers a concrete and important step in the right direction, as it will help all open source communities.''

About the OSDL Patent Commons Project

With increasing frequency, institutions, companies, and inventors wish to signal formally to the open source software industry and community that software patents they hold are not a threat to the development, distribution or use of open source software or open standards. Patent pledges and covenants – legally enforceable promises not to enforce patents under certain terms and conditions – eliminate the need for individual agreements and simplify the process by which access to patented technology can be granted. The Patent Commons Project catalogues the patent pledges and covenants in a central location and facilitates their use by the development community and others, reduces the number of issued software patents that are a threat to open source and open standards, and documents the boundaries of the ''common area.''
About the Open Source Development Lab

OSDL - home to Linus Torvalds, the creator of Linux - is dedicated to accelerating the growth and adoption of Linux. Founded in 2000 by CA, Hitachi, HP, IBM, Intel and NEC, OSDL is a non-profit organization at the center of Linux supported by a global consortium of more than 60 of the worldÂ’s largest Linux customers and IT industry leaders. OSDL sponsors industry-wide initiatives around Linux in telecommunications, in the enterprise data center and on corporate desktops. The Lab also provides Linux expertise and computing and test facilities in the United States and Japan available to developers around the world. Visit OSDL on the Web at http://www.osdl.org/.

OSDL is a registered trademark of Open Source Development Labs, Inc. Linux is a registered trademark of Linus Torvalds. Third party marks and brands are the property of their respective holders.
OSDL

ISAKMP flow found, check if your IPsec VPN is on the list

According to News.com the Finns say that the flaw in the Internet Security Association and Key Management Protocol, or ISAKMP could lead to denial-of-service attacks.
ISAKMP is used in IPsec virtual private network and firewall products from Cisco Systems and Juniper Networks.

The security hole was so serious that the Finnish results were jointly issued by the British National Infrastructure Security Co-ordination Centre and the Finnish CERT to give it some weight.

Cisco and Juniper have acknowledged that some of their products are at risk. Cisco said the security flaw could cause devices to reset which could cause a temporary denial-of-service attack.

It is providing free software upgrades to fix the problem and has published a security advisory. The list of affected products includes Cisco IOS, Cisco PIX Firewall, Cisco Firewall Services Module, Cisco VPN 3000 Series Concentrators and the Cisco MDS Series SanOS.
Juniper products affected include all of its M-series, T-series, J-series and E-series routers, as well as most versions of its Junos and JunoSe Security software. A spokesJniper said that software issued on or after July 28 provided fixes for the flaw. The Openswan Project, which is IPsec software used on many Linux products, is also affected and the project has already released Openswan 2.4.2 in response to the advisory.

SOYO's Z-Connect VoIP Phone in TechLiving Magazine's 2005 Hot List

TechLiving selects 50 products each year based on ease of use, unique features and affordability. The SOYO Z-Connect Value Pack will be featured in the November/December 2005 issue.

The Z-Connect Value Pack is a hardware/software VoIP phone solution that consists of a pair of SOYO Z-Connect G668 VoIP telephones, which plug directly into a broadband DSL connection or cable modem for instant VoIP capability. No computer is necessary. With SOYO's companion VoIP service, calls are free on the Z-Connect network

"TechLiving's selection comes in the middle of the fourth quarter shopping season, and is a strong validation for the SOYO brand in the consumer electronics market," said Ming Chok, SOYO chief executive officer. "The increased visibility we gain from TechLiving's endorsement is critical in this market. Consumers routinely make purchase decisions based on recommendations of publications and authorities they trust."

For more information about the Z-Connect line of VoIP solutions, please visit www.soyogroup.com. A full list of TechLiving's hot list winners is accessible at http://www.techliving.com.
Some info on the phone;
Protocol Support
* Standard and Protocol
* IEEE 802.3 /802.3 u 10 Base T / 100Base TX
* Major G.7XX and gsm610 audio codec
* H.323 V4
* MGCP RFC2705
* SIP RFC2543
* Net2phone private protocol
* TCP/IP: Internet transfer and control protocol
* RTP: Real-time Transport Protocol
* RTCP:Real-time Control Protocol
* VAD/CNG save band with
* DHCP:Dynamic Host Configuration Protocol
* PPPoE:Point to Point Protocol over Ethernet
* DNS:Domain Name Server
* Telnet:Internet's remote login protocol
* FTP:File Transfer protocol
* HTTP:Hyper Text Transfer protocol
* Build in H.323 proxy
Hardware
* Data Memory—2MB SDRAM
* Program Memory—1 MB Flash memory
* Ethernet Jack—Two 10/100M RJ-45 jacks
* AC/DC adapter—Input AC 100-240VCOutput 9V DC, 0.89A
* Software
* DHCP support for LAN or Cable modem
* PPPoE support for ADSL or Cable modem
* Set phone by HTTP web browser (IE6.0) or Telnet
* Upgradeable by FTP
* Support major G.7XX and gsm610 audio codec
* Dynamic voice test
* CNG (Comfort noise generation)
* Dynamic voice jitter buffer
* G.168/165 compliant 16ms echo cancellation
* Tone generation and Local DTMF re-generation according with ITU-T
* E.164 dial plan and customized dial rules
* 100 entries for quick dial
* Adjustable volume for both handset and speaker
* Voice prompt

JOINT FCC/NARUC TASK FORCE LAUNCHES A WEB SITE ON VOIP 911 ENFORCEMENT

FCC released the following consumer information about new E911 website. Release text follows;
Washington, DC -- The Joint FCC/NARUC Task Force on VoIP 911 Enforcement has launched a new Web site to provide consumers, industry and state and local governments information about the rules that require certain providers of Voice over Internet Protocol (VoIP) services to supply 911 emergency calling capabilities to their customers. The address is www.voip911.gov.

The ability to access emergency services by dialing 911 is a vital component of public safety and emergency preparedness. VoIP service allows consumers to place a call like traditional telephone service; however, recent incidents in which consumers using VoIP service dialed 911 but were unable to reach emergency operators have highlighted a critical public safety gap. The FCC has taken steps to close this gap by requiring that, effective November 28, 2005, interconnected VoIP providers deliver all 911 calls to the customerÂ’s local emergency operator. Interconnected VoIP providers must also provide the customerÂ’s call back number and location information to the emergency operator if the emergency operator is capable of receiving this information.

FCC Chairman Kevin J. Martin said, "“Anyone who dials 911 has a reasonable expectation that he or she will be connected to an emergency operator; this expectation exists whether that person is dialing 911 from a traditional wireline phone, a wireless phone, or a VoIP phone. This new Web site will provide an easy way for consumers, industry and other government agencies to get the most current information on this important issue."


- FCC -

FCC Contact: Lyle Ishida at (202) 418-8240, e-mail: lyle.ishida@fcc.gov

NARUC Contact: Eddie Roberson at (615) 741-2904, e-mail: eddie.roberson@state.tn.us

Cisco and Microsoft tries to cool voip with ICE Methodology!

I saw the news first at this site, Telecommunications Industry News. Then I found the PR newswire news release. So here are both links.
PR Newswire at MSNMONEY Site.

ICE, a standards-based methodology, allows information workers and businesses to more easily communicate in media-rich ways across network address translators (NATs), a significant barrier to VoIP and video connectivity. ICE provides a rich set of solutions for current NAT issues with media. Microsoft and Cisco are jointly supporting the ICE effort, demonstrating both companies' strong commitment to developing standards-based communications solutions built on methodologies that can be broadly adopted and integrated.

Massive Voip push by Linksys / Cisco

New linksys products in the middle of the push!



Linksys One Services Router with 16-Port 10/100 LAN $1195.00
# Enables Linksys One voice product services including QoS, VLAN, SIP ALG and IEEE POE
# Provides auto-discovery and auto-configuration of Linksys One devices, including IP phones
# Two 10/100 full duplex WAN ports, two 10/100/1000 expansion ports
# Sixteen 10/100 full duplex, auto-sensing MDI/MDI-X LAN ports




Linksys One Manager Phone $299.00
A business IP telephone with color-display for Linksys One
communications solutions

# Simple, automated installation with Linksys One Services Routers
# High-resolution, color, backlit display, full-duplex speakerphone
# Two port 10/100 switch with ability to accept IEEE802.3af PoE
# Integrated call processing features with security, management, QoS





Linksys One Analog Voice Gateway, 1FXS, 1FXO $279.00

# Simple, automated installation with Linksys One Services Routers
# One analog phone or FAX connection, one analog connection to the public telephone network
# Accepts IEEE802.3af PoE from connected Linksys One switch port
# Integrated call processing features with security, management, QoS

Now what can I do with all this Cisco VOIP stuff I have? May be EBAY!
The news release is below.

IRVINE, Calif., Nov. 14 /PRNewswire/

Linksys One Product Line of Voice, Video and Data Solutions to Offer Convenience and Value to Small Businesses
Linksys®, a Division of Cisco Systems, Inc., the recognized leading provider of voice, wireless and networking hardware for the consumer, Small Office/Home Office (SOHO) and small business customer, today announced the initial line up of its new brand of complete hosted communication solutions for small businesses. This new solutions brand, named Linksys One, delivers business telephone service, data networking, security, applications, and the Internet through one high-speed connection from a hosted service provider.

The first Linksys One products to be offered will be the SVR3000 Services Router with 16-Port 10/100 Switch, PHM1200 Color Manager IP Phone, and VGA2000 Analog Voice Gateway. These products include unique Linksys One plug and play technology that instantly detects and configures Linksys One devices. As part of a true "Services Platform" capable of enabling growth with new features and applications, Linksys One products are designed to automatically determine the optimal configuration and are functional the minute they are connected, eliminating the need for skilled IT professionals on staff, or on site. A unique network discovery process and automated configuration also means that new networks, additions to the system, and new technologies require only limited-touch deployment. A 16-employee system with Voice over IP (VoIP) can be installed in less than an hour, and costs less than comparable key system or PBX solutions available today.

"Linksys One gives small businesses a single system for voice, video and data services," said Marthin De Beer, vice president and general manager, Linksys Small Business Systems Business Unit. "This solution enables service providers and value added resellers to position themselves as the single source for their customers' complete communication needs, offering them increased margins and new revenue streams."

SVR3000 Services Router with 16-Port 10/100 LAN

The SVR3000 Services Router with an embedded 16-Port 10/100 LAN Switch is the heart of the Linksys One solution. This integrated switch and router enables plug-and-play installation and includes advanced features that allow a small business to protect its network and improve productivity.

* Converged Services: The SVR3000 Services Router provides access to a
single converged network for voice, applications and data. In
addition to enabling VoIP, the SVR3000 hosts an email server, email
services, web filtering, file server, print server, firewall, Virtual
Private Network (VPN) technology, personal Web server, and much more
in future releases. It also provides access to the data and voice
networks, and supplies power to all of the attached devices, including
phones, using Power over Ethernet (PoE). All of this, plus integrated
voicemail and auto attendant, makes for a low-cost solution that
eliminates multiple boxes, costly IT resources, and licensing fees.
* Optimized: Integrated quality of service (QoS) capability in the
Services Router automatically prioritizes all network traffic. Staff
can browse the Internet, access applications, send emails and talk to
customers on the phone with no impact on quality. The Services Router
also has integrated Virtual LAN (VLAN) capabilities, allowing users to
segment services across the network for greater administration
efficiencies, security-control mechanisms, and bandwidth-management.
* Highly Secure: The Services Router provides hardware encryption and
robust security measures to protect against attacks on the network.
An integrated firewall and VPN technology provide robust means of
protecting business communications that take place over the Internet.
* Manageable: Service Providers or Value Added Resellers (VARs) can
control management access so that the small business network is
managed, optimized and easily expanded as a single system. This
eliminates the need to manage individual products, and allows for a
single interface to configure all functions of the system.
Capabilities to manage other services, such as calling plans, calling
features, applications, network security, and bandwidth requirements
are built into the system and provisioned by the Service Provider
through the SVR3000.
* VPN Access: In the future, it will be possible to easily connect
offices via a secure "always on" Internet connection or allow mobile
or traveling workers to access the company's system using a secured
connection over the Internet

The SVR3000 delivers 16 ports of fast Ethernet LAN switching with full support for IEEE 802.3af Power over Ethernet on each. All Linksys One IP phones and voice gateways can be fully powered over these ports without a separate power adapter. Two 10/100 WAN ports provide access to the Internet, either directly or through a separate broadband router or bridge device. Cascaded expansion to other switches or application servers is supported through two 10/100/1000 BaseT ports, enabling the system to be easily and cost-effectively expanded as a business grows.

Features:
* Services platform
* 16 port Powered Ethernet 10/100 switch
* Firewall, NAT Router
* DHCP, DNS Proxy
* Optional VPN client to service node
* Priority Queuing
* SIP Proxy/Registrar
* Power over Ethernet (PoE)
* SIP Application Layer Gateway (ALG)

PHM12000 Color Manager IP Phone

Combining the money-saving advantages of VoIP with the familiar features of a small business telephone system, the PHM1200 Color Manager IP Phone brings together the same quality of voice service and reliability that business users have come to expect from traditional phone systems with the cost savings and convenience of combining voice and data traffic over a single Internet connection.

The PHM1200 Color IP Phones can be installed in minutes -- simply plug in the phone, enter the customer information, and the business is ready to make and receive calls. There is no central phone system to install and no special servers to set-up. Linksys One Service Providers help ensure delivery of business class voice quality -- unlike the 'best effort' audio clarity often provided with other Voice over IP (VoIP) solutions.

PHM1200 phones will easily integrate with current and potential future Linksys One systems, such as video systems to monitor surveillance areas from the phones. The Linksys One application toolkit also enables the development of custom IP phone Extensible Markup Language (XML) applications and Web services.

Features:
* SIP based
* Dual USB 1.2 (optional WLAN i/f)
* 16M Flash and 64M Ram
* 24 Feature/line buttons, 4 soft keys, 1 Selector button, 3 fixed
feature buttons
* Pixel-based color/backlit screen
* Application developer XML engine
* Full feature set (within phone)
* Speakerphone and local vmail
* Web customizable (HTTPS)

VGA2000 Analog Voice Gateway

Many small businesses have traditional phone system equipment, like fax machines or analog phones, that need to be integrated into any new communication system. The VGA2000 Analog Voice Gateway allows Linksys One installations to include this legacy equipment. The Voice Gateway also provides a local connection to the public telephone network for emergency services calls and redundancy in case of a power loss.

Operation of a VGA2000 requires the installation, configuration, and operation of a SVR3000 Linksys One Service Router at the same site. The gateway interacts with provisioning, management, and security software hosted on the SVR3000 through a 10/100 port.

Features:
* Local PSTN access, 911, and Fax
* 1 FXO/ 1 FXS
* E911 support

Enabling Future Hosted Applications

Linksys One will provide a unique application toolkit that enables Service Providers to deliver business applications and web services directly to the phones of their subscribers. These can be their own applications or be from a third party vendor, and can be as diverse as instant messaging, real time stock, currency or traffic information, or even environmental controls. The application toolkit is based on the widely-used and platform independent Extensible Markup Language (XML).

As new compatible applications and services become available, Service Providers simply enable customer access, and add a new charge to the bill. Linksys One phones instantly detect and "learn" about new applications through Linksys One technology, so that subscribers can start using them immediately.

Roadmap for Linksys One Products

A complete line of innovative and affordable Linksys One communication products will be offered as part of its integrated architecture. These may include additional IP phones, VoIP Wi-Fi phones, analog voice gateways, network attached storage devices, wireless access points, USB devices, video conferencing, and video surveillance systems. Product specifics and pricing will be announced at time of availability.

Availability and Pricing

Scheduled for beginning in December 2005, the SRV3000 Linksys One Services Router, PHM1200 Color Manager IP Phone and VGA2000 Analog Voice Gateway will be available to Linksys One Authorized Resellers in the United States. Reseller participation in the Linksys Partner Connection Program is required. More information on reseller requirements can be found at www.linksysone.com. Hosted Service Providers including MCI, airBand, NeoNova, and IP Systems plan to be offering the line of Linksys One solutions in 2006.

Estimated street prices of the SVR3000, PHM1200, and VGA2000, in USD, are $1,195, $299 and $279 respectively.

About Linksys

Founded in 1988, Linksys, a Division of Cisco Systems, Inc. (Nasdaq: CSCO - News) is the recognized leader in Voice, Wireless and Ethernet networking hardware for consumer, SOHO and small business users. Linksys is dedicated to making networking easy and affordable for its customers, offering innovative, award- winning products that seamlessly integrate with a variety of devices and applications. Linksys provides award-winning product support to its customers. For more information, visit www.linksys.com.

Linksys is a registered trademark or trademark of Cisco Systems, Inc. and/or its affiliates in the U.S. and certain other countries. Other brands and products are trademarks or registered trademarks of their respective holders. Copyright © 2005 Cisco Systems, Inc. All rights reserved.


Source: Linksys

SnapVOIP crunching score!

As you know I do crunching for Einstein@home, Einstein@Home is a program that uses your computer's idle time to search for spinning neutron stars (also called pulsars) using data from the LIGO and GEO gravitational wave detectors. Einstein@Home is a World Year of Physics 2005 project supported by the American Physical Society (APS) and by a number of international organizations. The software is based on BOINC (Berkeley Open Infrastructure for Network Computing).Well I am happy to be a Berkleyan.
I run it on a one slow pc by today's standards. But see that I am above average!. Join and have fun!


Statistics -> Einstein@home Hold Charts -> snapvoip

The amount of work done by snapvoip, broken down according to various criteria. These pages are updated every four hours.

General information
Member information
Member list

snapvoip General Info:
Name(and url) snapvoip (teamid 2754)
Created 2005-05-07 06:36
Total 12122.931409 Cobblestones
#1465 of 3546 teams (more than 58.686%) - Charts
Recent 61.295772 Cobblestones
Country United States - #559 of 1219 teams (more than 54.14%)
Members 1

Stop Skype, Research group tells its members.

to remove, stop or detect skype, read this article.

Info- Tech Research Group,Technology industry analyst firm is advising Corporations to ban Skype use. I reported about Skype network misuse, VOIP IP Telephony: Skype Seems to be hazardous to your network!.

"Companies that are already banning peer-to-peer applications, such as instant messaging, should add Skype to its list of unsanctioned software programs," says Info-Tech analyst Ross Armstrong.

"Approximately 17 million registered Skype users are using the service for business purposes," says Armstrong. "Unless an organization specifies instances where Skype use is acceptable, and outlines rules for client-side Skype settings, that's 17 million opportunities for a hacker to invade a corporate network."

In a research note prepared for Info-Tech Research Group members, Armstrong outlines five reasons for an enterprise to ban Skype:

- Skype is not standards-compliant, allowing it and any vulnerability to
pass through corporate firewalls.
- Skype's encryption is closed source and prone to man-in-the-middle
attacks. There are also some unanswered questions about how well the
keys are managed.
- Enterprises using Skype risk a communication barrier with countries
and institutions that have already banned the service.
- Skype is undetectable, untraceable, and unauditable, putting
organizations that are subject to compliance laws at risk.
- The question of whether VoIP calls constitute a business record is a
legal quagmire. Throwing Skype into the communications mix further
clouds the issue.
Comments Armstrong, "The bottom line is that even a mediocre hacker could take advantage of a Skype vulnerability. If you are going to use Skype within enterprise, manage it as you would any other IT service: with policy and diligence."

Tags: remove skype, stop skype, detect skype, skype, snapvoip, voip ip telephony

VOIP not so Free in india!

After reporting that VOIP IP Telephony: VOIP set free in India! I bewidered to see this article "Police bust international telephone call racket, arrest five", reading the article, one realises that they were providing a service based on VOIP. May be they will be set free under new law.

Video conferencing in Supercomputing 2005

Video conferencing has been a fixture in visions of how we imagine the world of tomorrow web cams nor most of the corporate Video and Voip Conferencing gives the real picture of the other side. It is mostly a face in a tiny window. The Voice part has improved due to the advance in VOIP (Voice over IP) developments. But Video over IP or VVOIP? is in need of advancement.
I want the Otherside on a say 50 inch TV! Well not! Unless you are the Director of Star Wars movies!!.
Now I am offered "Shared Spaces" on three 65-inch plasma displays.! Well now where can I put them on? is the question.
Then I found this on ASCRIBE news letter that The McGill University team is competing in Seattle on Tuesday, Nov. 15, at the Suprcomputing 2005. I was wondering where to publish this. Here or on my Grid Technology Blog, Well I will do it here and publish about Suprcomputing 2005 on the other site. Update: it is there now! gridtech: Supercomputing 2005 is in Session

From the announcement the McGill University team says;
"Modern video conferencing hasn't worked well as it doesn't allow you to interrupt one another and has never managed to support the quality of interaction that people experience in real life. We wanted to change that," says John Roston, director of Instructional Multimedia Services at McGill University.
"Our technology provides a life size, high definition view on a large panoramic screen, which gives users the impression that they're talking to people in the same room with a window between them," adds Professor Jeremy Cooperstock of the Department of Electrical and Computer Engineering.
Roston and Cooperstock, members of the Centre for Interdisciplinary Research in Music, Media and Technology, see the new technology being used in business, education, health care, and many other contexts. They are at the leading edge of a community of researchers who are out to change the current video conferencing experience by allowing people in different cities to feel as though they are in the same room together.

To show what they can do, Roston and his colleague Jeremy Cooperstock have arranged for jazz conductor Gordon Foote to teach his music students at McGill all the way from Seattle. Foote will conduct them in real time, coaching and guiding their performance as if they are in the same concert hall.

The event will take place as part of Bandwidth Challenge 2005, an elite annual competition at which nine teams of top scientists will showcase exceptional uses of new technology. The challenge will take in Seattle as part of Supercomputing 2005 where world experts from academia and industry, including Bill Gates, gather to discuss high performance computers and networking.

"Shared Spaces" is only one of several applications of new technologies being employed by the McGill Ultra-Video conferencing Research Group. The Group draws together top people from diverse disciplines who use technology to understand and enhance human experience. The group's projects also include a live undersea high definition video camera and remote sign language interpretation for the deaf.

The McGill team is competing in Seattle on Tuesday, Nov. 15 from 5-6 PM (PST). Find them at booth number 6017. Learn about their technology here.

VoWiFI phone

Perhaps to use with VOIP IP Telephony: VG2211i Wireless/VOIP (VoWifi) Personal Gateway
The VoWiFI phone VM1188T is designed for users who need mobility of WiFI and the cost saving of VoIP. The VM1188T is intended to help users make SIP calls without the hassle of turning on your computer. The compact and elegant design of VM1188T allows users to make VoIP calls at work, home, and on campus with POTS like voice quality

The modulation of VM1188T is based on the WiFi standard; it is compatible with almost all standard 802.11b/g access points. With the VM1188T, users can access to both PSTN and SIP networks as long as there is a trunking gateway available at the CO side. With the easy startup, users don’t have to remember the SSID or security key of each AP, the VM1188T stores these configurations and will connect to the access points automatically when boot up. Paired up with the cradle access point CAP2316A, users can enjoy the benefit of interference free wireless environment through the implementation of MSSID.

With Accton’s VM1188T, users can enjoy the benefit of cost reduction with SIP, mobility with 802.11b/g WLAN, and hassle free installation with easy startup. The VM1188T is pact with user friendly features all in a single device.

VG2211i Wireless/VOIP (VoWifi) Personal Gateway

The Accton VG2211i integrates the functions of wireless, gateway and VoIP into one fashionable and compact device. Users can select one of four wireless modes to meet users own specific requirement when in different working environment. Users also have the option to use the RJ-11 ports provided by the personal gateway for VoIP calls and life line function.

With built-in 802.11b/g module and internal antenna VG2211i can operate in either wireless client, access point or repeater mode. Fully compatible with IEEE 802.11g and backward compatible with 802.11b, this is the the future-proof device.

The VG2211i includes 1 FXS and 1 FXO interfaces with RJ-11 connectors that can be connected to telephone service for VoIP and POTS service. Once connected to an analog phone through the FXS interface the VG2211i can utilize VoIP services provided by the service provider to make VoIP calls at lower cost. Users can also use the FXO interface to connect to PSTN for normal telephone calls or life line function.

The VG2211i can provide secured and personal wireless space with the add-on benefit of VoIP for business travelers on the road to keep connected with family, friends and co-workers.

geemodo: Spyware spread through Bloggers!

geemodo: Outfit Used Unsuspecting Bloggers to Spread its Malicious Code and FTC comes to your rescue!