text:
Archive for the 'Thoughts' Category
[Updated] Blizzard makes plans to conduct largest internet social experiment ever
Update: Man, that was fast. As a friend suggested, maybe this was a PR move. "Now that we aren't going to do obviously outrageous thing, here's what we'll do instead…" We shall see.
Blizzard's Plan
If you are any sort of MMO player or you know one, you probably have already heard about Blizzard's plans to use what they call RealID. The basics are this:
- If you use the system and friend anyone else, they can see your real name. It's not based on character or server, it's your unique ID with your real name tied to it. No more anonymity in Blizzard games, basically.
- It will also allow you to do as Steam does now, track people across games. It's an external friending service, rather than a friend system in a specific game.
- Some other stuff Blizzard totes as really helpful.
The reality and the why
Here's the part they don't mention on that link above…yes, it's completely optional, but if you want to post on any Blizzard forum, you will have to use your RealID, and hence, your real name. Not to mention once the system gets started and guilds start to require it, as we all know they will, you'll be giving it up to people you often only known in an online capacity. Why play with someone anonymously, when you can play with them where they have some accountability. It'll sweep the entire game like wildfire. There will be holdouts of course, but for any serious (read catass) WoW players, they are basically seeing the writing on the wall of having to now be out, so to speak.
Why is Blizzard doing this? Well, if you don't know, public forums are really hard to control (great, long article). And Blizzard has one of the largest, if not the largest, gaming forum online because of how popular WoW is (most popular MMO of all time still). As you can see from this very scientific documentation, people online aren't always 1) aware they aren't anonymous and 2) able to control themselves when anonymous. It's hilarious and sadly true.
Internet Assholery (or IA) been a big problem for online forums of all types pretty much since the beginning of the internet. There is some disconnect with many people when they think there are no repercussions for their actions. This guy can probably explain it better, and even offers up how he'd like to track the changes.
The reasons no one has done this before
Well, let's start with some simple examples. Since Blizzard announced this, a few people defending it have offered up their real names. This guy and this guy have both probably thought better of it now. You see, you aren't actually anonymous anywhere. You aren't in real life obviously, but you aren't on the internet either (well, 99.9% of us aren't). Everything you do online can be tied back to you in one way or another. Even without your real name being given out; that just makes it easier. Look at both of the examples above. The people tracking down these guys used their real names to get started tracking them down, but if they had any other piece of information, or postings where you shared pretty much anything, or posted anything, you can be tracked. Just because you use a fake name on a forum, don't think that means that people can't find you. There are a thousand stories of people online being tracked down, well before Blizzard ever even thought of this idea or even running an MMO. It's a fact of internet life most people don't realize.
Ok, so we aren't really anonymous. Well, we aren't really anonymous once we start engaging. If we post on any forum, any page comments, facebook, etc, not only is that all trying to be tracked by your ISP, it starts to link information to your fake online identity. Which has an IP address, which has a real name attached, etc, etc, etc. So what's the problem then? Well, because people feel they are anonymous, they act as stated above AND because they aren't particularly well informed, they believe other people are anonymous. So things like this don't happen as often as they could.
For others, the big concerns are what are going to happen to the groups of people that really needed the anonymity. For instance, women and/or minorities. No longer can they go online in WoW and join up to a guild or post in forums without people knowing their ethnicity (many names alone suggest this, sometimes as wrong as not) or their sex. Why is this a problem? Well, as you might guess, many gamers are socially retarded. I don't say that without some self reflection, as a gamer, but I also say it as someone who thinks that invading people's personal lives is well beyond something I'd condone or do. For many others, it's just a matter of access to the information. Without the it-takes-more-work-than-its-often-worth-to-figure-out-someones-real name barrier, this sort of thing may happen more. If you've ever gone on XBox Live, you're probably aware of how idiotic people can be online.
To boot, who would be responsible for all of this in a court of law? Some might argue that it's the company that requires people to give up their anonymity to post on their forum or to be in any of the more serious guilds (I state this as fact, but it could turn out otherwise). Blizzard may very well see lawsuits based on their plan.
Why it's a potential good thing in the long run
Didn't expect that title, did you? Well, just think about it for a second. If people have to start thinking when they post online, this could be tied back to me, maybe, just maybe, they will stop acting like assholes. Maybe it will sink in that they aren't anonymous and that they should have the social grace to handle themselves like they were talking with actual people. Hey, it could happen. As a matter of fact, I think it will happen in the long term. And if this social experiment works for Blizzard, expect more and more companies to do it. The internet isn't going to directly expose you in the future, you are. You will have to sign up for and validate your identity at every site you want to post on. And you will be banned based on your real identity. No more just signing up another account. And once multiple sites start requiring real names, they will all start using the same service and to post, you'll have to give up a link that shows your entire internet posting history. For some, this going to be a HUGE change. For others like myself, that have been posting under their real names for a long time, it won't be such a huge thing. But notice I don't directly state my full name anywhere on this site, though any enterprising person could easily track me down. I like that layer of separation for my personal site. Heck, many people that read here do so because they know me in RL, so it's not that big of a deal, but that layer of obfuscation, that bit of work required to track me down, gives me at least a bit more of a wall between the world and my personal life. That I'm sharing for anyone in the world to read… Ya, dumb, I know. But that's the whole point. Many people aren't even aware that the anonymity is a lie.
What Would Matt Do: I'd wish Blizzard all of the best. I like the concept and I eagerly await details of how it goes. I'd love to see the things that Jamie mentions on his site above (povg) so we can get real metrics on how this social experiment on a grand scale goes down. Here's hoping Blizzard not only shares that information at some future date, but that they realize they may be setting a trend for the much of the future of the internet and go…wisely into that great beyond.
Update: Jay and Silent Bob thought about this a long time ago.
Update II: Lots of people are angry. I'm not sure I disagree with them even. It's a very tough issue.
[edited for grammer. probably more than once.]
What Would Matt Do II
(this post goes out to one Cap'n John. Keep on keeping on, my friend.)
Well, I made it. I did leave FL, even on the day I said I would (early March). I got up, finished loading up my car, said tearful goodbyes to all of my family. Got in the car and drove out. I had my trusty atlas and my iphone and a destination. Just as a note, I think of a lot of this from my perspective, but without the help of everyone involved, Josh and his family, my Mom, my brothers and most especially, my wife, none of this would have happened. To put an even finer point on it, wherever I say "I did this" or "I did that", I really mean, my wife and I did it. She supported me on everything and listened when I moaned.
The Trip
The atlas and iphone and some clothes were pretty much all I had (and my laptop of course). I didn't really have a firm plan other than get there and make it work. I did get there, it took about eight days total, but if I was in a big hurry, I could have probably cut a day off. I drove I40 across the states until I ran into California. I saw a lot of states for the first time and couldn't resist camping in the snow in Flagstaff. You should have seen the look on the KOA guy's face when I said I just wanted to throw up a tent and didn't have an RV. He actually had to ask his boss if that was possible. Thankfully she knew how do it and we all chuckled about them not having a non RV camper in ages.
A couple of nights later, after having talked with my sister on the road one night, I was sleeping on a hippie compound in CA. They have a big complex setup there that they rent out and support. My sister has some long lost family there and they offered to put me up for the night. Not only did they do that, they made a delicious breakfast and we all talked the morning away (after I got up butt early and walked the place. It's all cedar redwood forest on top of a mountain back in the middle of no where). They said don't forget to check out Whidbey Island while you're up there and promised to visit.
Stayed at my cousin's the next night, which was great. Great last stop on the journey. The next day (Monday) I arrived at my friend Josh's house.
Getting a Job and a Place.
Getting a job turned out to be pretty easy. Much easier than I thought. I had already been sending out resumes, but most people didn't want to hear from someone that wasn't already in the state. That was dumb of them, because I was literally applying in three or four states. The day after I got there, a company flew me out (arranged while I was driving across the country) to Idaho for a job interview, put me up for the night and flew me back the next day. Then I interviewed constantly for about 4 days. Had to turn down a couple of offers that were birds in the hand, but I thought I could get better in the bush. I did snag a good one out of the bush and had an offer on the table in about a week after arriving. I started in late March.
I did all of this from my friend Josh's place. Without him and his lovely family I wouldn't have been able to even consider doing this. They put me up, helped me eat (I had almost no funds really) and helped in any way they could. Once I got the job, I started driving back and forth from his house to the job, but it was almost three hours each way, so I ended up renting a room in Oak Harbor (south of Anacortes where I got a job and on Whidbey Island).
Then I started the process of trying to find a house (after a couple of paychecks, so I could actually afford it). My wife and I started searching everything we could find. I looked at so many houses! I started to be able to see just from the ad if it was a Facebooked job or I should at least give it a drive by and then maybe a walk through. I was being picky, but I had time and it was necessary. Just about the time we were going to have to settle for a house, we stumbled across a GREAT place in Oak Harbor of all places. Lots of space and some quiet. Can't wait to move in.
Now, for me, it's just waiting. I'm working and waiting…while my wife and kids finish the packing, sell the car and fly out here (and a million other little things). They will be here in two weeks, right after we get the house. It's finally happened.
When it's all said and done, it will have been almost exactly four months to the day since I've seen my wife and kids in person. I can't wait. I've always been very aware of how lucky I am to have them, so this wasn't a wake up call, but more an affirmation of the idea that I belong with them and they belong with me.
Worth it?
Completely and totally. I'm without a lot of stuff I've recently taken for granted (volleyball group, D&D group, my extended family, beaches for swimming and so on), but with so much more. I've been up the top of the mountains in the Cascades and in the Olympics and climbed the two highest points here in Anacortes. I've been to the Oregon Coast and hiking in the foothills in Oregon. I've seen so much already. This adventure is JUST starting and I'm incredibly excited.
My faith, that my family and friends and I could work this all out, was greatly rewarded. I've got a better job, a better place to live, three or four states to explore, family coming soon, so much to do!
P.S. – probably my last true blog post for awhile. Next up, change the world.
Dear Mr. President
Dear Mr. President,
I'm going to ask you to do something now. Listen. Not because I'm smarter than anyone else or because I'm more important. But because I'm not those things. I'm just a guy. Who thinks you might want be a better president than you have been so far. I'm sharing because as song goes, this land is your land, this land is my land.
First, please take over the BP operation right now. I don't mean stop BP and bring in your own people, I mean use your executive power and take over the corporation, at least temporarily. Use the fact they are criminals and have committed crimes here and now. Or however you need to phrase it. If you can do it with banks, you can do it with oil companies. Use their resources properly, aimed to stem the damage and clean it honestly, not white wash it with chemicals that may be worse than the original spill. Make them start giving you the truth. Then form a council. Invite not CEOs of oil companies, but the most experienced engineers in the field, experts of oil rigs and plugging of leaks underwater, long time navy guys who know about deep pressure, environmentalists who've worked with oil spills before, and more. Don't just talk to other countries, ask them to send their topmost experts. Then have them come up with solutions. All you have to do is ask, people will come help, again, at least temporarily. And the longer you wait, the more BP destroys livelihoods, environments and people.
Second, please start being serious about fixing our oil/coal addiction. I know big corporations with lots of money and power are aligned against that idea AND have funded your presidency. So what. Fight the power. There are many ways to approach this. I'm not an expert in these things. But you know what? Some people are. There are scientists that have great ideas to completely change the way our country runs and the energy it uses. Maybe you could form a council here too. A longer term bigger thing. A scientific council established for the betterment of the country. Hire them. Hire the best people in academics and in the industry. Pay them richly. You could hire then guys, pay them a couple of million each and it would cost less to have them for your whole presidency than it would to build one fighter plane. While you're forming dream teams, the Ring of Fire guys would be great.
Third, and this ties into the above one, please employ the country. Yeah, politicians will complain. So what? If you use the council created above and get a plan to overall the entire energy grid and power our country off say the deserts in the south because the grid can now transfer energy anywhere…guess who can build that? Our people. You can train them, you can employ them, you can enable them. Not give a handout, put them to work for the betterment of all. People will come in droves. You will educate them in a field and empower them to provide for themselves. And they will love you for it. You'll be fixing the country and helping the people. That's just one way. There are so many different ways you could take the 10% of population of who a large amount actually want work and put them to working to change how our country works. With a real energy grid (not a magical one, other countries have done this), we could be off oil and coal and nuclear in 20 years maybe.
That's not all, but that's a start. Start thinking of the people. If you start acting from that place, from the place of what's the best thing for the people, not for the other politicians or for just your voter base or for anything but the people, you'll start making decisions that will make us love you…instead of just being happy your not as bad as the last guy. In less that's all your looking to be, not as bad.
I hope this finds you and yours well, and I truly hope you listened. I voted for you. I didn't believe your words, but I knew you'd be better than the alternatives. But a part of me was really hoping you were the man you gave speeches like. You can still be that man.
Thank you,
What would matt do?
You know what I'd do? I'd travel completely across the country to see if I can't find a better life for me and mine. I'd go with almost no money, with some help from my friends, and hope for the best.
Well, that's a good question. And the answer is because nothing else fits.
I've been living in FL for another 11 years (I grew up here and then moved back) and I don't like it anymore. I don't like the economy build on the flesh trade (tourism), I don't like the angrily-waiting-to-die mentality many have here. I don't like that I can't find many like minded people. Mainly though, I want my family and I to live in a place that brings joy every time we go out in it. I want to visit more places than different swamps or beaches (though, it's hard to get too upset about beaches). We want mountains, snow, hiking, camping, kayaking, etc. Heck, I'd like it if it didn't take a clean 6 hours just to get of the state I live in. Don't get me wrong, FL has some great parts, and some great people, but it's time to move on.
Here's hoping I find all of that and more. Or at least I find a job and way to get by for a bit longer so I can plan something else. Because not having a job is great, but not having money just doesn't seem to work with the rest of society.
So what would I do? I'd move. I'm leaving in 5 days. Me first, and then when I get some funds and a place, I'll fly the family out. I admit, I'm a bit nervous that everything will work out. I do have faith.


text:
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subset: commoncrawl


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Joi Ito wants to revolutionise the internet. His vision is of a world unperturbed by a complicated, costly and outdated copyright system, where everybody can collaborate and share content on the web as they wish.
He doesn't want to get rid of copyright, he just wants you to be able to adapt it to your needs. He wants to build a sharing economy.
Ito is the CEO of Creative Commons, the nonprofit organisation that's creating and overseeing the use of free-of-charge licences and tools to ensure content creators can easily mark how they'd like their works to be used online.
Its collaboration with the World Wide Web Consortium could have a significant impact on how we deal with rights on the web: it could automate the licensing and make it machine-readable. "We work very closely with the W3C to create RDFa, which stands for Resource Description Framework in attributes," Ito explains.
"It allows you to put Creative Commons Rights Expression Language, or ccREL, inside of HTML and XML, so anyone can mark each object with the copyright and other attributes associated with the object directly in the markup. It will help greatly in searching for open content, doing proper attribution, allowing people to copy and paste stuff directly from the web and making it easier for software to keep the rights metadata attached.
"Our hope is that it will be widely adopted by people who make services, tools and content for the web. For web design this is really important. You used to have to put Creative Commons in the comments page, which is really stupid. Now you can wrap each object with the rights, and we'd love it if web designers started using valid RDFa to express the licences."
Already Slideshare is using RDFa to express its Creative Commons licences, as does the Creative Commons License Generator, but Ito stresses that RDFa can be used to express more than just copyright.
And so other key adoptors include Facebook (which uses RDFa for its new Open Graph protocol), Google, Tesco, Drupal and the White House, which has announced plans to make increasing use of RDFa (check out rdfa.info, wiki.creativecommons.org/RDFa and the RDFa primer at www.w3.org/TR/xhtml-rdfa-primer). However, it's still a recommendation and there's no news on when it will be included in the spec for HTML5.
Ben Adida, chair of the RDFa Task Force at the W3C, claims that "for all practical purposes, RDFa is already part of HTML5" and that it "probably doesn't matter" that the theoretical discussion around which spec it's included in, core HTML5 or an add-on, is still up in the air. However, HTML5 editor Ian Hickson recently admitted that RDFa still seems to have "some pretty serious problems".
Be interoperable
Today, about 350million pieces of content are licensed under Creative Commons in more than 50 jurisdictions. The biggest part, over 135million images, is taken up by Flickr. Other high profile users of Creative Commons licences include the White House, Nine Inch Nails and Al-Jazeera, which has put all of its Gaza footage under Attribution only. And last year Wikipedia migrated all its content over to a Share Alike licence.
"Wikipedia existed before Creative Commons," Ito says, "and they were using the GNU Free Documentation licence from the Free Software Foundation, which wasn't perfect for wikis because it's meant for printed books.
"The biggest problem was that it didn't have interoperability, so you couldn't mix it with Creative Commons Share Alike and then relicense it, because both require relicensing under their own licence. So it took about four to five years, and we got the Free Software Foundation to agree a time limited window, where wiki projects could convert from GFDL to Creative Commons. And then there was a board vote, and then a whole community vote at Wikipedia to decide whether they should switch. It took a very long time!"
In the past Ito had to convince people to use Creative Commons licences, but the opposition he previously experienced is waning.
"There are many new businesses using Creative Commons and for some it's a central part of their business model. We're not 'anti-business'. Like the internet, Creative Commons creates interoperability and lowers friction.
"This causes some businesses or business models to become less relevant. This is true for just about any technology and important social advance. I believe that business will overcome its own opposition to Creative Commons or that business will perish."
The biggest challenge now is to get people to use Creative Commons licences correctly. "It's like writing valid HTML – most people can't do it," says Ito. "It's important for us to try to get key players like the White House to use Creative Commons and RDFa in a very valid form.
"We're really trying to focus on being an infrastructure player and on making sure that the legal and technical stuff works. Because 99 per cent interoperable is like zero! We need to make sure everything is precise so we spend less time advocating and more time executing. I'd like Creative Commons to become so standard that everybody uses it and we're just the custodian to make sure it doesn't fail."
Ito believes the potential of the non-profit organisation for business, society and the environment is massive.
"Every layer – Ethernet, TCP and World Wide Web – created an explosion of innovation," he enthuses. "It created start-ups and non-profits and all kinds of socially beneficial disruption. I think that Creative Commons will also mark an explosion of innovation that will happen on the content level. As an investor, for instance, I can't invest in many of the music companies because of the risk of copyright litigation. And there are a lot of really interesting sharing business models that just can't happen because Creative Commons isn't widely adopted yet. Once it is, there is a whole bunch of neat things that can happen."
You're inclined to believe Ito. He's one of the most influential people on the web. His LinkedIn page lists no less than 35 jobs and involvements. One of the companies he's most passionate about is music streaming venture thesixtyone.
"It's looking at music from the perspective of the fans and independent musicians, not of the distributor or the record label like most other music sites," Ito says. "This is really a place where the musicians participate and interact with the fans."
Getting funding
For a prolific investor like Ito it must be frustrating having to go around and beg for Creative Commons funding. The organisation still has no real business model.
"We kicked around ideas like citations," says Ito, "which is a big problem for Wikipedia, for instance. People take a picture from Flickr, post it and then the original photographer will change the licence and tell Wikipedia to take it down. Creative Commons licences are perpetual – we don't want people to change the licence. It's important to be able to take a snapshot and say 'This content was available under this licence on this day'.
"Things like that may have value and may be something that we would build a business around, but we're still thinking about it. It would be great to have a revenue stream, because personally going around and doing fundraising every year is a lot of work.
"On the other hand, we always want to make our licences available for free and have the widest adoption possible. There is an inherent difficulty in coming up with a business model when you're trying to give everything away."
Ito continues to travel the world to talk about open innovation. One of Creative Commons' most recent tools, CC0, enables a copyright holder to waive all their rights, including attribution. It's as close to public domain as possible and already having a big effect on how science and education work.
If Ito's vision becomes reality, the culture of sharing made possible by Web 2.0 was just the first step.
Article continues below


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The August IxDA Cincinnati event
will feature a presentation on Gathering Insights: using research to
understand your customer prior to engagement and how to apply those
learning within the project work.
About Our SpeakerJennifer King is a Senior User
Experience Architect in Consulting Services at HP. She brings over 15
years of operational experience in the information architecture
discipline with 10 years specifically working on automotive website
operations and user experience. From 2001 to 2006, Jennifer worked on
all consumer facing websites and online media campaigns for Ford
Division, including fordvehicles.com,
the primary website for consumers researching Ford vehicles, with over a
million unique visitors a month. At EDS and HP she has worked
extensively on project work in addition to sales pursuit opportunities.
She brings demonstrated expertise in applying information architecture
principles and usability findings to improve user experience and
Agenda5:30-6:15 pm: Networking, exchanging of ideas, business cards and refreshments
6:15-7:15 pm: Introductions and presentation
7:15-7:30 pm: Informal discussion, networking
LogisticsHP will host the event in their
Cincinnati Office, and refreshments will be provided. For parking,
several garages are loated within a block of the office. Attendees will
need to arrive by 6 pm so security can let them in. We will be gathering
in Room TER-115.
About Our HostHP is a Fortune 500 companies in a
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green screen to responsive design. From crash-test labs to airports, our
team is designed to solve your problems.
You can register for this free event at http://www.eventbrite.com/event/7680766379


text:
Source: Central Statistics Office
Just over 6,000 days were lost in 2007 due to industrial action. This was the lowest figure ever. This chart shows both the number of working days lost and also the number of people involved by year. As the workforce grows the drop in days lost to industrial disputes is even more significant. In 2007 the workforce in Ireland hit 2.1 million.


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Syncro Soft SRL, Craiova
Informaţii detaliate Syncro Soft SRL, Craiova
- Nr. de înmatriculare
- Număr de agajați
- 11-50
Descriere Syncro Soft SRL, Craiova
SyncRO Soft is a software development company located in Craiova, SW of Romania. We develop software products and offer IT services for companies. The company was established in 1998 and has significantly extended the business since. As a pro-active customer-oriented company, our goal is to achieve profitability through productivity. Yet another driving force is quality, which is ensured by establishing and maintaining quality standards throughout the organisation. You are welcome to visit our office and we are open to any suggestion, comment or idea.
Our specific objectives are: Develop new software products and tune our IT services to clients' specific needs - SyncRO is preparing to launch more new products and services in the near future. An increasing emphasis will be put on providing XML based software solutions. Establish international partnership with outside software companies in order to mutually promote our software products. Find reliable partners to distribute and offer our products and services on outside markets. A special partnership program will be developed.
To maintain and improve customer satisfaction levels. We intend to have a long list of happy customers, who agree that we have provided outstanding products and services, and have saved them time and money. We will supply a full support centre for those using our software providing high quality technical support assistance. In our constant search for challenge we are continually increasing our knowledge base, and expanding the range of services we offer. We believe in being flexible, pro-active, customer-oriented software professionals, who go all the way to meet customers' needs.
Cuvinte cheie:Servicii It, Dezvoltare Software, Software Development
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Passerby blocks traffic to protect injured manUPDATED 1:46 PM EDT Apr 18, 2013
It's unknown how the young man was injured, or why he was lying in the middle of Paulson Avenue in Lincoln-Lemington. But he may have suffered further injury, if not for the protection offered by a stranger who waited for Pittsburgh police and paramedics to arrive.
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|Publication number
|Publication type
|Application number
|Publication date
|Feb 23, 2010
|Priority date
|Dec 3, 2002
|Fee status
|Publication number
|Shawn W. Smith
|Original Assignee
|Global Ip Solutions, Inc.
|Export Citation
|BiBTeX, EndNote, RefMan
|Patent Citations (32), Referenced by (23), Classifications (11), Legal Events (5)
|External Links: USPTO, USPTO Assignment, Espacenet
This application is a continuation-in-part of the co-pending application for Continuous Bandwidth Assessment and Feedback for Voice-Over-Internet-Protocol (VOIP) Comparing Packet's Voice Duration and Arrival Rate, U.S. Ser. No. 10/065,951, filed Dec. 3, 2002.
This invention relates to voice-over-Internet-Protocol (VoIP) systems, and more particularly to adjustments of current bandwidth usage of VoIP channels on an unregulated network such as the Internet.
Traditional applications such as telephone calling can now use the Internet rather than traditional telephone networks. Voice-over-Internet-Protocol (VoIP) applications capture a user's voice, digitize and compress the voice, and transmit the coded voice as data inside Internet-protocol (IP) packets that can be sent over the Internet.
VoIP applications can be installed on personal computers (PC's), other devices connected to the Internet, or on translation servers such as Internet-to-Telephone gateways or Protocol Converters. Each party to a call runs a local copy or client of the VoIP application.
VOIP application 12 receives these IP packets 20, extracts and de-compresses the voice data, and plays the voice as audio to user B. User B's voice is then captured, coded, compressed, and fitted into IP packets 22 by VOIP application 12. IP packets 22 containing user B's voice are also routed over the Internet back to VOIP application 10 for playback to user A, achieving a full-duplex voice call.
IP packets can be routed over a wide variety of paths using the Internet. Indeed, the de-centralized nature of the Internet allows routing decisions to be made at a number of points along the paths between applications 10, 12. The paths taken by packets 20 in the A-to-B direction can differ from the path taken by packets 22 in the reverse (B-to-A) direction. For example, packets 20 may pass through intermediate routers 14, 16, while packets 22 pass through router 18. Such non-symmetric routing can produce non-symmetric routing delays and challenges for the VOIP system.
Various network problems may occur. A router may temporarily fail, causing some packets to be delayed or lost entirely. The number of arriving packets may suddenly jump, producing congestion such as at router 18. Router 18 may delay packets 24 while the increased packet load occurs. Packets may continue to be delayed after the initial failure is fixed as the packet backlog is worked off. If the input buffers for router 18 overflow, packets 24 may be dropped or lost rather than simply delayed.
Bandwidth limitations may also occur. Packets may need to reach a user through a low-bandwidth dial-up modem line. Some types of modems such as satellite Internet Access or Asymmetric Digital-Subscriber Lines (ADSL) may have much more bandwidth in one direction (download) than in the other (upload). Occasional interference may further delay packets. The modem user may send email or browse a web site, reducing further the limited bandwidth available to the VOIP application's packets. Thus bandwidth limitations may be both permanent and temporary.
VOIP applications 10, 12 may have limited network status communicated between them. Sending VOIP application 12 may be unaware of packet routing problems on the packets it sends. The problems may not exist in packets received by VOIP application 12 from VOIP application 10, as the routing paths may not be symmetrical. Even on a symmetric network congestion or limitations on bandwidth may exist only in one direction, such as upload and download directions on a cable modem or ADSL.
VoIP application 10 cannot determine its outbound bandwidth simply by looking for delays of incoming packets received from VoIP application 12, since different routes may be taken by packets 20 sent and packets 22 received by application 10.
During initialization of a call between applications 10, 12, some provisioning may be performed to determine the initial bandwidths available between applications 10, 12. Such provisioning may be similar to fax machines that negotiate compression standards used and bandwidth or baud rate for each call. However, this initial provisioning is often not continuous. Changes to the Internet that later occur during the call are not detected once provisioning is over and the call is started.
In this example, packet 2 is delayed slightly, causing a gap to occur between the end of playing the voice for packet 1, and the start of voice play for packet 2. A larger gap occurs between packets 2 and 3, between times S2 and S2′. These gaps may be filled in by interpolating voice data, or by adding silence. However, the pace of the user's voice may seem uneven or jerky due to such gaps.
Of course, all voice could be delayed by a large amount, such as 5 seconds, to allow for late packets. However, this requires a larger packet-input buffer and would greatly increase the delay or latency that the user hears. This delay may be noticeable to the user and annoying. Full-duplex conversation becomes impractical as the delay grows to several seconds. Thus the input buffer has a practical size limit, and packets cannot be delayed for too long. The quality of a full-duplex conversation is better when buffering and delays are kept as small as possible.
Such gaps caused by delayed packets can reduce the quality of the voice played. When a temporary interruption occurs along the path taken by the VoIP packets, packets may pile up in buffers near the point of interruption. Should service be quickly restored, the stored packets in the buffers may be sent after some delay. However, longer-duration interruptions can cause router buffers to overflow. Packets may then be dropped or discarded before reaching their destinations.
Once the interruption ends, the older packets are likely to be sent first by the router. Newer packets may be delayed even after the interruption ends as the backlog of packets is transmitted. Thus stale packets of older voice data may be delivered before more current voice data. These older packets may already be too old to be played, resulting in a lengthening of what was a brief moment of congestion.
Some VOIP systems may detect packet loss or a drop in voice quality. However, packet loss is not sufficiently sensitive to early stages of congestion, since congestion usually begins before packet loss occurs. Waiting to detect packet loss allows the congestion to become much worse before action is taken.
Congestion and Bandwidth Detection
Rather than simply wait for packet losses to mount up, the parent application detects such congestion beginning to occur. In addition to congestion estimates, bandwidth may also be estimated to detect when bandwidth is limited but packet loss is not yet occurring.
Congestion and bandwidth restrictions are detected by one VOIP application and sent to the other VOIP application. Each VOIP application monitors and estimates congestion and bandwidth on its inbound direction for use by the other VOIP application. Congestion and bandwidth estimates can be embedded in the VOIP packets for sending to the other VOIP application.
VOIP applications can respond to the congestion and bandwidth estimates in a variety of ways. The present application describes a closed-loop system wherein one VOIP application estimates congestion and bandwidth while the other VOIP application receives the bandwidth and congestion estimates and adjusts its packet flow to compensate.
What is desired is a VOIP application that can continuously receive estimates of network problems such as congestion, limited bandwidth, and delays. A VOIP system that can continuously respond to such estimates by adjusting audio-compression and bandwidth usage is desirable. A pair of VOIP applications that continuously monitor network conditions and adjust packet flow is desired. A closed-loop feedback VOIP system for network monitoring and audio-packet-flow adjustment is desired.
The present invention relates to an improvement in voice-over-Internet-Protocol (VOIP) systems. The following description is presented to enable one of ordinary skill in the art to make and use the invention as provided in the context of a particular application and its requirements. Various modifications to the preferred embodiment will be apparent to those with skill in the art, and the general principles defined herein may be applied to other embodiments. Therefore, the present invention is not intended to be limited to the particular embodiments shown and described, but is to be accorded the widest scope consistent with the principles and novel features herein, disclosed.
Packets 34 from user A to B travel through path 38, which has a restricted bandwidth. For example, a router may be congested or a dial-up modem may be in path 38. Packets 36 from user B to user A travel through Internet 44 on a different route, path 39, which has a larger bandwidth in this example and at the time shown.
Bandwidth detector 40 is part of VOIP application 30. Incoming packets 36 are analyzed by bandwidth detector 40 to determine the packets' travel time along path 39 and indirectly estimate the bandwidth of path 39. This bandwidth estimate from bandwidth detector 40 is added to outgoing packets 34. Packets 34 contain both voice data from user A, VA, and the bandwidth estimate for packets 36 sent by user B, BW_B.
When packets 34 are received by VOIP application 32, user A's voice data VA is extracted and played back to user B, and the bandwidth estimate BW_B is read, allowing VOIP application 32 to adjust or halt its transmission of outgoing packets 36. For example, when bandwidth is reduced, VOIP application 32 can signal user B of the problem, such as by generating an audible beep or displaying a quality meter on a visual interface to indicate the poor bandwidth.
VOIP application 32 can adjust its audio compression ratio to change its bandwidth consumption in response to changing bandwidth estimates received from VOIP application 30. When bandwidth estimates are reduced, VOIP application 32 can increase audio compression or use more efficient coding methods. Audio frames can be dropped to reduce bandwidth consumption. The quality of audio playback can be reduced to squeeze the audio packets into the more-restricted network pipe. Larger more efficient packets can be used.
Bandwidth detector 42 in VOIP application 32 also measures the arrival rate of incoming packets 34 to estimate the bandwidth of path 38. This bandwidth estimate for user A, BW_A, is added to outgoing packets 36 which contain user B's voice data, VB. Thus packets 36 contain VB and BW_A, while packets 34 contain VA and BW_B.
BW Estimates and Audio-Compression Adjustments Yield Closed-Loop VOIP
VOIP application 30 can also adjust its audio compression ratio to change its bandwidth consumption in response to changing bandwidth estimates received from VOIP application 32. When bandwidth estimates are reduced, VOIP application 30 can increase audio compression or use more efficient coding methods for packets 34. Audio frames can be dropped to reduce bandwidth consumption. The quality of audio playback can be reduced to squeeze the audio packets into the more-restricted network pipe. Larger, more efficient packets can be used.
For example, packets 34 have varying audio-playback durations. Less-efficient packets 34 have durations of 20 or 30 ms. When bandwidth estimates are reduced, larger, more bandwidth-efficient packets 34 having 40 or 80 ms of audio-playback can be used. The overall system of VOIP applications 30, 32 is closed-loop. The bandwidth of the A-to-B network path 38 is continuously monitored by VOIP application 32, which sends estimates BW_A in packets 36 back to VOIP application 30. On receiving these bandwidth estimates, VOIP application 30 adjusts its audio coding and packet size to compensate for increases or decreases in the bandwidth estimate. The bandwidth used by packets 34 is adjusted, which can cause different readings of available bandwidth by VOIP application 32.
For example, when VOIP application 32 detects a lower bandwidth on A-to-B path 38, it reduces BW_A. VOIP application 30 receives packets 36 with a lower BW_A, and increases the packet size and audio compression ratio at the expense of reduced audio quality. This reduces the bandwidth occupied by packets 34. The reduced bandwidth consumption of packets 34 better fits through restricted path 38.
One-Way Latency Measured, Not Round-Trip Time
Bandwidth detector 42 in VOIP application 32 also can measure the travel time or latency of incoming packets 34 to estimate the congestion of path 38. When latency begins to increase, congestion is starting to appear.
The latency or travel time measured by bandwidth detector 40 is not the round-trip travel time. The round-trip travel time includes both paths 38, 39. Instead, only the one-way latency is measured, from VOIP application 32 to VOIP application 30 over path 39. Separate bandwidth and congestion estimates allow for asymmetric latencies, such as when path 38 is restricted while path 39 is not. More precise bandwidth estimates are thus possible.
Congestion estimates can also be responded to in a closed-loop fashion. For example, congestion in restricted path 38 can be detected by VOIP application 32 from the increased latency of incoming packets 34. A congestion estimate is increased and sent in packets 36 back to VOIP application 30. VOIP application 30 may delay sending packets until the congestion ends and the congestion estimates from VOIP application 32 drop. If the congestion is momentary, the delayed packets can be sent late by VOIP application 30, or fresher packets can be sent and the older packets dropped before transmission.
This prevents compounding the congestion problem by sending more packets into a congested router. The user can also be notified that his packets are not getting through, so he can delay speaking.
The incoming packets contain bandwidth and/or congestion estimates from the other VOIP application. Jitter buffer 48 sends these estimates to variable packetizer 50. Variable packetizer 50 forms the outgoing IP packets by adding headers and catalogs of the voice data, to the encoded voice data from codecs 52. The codecs 52 selected can be responsive to the bandwidth estimate received from the packets in jitter buffer 48.
Variable packetizer 50 can adjust bandwidth consumption by varying the packet size and audio compression. Larger, more efficient packets can be used to reduce overall bandwidth consumption. A codec with greater audio compression can be used, and audio frames can be dropped altogether to significantly reduce bandwidth consumption when a severe loss of bandwidth occurs.
Incoming packets with user A's voice data are received and stored by jitter buffer 48. Some delay and variation in packet reception is accommodated by jitter buffer 48, and packets can be re-ordered by sequence number if received out of order. The packets are sent to core manager 56 of VOIP application 32, which extracts the voice data from the packets, examines the voice catalog, and selects the specified codec to decode and decompress the voice data. The final decoded, decompressed voice data is played as audio to user B. Core manager 56 may contain a variety of software modules including a user interface or may call other modules, library, or operating system routines.
Two Different Bandwidth Estimates: Incoming BW_A and Outgoing BW_B
Two different bandwidth estimates exist. BW_A is generated by bandwidth detector 42 (
The second bandwidth estimate, BW_B, is embedded inside the incoming packets 34 that arrive at jitter buffer 48. This bandwidth estimate was made by the other VOIP application. Variable packetizer 50 and/or core manager 56 use BW_B to alter the audio compression ratio, packet size, and audio frame skipping to adjust the bandwidth consumed by outgoing packets 36. The bandwidth consumed is adjusted to more closely match the bandwidth estimate received, BW_B.
Latency Measured by Time-Stamps
Time stamper 46 provides time-stamps or clock values that are an indication of time. Time stamper 46 generates the arrival time for each packet received by jitter buffer 48. Each packet also contains a send time that was included by the other VOIP application. Bandwidth detector 42 compares the arrival time with the send time for each packet to get the packet's one-way travel time or latency. The change in latency over time is used to determine when congestion occurs.
The arrival rate of incoming packets is used to estimate incoming bandwidth. For example, when the arrival times between packets increase, bandwidth is reduced. Bandwidth detector 42 generates current estimates for the incoming bandwidth, BW-EST, and congestion, CONG-EST.
Variable packetizer 50 receives the bandwidth and congestion estimates from bandwidth detector 42 and adds these to outgoing packets. The estimates may be numerical values such as 5-bit or 8-bit binary numbers that represent a magnitude of bandwidth or congestion, or may be more qualitative values such as 2 or 3-bit values that indicate "good", "average", "poor", or "blocked" paths. One-bit values such as a congestion flag may also be used.
When packets fail to arrive at jitter buffer 48, or are substantially late, such as more than 2 seconds, the packet loss counter is incremented. The packet loss counter PKT-LOSS may also be included in outgoing packets.
IP header 60 contains the destination and source IP addresses while TCP/UDP header 62 contains the TCP or UDP port or other TCP information. Checksums and other information may also be included. Application audio or voice data field 68 contains the compressed and encoded voice data and may be sub-divided into several sub-fields.
Send time field 64 contains the send time S(N) or time-stamp value placed into packet 36 when the packet was transmitted. Catalog 66 is a directory of the voice-data contents of voice-data field 68. Voice-data field 68 may be broken into segments called audio frames, Data1, Data2, Data3, which are listed in the index of contents in catalog 66. Catalog 66 can specify a number of attributes for each segment, such as the codec used, the audio duration of the segment, the type (such as voice/silence/silence-identifier) and the sequence number of the segment. Segments can carry their own sequence numbers independent of the packet sequence number, or in a simpler implementation the packet sequence number can be used in conjunction with accumulated catalog frame duration markers to know the exact timestamp of each segment in the packet.
The playing time for the entire packet of voice data, such as 20 milli-seconds, is the duration D(N). This voice duration can be explicitly or implicitly contained in catalog 66. The packet duration may have to be calculated by adding durations of segments (Data1, Data2, Data3 . . . ) of voice data in voice-data field 68, or by considering the kind of codec and compression used and the number of bytes of voice data. The overall size of the packet and its header is variable as the packetizer can choose how many audio frames to include in a packet. The packetizer can mix several different types of compression and select a subset of audio data to not be sent to reduce the amount of data that must be sent for a given period of time represented by the packet.
The bandwidth estimate BW-EST, congestion estimate CONG-EST, and packet-loss counter PKT-LOSS can be added to the end of packet 36. Often unused bits are available at the end of the compressed voice data in voice-data field 68, or additional bits can be added to packet 36 for estimate fields 70, 72, 74, which contain the bandwidth, packet-loss, and congestion values. Not all of fields 70, 72, 74 need to be present in all packets.
Different codecs and coding methods can be used for different segments or frames of the audio data. For example, voice frames contain talking voice data and use higher-quality coding methods. Voice data is in frames 1, 2, 4, 6, 7, and 8. When the user pauses, a period of silence is detected. Silence Identifier SID identifies a period of silence. A lower-quality codec that produces higher compression can be used for these silence periods, such as frames 4 and 5. In some cases the length of silent time can be coded rather than coding the background noise. Compression can thus be very high for silent periods.
The coded, compressed data 302 can be further reduced in size by decimation. Some of the less-important frames can be deleted. The location of the deleted frames are noted in the audio packet catalog and the receiving VOIP application must re-create the deleted frames, perhaps by interpolating from the surrounding audio frames, or by inserting silence or some pre-defined sound.
In this example, frames 4 and 7 are deleted by decimation. Frame 4 is a silence frame but frame 7 is a voice frame.
The coded, compressed, and decimated audio data 304 has a smaller size than original voice data 300 or compressed data 302. The amount of reduction, or the audio-compression ratio, can be adjusted to reduce or expand the bandwidth used by the audio packets to better match the bandwidth estimates. Audio quality versus use of bandwidth can be adjusted over a very wide range using this technique.
Decimated audio data 304 is then divided into IP packets by the variable packetizer. The packet size is larger when more frames and larger frames are included in a packet than when fewer and smaller audio frames are included. For example, packet 306 is larger than packet 308. Packets the size of packet 306 rather than the size of packet 308 can be used to increase efficiency when the bandwidth estimate is reduced. Efficiency is higher with larger packets because the header is shared by a larger number of frames in the larger packets.
Each packet 306, 308 has an IP header attached and a UDP or TCP header following the IP header. Packet 306 has audio frames 1, 2, 3, and 5, while packet 308 has audio frames 6 and 8. The frame order could be non-sequential when frame sequence numbers are included in the packet's catalog.
When the bandwidth estimate is greater than the sending bandwidth, step 106, then there is excess bandwidth available in the path through the Internet. The sending bandwidth can be increased by decreasing coding, compression, decimation, and packet size efficiencies, step 114. This produces a better audio quality and potentially less delay but consumes more bandwidth as more bits of data are transmitted over the Internet path.
When the bandwidth estimate is less than the sending bandwidth, step 106, then there is reduced bandwidth available in the path through the Internet. The network is operating above its limit, and congestion may soon occur. The sending bandwidth can be decreased by increasing coding, compression, decimation, and packet size efficiencies, step 112. This produces a reduced bandwidth consumption at the expense of deteriorated audio quality and increased delay. Fewer bits of data are transmitted over the Internet path due to the higher compression ratio.
When the packet size is already at the maximum size, step 120, more drastic configuration changes need to be taken. While increasing the packet size doesn't reduce the sound quality, coding, compression, and especially decimation changes can reduce audio quality. When the configuration does not use the maximum compression, step 122, a different, more compressed, coding method is configured, step 128. A different codec or a different compression algorithm that more highly compresses the voice data may be used. In general, the higher-compression codings have reduced voice quality, although different voice samples may be more accurately compressed with some methods than for others. VOIP packets generated by the variable packetizer use the new, more-highly compressed encodings when audio data is coded by the codecs. The new configuration with the new codec is used to code, compress, decimate, and packetize the audio data, step 132.
When the compression is already at the maximum size, step 122, even more drastic configuration changes need to be taken. Decimation can significantly reduce audio quality since entire audio frames are dropped. When the configuration does not use the maximum decimation, step 124, more decimation is configured, step 130. A greater percentage of audio frames may need to be dropped, such as one out of every six frames rather than one out of every ten frames.
Dropping more frames degrades the voice quality more, so decimation is not increased until packet size and compression changes are made. VOIP packets generated by the variable packetizer are now more-highly decimated. The new configuration with the new decimation ratio is used to code, compress, decimate, and packetize the audio data, step 132.
When the maximum packet size, step 120, maximum compression rate, step 122, and maximum decimation, step 124, are already configured, additional efficiencies are impossible. The audio data is compressed, decimated, and packetized using the current configuration which is already the most efficient. Of course, very high degrees of decimation could be used, such as dropping every other audio frame, but a decimation limit may be imposed.
Alternatively, when the most efficient configuration is reached and bandwidth estimates are still too low, all packet transmission could be halted and the user notified, or only a few small packets sent out. Other techniques could be used, such as re-initializing the call with the remote VOIP application, or waiting a period of time before sending any packets. If an alternate transport is available (different network, different routing, or conversion to Public-Switched-Telephone-Network phone call) it could be triggered at this point.
When the configuration does not use the least decimation, step 140, less decimation is configured, step 146. A reduced percentage of audio frames may need to be dropped, such as one out of every ten frames rather than one out of every five frames. The minimum decimation is typically zero, which has the best voice quality since no frames are dropped. Reducing or eliminating decimation can significantly increase voice quality. VOIP packets generated by the variable packetizer are now not decimated as much. The new configuration with the new, reduced decimation ratio is used to code, compress, decimate, and packetize the audio data for transmission, step 152.
When the last configuration does not use the minimum compression, step 142, a different, less compressed, coding method is configured, step 148. A different codec or a different compression algorithm that compresses the voice data to a lesser degree may be used. This tends to improve voice quality, although different voice samples may be more accurately compressed with some methods than other voice samples. VOIP packets generated by the variable packetizer use the new, less compressed encodings when audio data is coded by the codecs. The new configuration with the new codec is used to code, compress, decimate, and packetize the audio data, step 152.
When the decimation is at the minimum, such as zero, step 140, and the least-compressed codec is used, step 142, then the packet size being used by the variable packetizer is examined, step 144, to determine if the packet size is at the minimum configurable packet size. If the packet size can be decreased further, the packet size is reduced, step 150. Future VOIP packets generated by the variable packetizer use the new, smaller packet size when audio data is packetized. The new configuration with the smaller packet size is used to code, compress, decimate, and packetize the audio data for transmission, step 152.
When the decimation is at the minimum, such as zero, step 140, and the least-compressed codec is used, step 142, and the minimum packet size is used, step 144, then further increases in audio quality are not possible. The current configuration settings are used, step 152.
Each new incoming packet with a bandwidth estimate can further change the configuration. For example, each incoming packet with a bandwidth estimate that is still below the sending bandwidth can cause the packet size to be increased further until the maximum packet size is reached. The sending bandwidth increases as the configuration is changed, so eventually the sending bandwidth may fall to meet the bandwidth estimates. Then further increases in packet size are not needed unless the bandwidth estimates fall further. Relatively small changes in configuration per packet can add up to larger, rapid changes over many packets received in a short time.
The sending bandwidth can be updated once the configuration of coding, compression, decimation, and packet size is changed. A table can be used to determine the sending bandwidth based on the configuration of coding, compression, decimation, and packet size. The various steps can be combined into a configuration-table lookup that selects the next configuration, which may be a combination of changes in coding compression, decimation, and packet size.
The sending bandwidth can be based on the last bandwidth estimate, or an average of the last several bandwidth estimates. Then changes in the bandwidth estimate cause changes in configurations for coding, compression, decimation, and packet size, such as selection of the next configuration line in the configuration table.
During time period 200, packets arrive along ideal line 250.
However, at time period 202, packets are delayed and arrive with longer inter-packet arrival times. Arrival times T4 and T5 are delayed, causing packets to arrive below ideal line 250, with a lower slope or arrival rate.
The bandwidth estimate is reduced by a portion of the lateness, and falls sharply during time period 202. When packets are very late, the bandwidth estimate can be reduced even before the packet arrives. A timer can wake up periodically to examine the most-recently-arrived packet. The maximum-size packet's duration can be compared against the time that has transpired since the last packet arrival. In an example where the network comes to almost a complete halt for an extended period, late packets can be detected by expiration of a maximum inter-arrival time. This can be factored into the bandwidth and congestion estimates.
Packets begin arriving at the ideal rate during period 204. The packets have the same slope as ideal line 250, but are below line 250 due to the delays from period 202. The bandwidth estimate rises slightly during this period.
The network recovers quickly during period 206 as many packets arrive in a short time. This can occur as a router recovers from a delay and works off its packet backlog. The packets rapid arrival produces a slope higher than that for ideal line 250, and eventually the packets reach line 250. The bandwidth estimate rises quickly during period 206 in proportion to the difference of inter-packet arrival time and the voice duration of the voice data inside the packets.
Finally in period 208 the network is again stable and packets arrive along ideal line 250. The bandwidth estimate is edged up slightly to test the upper limit of bandwidth.
While estimating bandwidth based on packet arrivals is performed by one VOIP application, responding to these bandwidth estimate changes is performed by the other VOIP application for the other user. Thus
The relatively stable but low-bandwidth network conditions during period 204 allow the sending bandwidth to be increased slightly. Relatively similar configurations occur in period 204, causing the sending bandwidth to increase only slightly. This allows time for the network to recover before being flooded with more audio packets.
The rapid network recovery in period 206 causes the bandwidth estimates to rise. Successively less efficient configurations are selected to increase the sending bandwidth during this period. Voice quality improves.
The network is again stable in period 208. Bandwidth estimates increase slightly allowing the sending bandwidth to be increased somewhat.
Latencies are rising slightly over long time periods, as shown by the upward bias to the moving average during periods 210, 214. The congestion estimate remains relatively flat during periods 210, 214.
During period 212, a network problem or constriction occurs, causing the current packet latencies to rise sharply above the moving average. This can occur when a user sends or receives email over a modem line that is being used by the VOIP packets. The congestion estimate quickly rises as the latencies rise.
Rather than fall back as quickly as the latencies as the peak ends, the congestion estimate remains high as the current latencies fall sharply as
Once the current latencies cross the moving average line at the end of period 212 and the beginning of period 214, the congestion estimate starts to fall as the estimate is reduced by a small amount for each of many packets. At this point, the network should be roughly caught up from any packet backlog. As more packets are received, the congestion estimate falls back to the base level in period 214.
Congestion estimates can be responded to by pausing or halting packet transmission. This allows time for the congested router to fully recover before more packets are added to its buffer. Otherwise new packets could fill up the router's buffer, causing an overflow and dropped packets, compounding the congestion problem.
When the congestion estimate rises above a congestion threshold during period 212, the local VOIP application takes drastic action. All VOIP packets are held up and not sent out over the congested network path. Once the congestion estimate falls back below the congestion threshold during period 214, these packets are transmitted.
When the congestion is short, most or all of the packets held by the local VOIP application can be sent upon recovery. However, when the congestion occurs for a longer period of time, packets may be discarded by the local VOIP. Instead, fresh VOIP packets are sent once congestion ends. Very short moments of congestion can result in a tiny delay in sending the next packet almost similar to using a larger packet size in that it creates a small increase in the audio delay to the far-end user.
Other responses could be taken by the local VOIP application. For example, rather than halting all packet transmission, decimation could be significantly increased. Alternately, only a fraction of the packets could be transmitted to test the network path. Even though voice quality would be seriously degraded, detection of the end of congestion is improved. An error message could also be sent in a packet to the remote VOIP application, and the local and/or remote users notified of the problem by an audible tone or a displayed message.
Congestion Detected Before Packet Loss Occurs
Congestion can be detected before packet loss occurs by detecting a rise in latencies that often occurs before packets are dropped. Congestion is quickly detected by the use of the moving average. Congestion estimates rise quickly but fall more slowly, allowing time for congested packets to be cleared out. The congestion estimate is fed back to the sender, allowing the sending application to reduce the bandwidth of packets being sent until the congestion ends. Drastic and quick bandwidth reductions may be useful when congestion occurs.
The congestion estimate can quickly respond to delayed packets. The bandwidth estimate shows more of an overall picture of the total available flow of packets. The congestion estimate can more quickly react to sudden changes while the bandwidth estimate can be a smoother measure of the overall carrying capacity of the network path that is less sensitive to individual packets.
The congestion estimate may be designed to detect short term or sudden increases in the ability of the network to deliver packets, while the bandwidth estimate tracks the slower overall carrying-capacity of the network. Sharp changes in inter-packet arrival time (or lack of packet arrivals) trigger the congestion estimate to rise.
It is common for congestion to subside just as rapidly. Very gradual changes in the overall carrying-capacity of the network may be followed by the bandwidth estimate, which is less sensitive to momentary spikes of congestion.
Several other embodiments are contemplated by the inventor. For example, the packet size may be held constant while only the decimation and compression are adjusted, and combinations of adjustments can be made. More drastic changes in compression and decimation configuration could be taken when large changes in bandwidth estimates are received than when smaller estimate changes occur. In some cases a table of bandwidth conditions could be used to select specific combinations of packet size, compression and decimation, since in some cases the bandwidth gains by going to a higher level of compression may mean that decimation can be eliminated or a larger packet size be used and still arrive at a lower total bandwidth use.
Various combinations of software, hardware, or firmware implementations are possible and various routines can be called and executed sequentially or in parallel. While the VOIP packets have been described as being routed over the public Internet, packets may be routed over other networks or combinations of networks such as Ethernets, Intranets, wide-area networks, wireless networks, satellite links, etc. Unmanaged networks can be used or networks with some management. The audio packets can also include multi-media data such as images, video, or text.
Rather than estimate bandwidth by calculating the latency for each packet, only a subset of the packets could be checked, such as every 5th packet or every 50th packet. Changes in configuration may not occur for each packet but could be delayed. The durations of intervening packets could be summed. The bandwidth and congestion estimates could likewise be embedded in only some of the outgoing packets rather than all outgoing packets. The bandwidth and congestion estimates could also be sent in separate packets without voice data. The voice data is really audio data that is often voice, but could include other audio data such as songs, music, traffic noise, etc.
The bandwidth estimate could also be kept constant when the network is stable, or could be increased by a different amount or by a variable amount. The congestion estimate could be performed before or after the bandwidth estimate, or at the same time. Parallel processing could be used on some systems.
Some systems may use only the bandwidth estimate or only the congestion estimate. Other systems may only send bandwidth, packet-loss or congestion data when it has changed from previous values, or only send it on a subset of packets where there happens to be excess space at the end of the audio payload.
Network recovery typically is very quick, and the congestion estimate can be raised immediately, or as shown in the previous embodiment, the congestion estimate can be left at its present level until such time as the network has cleared any backlog of stale or delayed packets.
The bandwidth and congestion estimate and configuration-response routines could be activated by the jitter buffer when packets are late in arriving but before the packets arrive. Since the sending times of the missing packets are not known, they may be interpolated from other packets, or a fixed number used to calculate the new arrival time, latency, or voice duration. The amount of voice data in packets can vary from packet to packet rather than be the same for all packets as described in the simplified examples. The jitter buffer may perform other functions, such as detecting and processing duplicate and missing packets.
The send and receive times may be relative times or somewhat different times, such as a time-stamp added just before transmission or some delay after the packet arrives, or could be added at other times. The time-stamp may be a full time in a 24-hour format, or may be a subset of the full time, such as the current minute and seconds values, or may be a relative time value such as from a counter that changes with time. A processor or other hardware timer may used, or perhaps accessed using software routines. The sending and receiving VOIP application timer can be synchronized by a third-party timer, or by using round-trip packet transit times to adjust or correct timer differences.
Synchronization between the remote and local VOIP applications can occur at the start of communication. A series of packets can be exchanged simultaneously in both directions between the local and remote applications. Each synchronizing packet can contain a sent time-stamp to which is then appended a received time-stamp. The packet may be returned to the opposite side where a third time-stamp of the return arrival can be made. From these packets, the round trip delay is easily determined, and by comparing the sent, received, and returned time-stamps on packets which went in opposite directions an estimate of the latency in each direction can be made. Using this information, the clocks at both ends can either be synchronized, or a known offset can be recorded so that remote-application's time-stamps can be adjusted into local time of the local VOIP application. In an alternate embodiment, absolute time-stamps can be abandoned and the methods can be implemented purely on relative time-stamps. For example, a send time of 12653 milli-sec from the start of a call and can be compared to a previous send time-stamp of 12571 milli-sec to get an elapsed time measurement.
Outlying data points such as from very slow packets could be removed to allow for an occasional transient or random dropped or delayed packet. Additional filtering could be performed. Many kinds of moving averages can be used, such as a simple arithmetic moving average, weighted moving averages that increase weighting of more recent data points, exponential moving averages, etc.
Data values can be considered "equal" if within a certain range of each other, such as within 1% or 5% or 0.1%. Also, rounding of values can be performed before comparison, effectively providing a range of "equal" values. Congestion and bandwidth estimates can use only a few bits to indicate qualitative measurements such as "normal", "minor restriction", "major restriction", "blocked", or may use more bits to represent a quantitative estimate such as a percentage or data rate. One or both users could be notified of problems by an audible tone, vibration, or a display message, or the estimates could be logged to a file for debugging. The application may visually display a network-quality meter to the user.
VOIP calls may be between two users on personal computers, or may consist of one user on a personal computer talking to a computer server or gateway, which converts the call from VOIP to telephone or PBX or private IP phone system formats. The call could also be between two telephone or private IP-phone users with a VOIP segment somewhere in the middle carrying the call from one location to another over the Internet or similar unmanaged network but terminating the call at each end on a telephone or PBX or IP phone. Calls could also involve a conversation between one user on a PC or telephone or IP phone, and at the other end an automated voice response system such as a banking application, voicemail, auto attendant, talking yellow pages or other automated voice service. More than two parties may exist in multi-way calling.
Larger packets are more efficient due to a greatly reduced overhead of the packet header information compared to the voice data payload. A smaller number of larger packets can carry the same voice data payload more efficiently using fewer total bits of bandwidth than a larger number of small packets. In general, small packets are preferred since they reduce the latency of sending the voice from one application to another. But if the network is not keeping up with the quantity of packets being sent, delays will be rising already, and overall better performance can be reached by using larger more efficient packets that prevent the network from being overwhelmed.
Larger packets increase the delay in shipping audio data from one application to the other, so in the face of excess bandwidth it is desirable to use very small packets to minimize delays and make full duplex conversation and the ability for one user to interrupt the other as comfortable and immediate as possible. The practical minimum size of a packet is one audio frame which for most compression algorithms is in the region of 5 mSec to 30 mSec. There is no hard upper bound to how large a packet can be used, but as packet size is increased there can be diminishing returns on the savings in header data. Packet sizes of several seconds in length could theoretically be used, but could ruin full-duplex conversation just as buffering many seconds of data at the receiving jitter buffer would do. Depending on the nature of the application and how much bandwidth the network appears to have against the compression level which has been selected and the observed level of packet loss, an upper limit in the region of 40 mSec to 250 mSec for packet size is typical. The network IP protocol may also restrict the useful packet size to 1024 bytes or a multiple thereof depending on the details of the network and the routers supporting packet flow. Packet sizes of tens or hundreds of bytes are much more commonly used.
Some compression algorithms include support within their standard for marking frames of audio as voice or silence, and can forward representations of the background noise. This is called a Silence Identifier or sid frame. It can be used at the far side for efficient re-synthesis of quiet sections of near silent and fully silent audio. Variable thresholds and use of these compression algorithm silence/sid features can be used as can moving from one compression algorithm to another.
Decimation can pick from among the available audio frames ready to be shipped and select to decimate the ones least likely to have acoustic impact.
Sending bandwidth can be calculated as an absolute number based on the total number of bytes in each completed packet divided by the time interval which that packet represents, or it can be monitored on a relative basis compared to the bandwidth estimate being sent from the far side (one notch up, one notch down), or it can consist of a series of pre-determined combinations of packet size, compression and decimation in a table with known associated bandwidth averages for each possible combination calculated ahead of time by the designer.
The abstract of the disclosure is provided to comply with the rules requiring an abstract, which will allow a searcher to quickly ascertain the subject matter of the technical disclosure of any patent issued from this disclosure. It is submitted with the understanding that it will not be used to interpret or limit the scope or meaning of the claims. 37 C.F.R. � 1.72(b). Any advantages and benefits described may not apply to all embodiments of the invention. When the word "means" is recited in a claim element, Applicant intends for the claim element to fall under 35 USC � 112, paragraph 6. Often a label of one or more words precedes the word "means". The word or words preceding the word "means" is a label intended to ease referencing of claims elements and is not intended to convey a structural limitation. Such means-plus-function claims are intended to cover not only the structures described herein for performing the function and their structural equivalents, but also equivalent structures. For example, although a nail and a screw have different structures, they are equivalent structures since they both perform the function of fastening. Claims that do not use the word means are not intended to fall under 35 USC � 112, paragraph 6. Signals are typically electronic signals, but may be optical signals such as can be carried over a fiber optic line.
The foregoing description of the embodiments of the invention has been presented for the purposes of illustration and description. It is not intended to be exhaustive or to limit the invention to the precise form disclosed. Many modifications and variations are possible in light of the above teaching. It is intended that the scope of the invention be limited not by this detailed description, but rather by the claims appended hereto.
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|U.S. Classification
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|International Classification
|Cooperative Classification
|European Classification
|Mar 6, 2003
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text:
Boca Juniors 2012-13 away player version authentic shirt in white/black stripes.White black colors were first shirt used by Boca in 1905.Nike Dry Fit fabric made by recycled polyester materials.Mesh throughout for breathability.Features screened on team badge,Nike logo and screened on BBVA sponsor in front and on the sleeves.`CABJ` is screened on the back of the neck and comes with Authentic tag stitched on at the lower left of the shirt with the shirts own serial number.Nike Dry Fit technology for comfort.Inside the neck is scripted`El rey de mundial clubes`. Lightweight ,100% recycled polyester.Perforations at sides for excellent ventilation.
|Name and Number


text:
Research at the Beckman Institute
The Beckman Institute's mission is to foster interdisciplinary research of the highest quality, transcending many of the limitations that are inherent to traditional university organizations and structures. The Institute was founded on the premise that reducing barriers between traditional scientific and technological disciplines can yield research advances that more conventional approaches cannot.
Overview of Beckman Institute Research
More than 1,500 researchers from more than 40 different University of Illinois departments as diverse as psychology, computer science, electrical and computer engineering, and biochemistry work within and across four research themes at the Beckman Institute.
- Beckman Institute research is focused around four research themes:
- Biological Intelligence
- Human-Computer Intelligent Interaction
- Integrative Imaging
- Molecular and Electronic Nanostructures
- The Beckman Institute is also home to strategic initiatives that seek to unify campus activities in their respective areas:
- Cognition, Lifespan Engagement, Aging, and Resilience
- Illinois Language and Literacy Initiative
Research Organized by Themes
Biological Intelligence
The BioIntel research theme seeks to understand the links between the brain and intelligent behavior.Learn more about BioIntel →
Human-Computer Intelligent Interaction
The HCII research theme seeks to enhance human-machine interface design through the optimization of state-of-the-art technology development and engineering of multimodal interface design concepts.Learn more about HCII →
Integrative Imaging
The IntIm research theme is dedicated to bringing together ideas, modalities, and people in imaging to foster the interdisciplinary discovery of fundamental principles in imaging science, new enabling technologies for the next generation of imaging instruments, and novel techniques for basic and translational research.Learn more about IntIm →
Molecular & Electronic Nanostructures
The general goal of the Molecular & Electronic Nanostructures (M&ENS) research theme is to develop a fundamental understanding of chemical and physical processes involving structures on the nanometer scale.Learn more about M&ENS →
Research Facilities
Biomedical Imaging Center
BIC is home to a wide variety of research programs, providing facilities, equipment, and training for research on nuclear magnetic resonance imaging and spectroscopy. A recent initiative of the Center is the integration of magnetic resonance methods with simultaneous optical imaging. It became a unit of the Beckman Institute in the summer of 2002, fostering access to a wide variety of resources.Learn more about BIC →
Illinois Simulator Laboratory
ISL is a Beckman Institute facility for advancing scientific understanding of human-computer interactions. As an Institute-wide facility, the ISL uses its expertise in the integration of advanced visualization, sonification, and interface technologies to enable researchers to conduct experiments in human multi-modal perception and cognition, cognitive and motor development, and multi-dimensional dataset visualization.Learn more about ISL →
Imaging Technology Group
ITG provides state-of-the-art facilities in the Microscopy Suite and the Visualization Laboratory, serving over 400 researchers from the Beckman Institute and nearly every department on campus. A secondary focus of the group is to develop advanced imaging technologies, with an emphasis on projects in remote instrument control and scientific visualization.Learn more about ITG →


text:
Definition of zigzag in English:
1A line or course having abrupt alternate right and left turns: she traced a zigzag on the metal with her finger
More example sentences
- The police are having to take positive action in a bid to stop parents dropping off or picking up their children on the zigzag lines outside three more schools.
- Rectangular cells, gently arching lines and compressed zigzags proliferate across the supports.
- she shows weals on her thigh, thin green lines in a short zigzag.
1.1A turn on a zigzag course: the road descends in a series of sharp zigzags
Having the form of a zigzag; veering alternately to right and left: when chased by a predator, some animals take a zigzag course
More example sentences
- He initiated this possibility by manipulating versions of the liar's paradox with zigzag graphs of truth and falsehood states.
- Gung-ho climbers can tackle the zigzag trail up the steep incline.
- Have a look inside for the jazzy zigzag Norman chancel arch.
meandering, zigzagging, snaking, snaky, winding, wiggly, squiggly, crooked, tacking, twisting, twisty, full of twists and turns, curving, curvy, wavy, deviating, undulating, sinuous, serpentine, tortuous, irregular
rare anfractuous, flexuous, meandrous, serpentiform
So as to move right and left alternately: she drives zigzag across the city
verb (zigzags, zigzagging, zigzagged)[no object]
Have or move along in a zigzag course: the path zigzagged between dry rises in the land
More example sentences
- I once rented a scooter and zigzagged along most of San Miguel's streets to see the neighborhoods.
- For two weeks he sat on a bus that zigzagged along the eastern seaboard picking up other criminals who were being reassigned from one jail to another.
- They zigzagged along until Estrella found herself in the familiar surroundings of the underground meeting hall.
zigzaggedly adverb
- Example sentences
- It was like long snakes zigzaggedly going through a forest full of fireflies showing them direction.
- His mother just did farm chores for her in-laws and babbled on zigzaggedly since his birth.
- He comes quite near and I swim up towards him, but my buddy makes biting motions, so I back off and the leopard shark swims zigzaggedly on his way.
Early 18th century: from French, from German Zickzack, symbolic of alternation of direction, first applied to fortifications.
For editors and proofreaders
Line breaks: zig¦zag
Definition of zigzag in:
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= trending


text:
Take a look in Check Point's 2013 Security Report
Securing company data has always been a concern for the IT department. However, with Google stating that every two days we generate as much information as was created between the dawn of civilisation and 2003 – the IT department is under more pressure than ever to secure this increasing amount of data.
This is not the only headache for the CIO; the IT department is facing an increased threat from securing increased data, more advanced hacking techniques, across more devices. Therefore, some of the results from Check Point's survey of UK IT and security professionals are of little surprise. Not least that, external threats have increased significantly in the past 12 months; with more than half of respondents reporting an increased risk of internal data breaches.
What the results do highlight is that organisations, despite all their best efforts so far, are struggling to keep up with the ever-increasing consumerisation of IT and the sophistication of hacker's techniques.
If you would like to gain greater insight into the network security events that occurred in organisations worldwide, an explanation as to how some of the attacks were conducted and receive recommendations on how to protect against security threats, download Check Point's 2013 Security Report.


text:
Issue No.05 - May (1981 vol.3)
pp: 504-519
Linda G. Shapiro , SENIOR MEMBER, IEEE, Department of Computer Science, Virginia Polytechnic Institute and State University, Blacksburg, VA 24061.
Robert M. Haralick , SENIOR MEMBER, IEEE, Departments of Electrical Engineering and Computer Science, Virginia Polytechnic Institute and State University, Blacksburg, VA 24061.
In this paper we formally define the structural description of an object and the concepts of exact and inexact matching of two structural descriptions. We discuss the problems associated with a brute-force backtracking tree search for inexact matching and develop several different algorithms to make the tree search more efficient. We develop the formula for the expected number of nodes in the tree for backtracking alone and with a forward checking algorithm. Finally, we present experimental results showing that forward checking is the most efficient of the algorithms tested.
Linda G. Shapiro, Robert M. Haralick, "Structural Descriptions and Inexact Matching", IEEE Transactions on Pattern Analysis & Machine Intelligence, vol.3, no. 5, pp. 504-519, May 1981, doi:10.1109/TPAMI.1981.4767144