VoIP Soft

SIP Settings in QIP Infium Messager

2010-01-29T14:57:00.000-08:00

SIP and XIMSS integrated in QIP
Friends! In order to many not mistaken, trying to catch Infium to Asterisk and other SIP-providers, clarify the following - the new QIP Infium does not support SIP as a protocol for communication and exchange of information (including voice). Точнее поддерживает, но только для связи с VoIP провайдерами, использующими сервер Commungate Pro, он же CGP - подробнее на www.Stalker.com . More specifically supported, but only in connection with the VoIP providers that use the server Commungate Pro, also known as CGP - More on www.Stalker.com.

XIMSS - a proprietary protocol CGP, or rather the management interface server from the client via XML. In our case Infium configured for server management of domestic VoIP-provider http://sipnet.ru.

Work directly with an asterisk, and other SIP-servers Infium CAN NOT!

Disgusting is that the standard SIP itself is defective. It simply is not designed for use in the real world. In the laboratory all the SIP UA are equally good.In complex conditions of modern local area networks (NAT, VPN, firewall) it works too, but not in 100% of cases. Even a small percentage of the impervious failures is large enough to cast a shadow on the vendor.

Therefore, companies are creating alternative interaction protocols (Skype, MSN, Yahoo, etc.) or added to the SIP incompatible extensions (Microsoft RTCDLL, Asterisk, eyeBeam, ...).

I have a question for you guys, I do not really understand these networks and technology transfer data, but I would understand if based on SIP (I still was convinced that this is only kleentsky interface) Infiun can connect to other VoIP networks such as VoIP Buster or something like that, which differs from Skype is not CODED. And if not, whether the planned development of QIP, in any of its interpretation, in the direction of commonly used networks, VoIP, which is not limited to companies, or working only in Russia and former Soviet countries?
I do not want to offend anybody, just so happens that so far in Russia organized a number of VoIP companies, each of which has its zamorochki and features and they are not compatible with each other: (But almost all of them have the same communication protocols as well as international (European American) companies VoIP, that's only compatible with them for some reason no.

Portable Asterisk Download Link

2010-01-28T14:52:00.000-08:00

Portable Asterisk Download Link the best on VoIP

Portable Asterisk for other devices, USB Drive and HDD, is continue..

VoIP on iPAD is REALLY! (NEW iPhone SDK now available)

2010-01-28T14:21:00.000-08:00

MacRumors is reporting that the new iPhone 3.2 SDK, now available via developer.apple.com, but (currently?) only supporting the just-announced iPad, brings a couple interesting new features to the ~~table~~ tablet.

External displays, which had potential but no official APIs in previous iPhone SDKs, looks like they’ve gone legit. Video out to come? No word on an HDMI cable still…
Shared file directory, which mounts on both Mac and PC. It isn’t the Mobile Finder we’ve wanted, but it sounds like it might make getting docs on and off a whole lot easier.
Universal apps, which means developers can make a single build that will support both the smaller iPhone and iPod touch, and the bigger iPad.

In addition, MacRumors says iCall Free VoIP [Free - iTunes link] has told them:

With the latest revisions Apple has made to the iPhone developer agreement and Software Development Kit, iCall for the iPhone and iPod Touch now enabled unrestricted free local and long distance calling over 3G data networks.

Finally (for now), 9to5mac shows that while the demo iPad only had 4 apps in its dock, like the iPhone, the SDK simulator has room for 6. Makes sense given the extra width. (Pictures above)

LiPhone or iPhonux - iPhone Linux support really!

2008-12-01T07:04:00.000-08:00

Linux 2.6 kernel has been ported to Apple's iPhone platform, with support for the first and second generation iPhones as well as the first generation iPod touch. This is a rough first draft of the port, and many drivers are still missing, but it's enough that a real alternative operating system is running on the iPhone.

LiPhone 3G

What we have:

- Framebuffer driver
- Serial driver
- Serial over USB driver
- Interrupts, MMU, clock, etc.

What we have in openiboot (but hasn't been ported yet):

- Read-only support for the NAND

What we don't have (yet!):

- Write support for the NAND
- Wireless networking
- Touchscreen
- Sound
- Accelerometer
- Baseband support

The current userland we're using, in the interest of expedience, is a Busybox installation created with buildroot, but glibc works fine as well, and we're going to build a more permanent userland solution.

A demonstration video can be seen here: http://www.vimeo.com/2373142

Instructions here: http://www.iphone-dev.org/planetbeing/LINUX-README.txt

Download here: http://91.186.26.18/iphone/files/iphonelinux-demo.tar.gz (look for mirrors in the comments)

EDIT: The instructions are missing the step that you have to select openiboot console from the menu before performing the "sudo ./oibc" step. Just be aware you have to do that if it seems like you're not getting a response from the oibc client.

Project lead: planetbeing

Contributors: CPICH, cmw, poorlad, ius, saurik

If you're experienced with hacking/porting Linux and especially if you're experienced with porting Android, I'd definitely like to hear from you. Come chill in the #iphonelinux channel on irc.osx86.hu. Thanks. :)

EDIT: I was asked a couple times by people who wanted to donate (financially) to the project. I made a post discussing this possibility, if you are interested.

No Web-based Skype for iPhone on SHAPE Services

2007-08-26T15:33:00.000-07:00

Since Skype doesn't offer web access or clients for all platforms, there's a boutique industry filling gaps in Skype's product line. The IM+ line added an iPhone browser client with callback service.

SHAPE Services hosts a huge bank of Skype clients. Their IM+ for Skype web app, Skype for iPhone, talks to those Skype clients via the Skype API.

Conclusions:

Free for now, except for SkypeOut fees for inbound and outbound voice calls.

Underfeatured, lacking Skype 1.0 tools like group chats, file transfers, and end-to-end encryption.

Skype should have offered web chat/calling features and a set of web services available a year ago. So it is very nice of Shape to fill this gap while Skype catches up (and Skype is on this trajectory).

A walk through Skype for iPhone.

See your contacts.

Touch one, Jim Courtney in this case.

Click on the blue Chat button and chat away.

Or press the green Call button to call Jim. You can see how many Skype credits are in your account. IM+ makes a SkypeOut call to the phone number you put here, so you can take this call on your mobile, landline or SkypeIn number.

If you click the "More" button you see:

Conference Call (coming soon)
Add Contact (coming soon)
About
Help
Hide Offline
Sign Out

Mobile Skype and the Nokia N95

2007-08-26T15:23:00.000-07:00

Browsing one of my favorite mobile computing blogs, I chanced upon a software that will allow one to use the N95 when calling by Skype. The software is called iSkoot and is now available for free download.

I went to the website to find the version for the N95 but it seems not to be supported at the moment. I downloaded instead the version for the Nokia N80 and decided to try it out. Installing the traditional way using Nokia PC Suite via USB, I was up and running in no time.
Here is what I did after installation:
I connected the N95 via my home wi-fi.
I then initialized iSkoot which then asked for my skype login data.
In about 20 seconds, I have showing in the N95 window all my Skype contacts.
I selected one of my US based contacts then selected CALL from the context menu.
You will then hear the iSkoot server informing that the call is being processed.
The other party picks-up and swears that my voice is very clear.
I have tried many times before using a cellphone to call by Skype but have always given up for various issues. iSkoot is the first software that allowed me to do the same totally hassle free.
Although iSkoot does not list the Philippines as one of the supported countries, I can assure you that I was able to connect to the US from the confines of the However, when Skyping a local based contact, I always get a connection failed error. It seems that not being supported means that you can only call from here to another country where iSkoot is supported. Oh well, let's give iSkoot sometime to resolve this.
iSkoot which is still in beta is available for free. I won't hold my breath though that it will stay that way for long.
If you find this article interesting, you must also be interested in how to connect a bluetooth keyboard to your N95. Also, for the more daring, you can even turn your N95 into a powerful carputer.

Microsoft Office RoundTable Review

2007-06-25T06:12:00.000-07:00

Microsoft Office RoundTable is a very cool videoconferencing system featuring 360° panoramic views powered by its 5 built-in cameras. Microsoft sent me a RoundTable system for review. I figured I may as well install it in one of TMC's two conference rooms to have some "real world" testing scenarios. After plugging in the various cables, including a USB cable to the host PC, I then installed the Microsoft Office Live Meeting 2007 (LM 2007) beta software client, which was designed to handle the RoundTable's 360° panoramic cameras. I should point out that Microsoft is offering both a hosted model for Microsoft Office Live Meeting 2007 as well as a CPE (customer premise equipment) solution, namely Office Communications Server 2007 (OCS 2007). The RoundTable can work with either. However, I tested it with a Microsoft Office Live Meeting online beta account.

The RoundTable actually installs two USB camera devices. The first is the Active Speaker which uses the RoundTable's 6 microphones to locate where in the room the active speaker is and then focus one of the 5 cameras onto that person. I suspect it may actually use 2 cameras to focus on a person, since it always appears that the speaker is "centered" which probably would require at least 2 camera images and then the images are "spliced" together, processed, and then transmitted over the USB cable. The second USB camera device is the panoramic camera which combines the 5 camera images into a single panoramic image.

Next, I initiated a Live Meeting session in the conference room and went back to my desk to join the meeting. Unbeknownst to me, by the time I got to my desk, some TMC sales team members walked into the conference room to start a meeting. When I launched my LM 2007 client I saw Anthony Graffeo, one of TMC's sales staff staring right into one of the 5 triangular mirrors located just above the cameras. I headed over the conference room to tell them I could see and hear everything that was going on. After giving a brief demo, Anthony said, "that frickin thing is awesome!". I heard similar comments from other TMCers over the next few days - including "That thing is cool" and "Wow! What is that?"

During my tests with Office Live Meeting 2007, I was able to see the entire conference room in the panoramic view. The other video view is the "active speaker" and it would change to whoever was speaking. It was actually quite accurate and surprisingly fast. When I tested it with Rich Tehrani, Joe Fabiano, Mike Genaro, and Dave Rodriguez, we all took turns speaking and the camera view would almost instantly switch.

Overall, the video quality was excellent, Very smooth. Later on I did some more tests with just me in the conference room. I tried to trick the cameras by snapping my fingers above the microphone and the video wouldn't switch. I figured it must be using a combination of audio and video cues to determine the active speaker. A quick glance in the manual reveals, "To avoid switching to a non-human sound source, the video functions to assist in determining whether the sound detected is coming from an attendee or some other source." Answers that question!

Below is a Flash demo demonstrating the active speaker functionality. When I ran this test, I did it by myself and just walked around the conference room while continuing to speak. I will make a 2nd video with several fellow TMCers if I can get some volunteers. This would better demonstrate the active speaker feature. Also, the video I did record appears much jerkier than it actually was live. Part of that is because it was my first time using Camtasia Studio to capture the screen & audio, so I only had it set to 15fps (not 30fps). Secondly, the PC I was recording on isn't the greatest, so I'm sure Camtasia skipped a few video frames since it the processor was taxed. Nevertheless, this video gives you a good feel for the active speaker functionality. If you want a larger-resolution version, click here.

The base has an LCD touch-screen that serves as a dialpad and to configure various options. An external dialpad is available as well, but I didn't get one with my unit, so not sure if that's an extra option. The LCD touch-screen dialpad seems good enough to me. Physical buttons also exist to increase/decrease speaker volume, a mute audio / halt video button, Flash button, on/off hook, and an Information button. There is also a privacy cap that goes on top of the unit to block the cameras and prevent anyone from viewing the conference room.

Although the Microsoft RoundTable was designed for Office Live Meeting 2007 & OCS 2007, since it is simply 2 separate USB camera devices in the Device Manager, I figured it would most likely work with other videoconferencing, audioconferencing, or VoIP software. So I fired up Skype on the conference room PC and went into the video wizard. I tried the panoramic device first, but all that was displayed was a black screen. Then I changed it to the active speaker USB camera device and it worked. I made a test Skype call to the Microsoft Roundtable, again with some fellow TMC'ers and the active speaker functionality worked just as well in Skype. Obviously, it's the RoundTable hardware performing the video switching and not LM 2007. Also, I was able to use Skype's "full screen" video mode and it had extremely high-resolution video with very fast frame rates. Here's a screenshot of Skype in action viewing the RoundTable video:

Microsoft designed RoundTable to equalize the varying light levels when one part of the conference room is darker than another. In fact, TMC's conference room has sunlight coming in the window and I noticed the RoundTable was able to do a pretty good job equalizing the light across the entire room.

The Microsoft RoundTable features both an RJ11 (phone) and an RJ45 (network) jack. The RJ11 wire lets you connect to a PSTN line and use the RoundTable as a high-end conference room speakerphone. The RoundTable lets you place outgoing calls, answer an incoming call, conduct a video conference with audio from the microphones or a video-only session with no audio from the microphones. In the video-only mode you can use the PSTN line or even your cell phone for the audio portion.

I like the aesthetic design of the Microsoft RoundTable. It is one cool looking device. Reminds me of the alien spaceships in H.G. Wells' War of the Worlds movie (the original 1953 movie, not the Tom Cruise remake). Compare for yourself! Both have "eyes" and both have a long neck connected to the main base.

Interestingly, the RoundTable doesn't aim its 5 cameras directly at you. As previously mentioned, it uses triangular mirrors. I'm not sure if this was done because people are more relaxed and natural if they don't think a camera is staring right back at them or if it was simply for aesthetic design that Microsoft chose to use mirrors. I did some research on the history of the RoundTable and discovered it was previously called the Ringcam. During Microsoft's research, they had some interesting "beta" versions before the final RoundTable, including these two below. Note how both models point cameras directly outward. I definitely prefer the less obtrusive "mirror" version in the production model. I tried to take apart the RoundTable to see the inner guts of the camera head, but the damn thing is glued on or something since I couldn't find any screws.

Even more interesting in a Microsoft Research PDF article it talks about head-size spatial equalization. That is, making sure everyone's head is equally big in the videoconference even if a person is further from the camera. The article has several diagrams and complex trigonometry formulas (sin & cos functions) that bring back high school. The article states the following:

In the past a few years, there has been a lot of interest in the use of omni-directional cameras for video conferencing and meeting recording. While a panoramic view is capable of capturing every participant’s face, one drawback is that the image sizes of the people around the meeting table are not uniform in size due to the varying distances to the camera. Figure 1 shows a 360 degree panorama image of a meeting room. The table size is 10x5 feet. The person in the middle of the image appears very small compared to the other two people because he is further away from the camera.

This has two consequences. First, it is difficult for the remote participants to see some faces, thus negatively affecting the video conferencing experience. Second, it is a waste of the screen space and network bandwidth because a lot of the pixels are used on the background instead of on the meeting participants. As image sensor technology rapidly advances, it is possible to design inexpensive high-resolution (more than 2000 horizontal pixels) omni-directional video cameras [1]. But due to network bandwidth and user’s screen space, only a smaller-sized image can be sent to the clients. Therefore how to effectively use the pixels has become a critical problem in improving the video conferencing experience.

Spatially-varying-uniform (SVU) scaling functions have been proposed [2] to address this problem. A SVU scaling function warps a panorama image to equalize people’s head sizes without creating discontinuities. Fig. 2 shows the result after head-size equalization.

Note how the guy in the white shirt is larger after head-size equalization. Unfortunately, it doesn't appear that the RoundTable I'm testing incorporates the head-size equalization, since none of the heads in the conference room appeared to be equalized. Guess my head (& ego) will have to continue to be the biggest. I read up on this some more and according to this Microsoft article, RoundTable "uses visual cues to pinpoint, enlarge and emphasize the face of the speaker". So it sounds like it does have this feature. I'll have to test this further.

Conclusion
All in all, I was pretty impressed with the Microsoft RoundTable. With its panoramic view and active speaker video switching functionality, you almost feel as though you are there in the conference room. I do wish the panoramic USB camera device worked in other applications besides LM 2007 and OCS 2007, but at least the active speaker video works. Although not released for general availability, the Microsoft RoundTable is expected to retail for around $3,000 putting this in the category of high-end videoconferencing systems. However, with fuel costs and other travel expenses, a high-quality, high-end videoconferencing system can pay for itself very quickly.

Mobile operators lock VoIP

2007-06-25T06:09:00.000-07:00

Edited - Thanks Andy for the heads-up. Much of this content came from the ZDNet article. Cheers David for the scoop.

The mobile cartels are at it again. So soon after the telecoms regulator Ofcom forced a reduction in their termination rates, they are again running scared, this time over phones that enable VoIP.

Orange and Vodafone today decided to remove VoIP functionality from Nokia’s N95 handset.

The feature-rich Nokia N95 supports SIP, which enables integrated IP telephony capabilities, a service that business users are increasingly demanding. Orange and Vodafone, however, have disabled that feature on their N95 handsets.

A spokesperson for Orange said it was “not Orange’s policy to remove VoIP functionality from devices. This is a handset-specific issue and, in this particular instance, Orange was asked by Nokia whether they wanted the VoIP functionality switched on or off, and Orange selected off”.

Vodafone takes a different view. “Vodafone currently does not offer a VoIP proposition that would allow its customers to have universal access to telephony services. Vodafone believes that VoIP over mobile is not yet a mature service proposition, doesn’t have guaranteed quality of service, and does not provide the customer experience demanded of any service we launch,” a spokesperson for the mobile operator said on Thursday. “There is a misleading perception that VoIP services are ‘free’. This is, however, not the case when it comes to using VoIP over mobile, where customers… may incur further charges.”

It looks to me like the mobile cartels are seeing their house falling down.

The real truth is far more simple. VoIP will enable people to call without the need to go through mobile operators. Not surprising then that they have turned off this feature.

How does this affect Rebtel?

Don’t worry – our service allows you to use your free minutes to call anyone internationally for free.

QIP Infium: Now SIP Inside!

2007-06-25T05:41:00.000-07:00

The Russian subscribers have got a real alternative to Skype, one of the most famous internet communication programs. CNews found out, QIP developers finished QIP Infium program creation, integrated with SIPNET Russian telephony operator products and USB equipment for Skypemate voice communication. Besides the multi services messenger (like QIP Infium) the Russian users have got a service, enabling them to communicate with the whole world practically free of charge.

CNews learnt out, QIP Infium messenger beta-version development was finished in June 22, 2007. QIP Infium multi-services program supports ICQ, Jabber and XIMSS (SIP). The presence of the XIMSS module in the new product means that all the QIP Infium users will get the opportunity to communicate through the internet practically free of charge using the mentioned program. QIP Infium provides many new opportunities: the product supports not only numerous services but makes it possible to work with several profiles, plug-ins, unicode, etc.

QIP Infium is integrated with Skypemate hardware. It means that to use the internet telephony, it is necessary to acquire one of the many SkypeMate devices (for example, any USB handset). Then one may simply plug the bought device to the computer or notebook USB-port, and it will automatically attach to QIP Infium. The SIPNET card is already included into all the boxes with the SkypeMate products. The login and password recorded on the card are to be indicated in the XIMSS messenger tuning module.

Thus a user gets the opportunity to phone to any part of the world for token payment. Current Moscow and St. Petersburg connection is carried out absolutely free of charge. Anyone can phone to the user’s computer using a stationary telephone or a handset. It is enough to simply dial a special gate number (each town has its own number), and to dial SIP ID (recorded on the SIPNET card).

According to the developers such a solution has many advantages, form the possibility to significantly reduce telephone conversation expenses to the fact that the user receives an additional full-scale telephone line above all.

“QIP Infium is quite a unique product, - Sergey Kravtsov, Head of the Department on cooperation with SIP equipment suppliers told to the CNews correspondent. - At present it has attracted millions of communication service users both in Russia and abroad. It is just what the market needs, while SIPNET services integration into QIP Infium enables an ordinary internet user to estimate the VoIP advantages in full. We see that the product is rather easy in exploitation, and that is very important. At present it is enough to enter Euroset, Bely Veter, Polyaris or PCHome.ru, buy any SkypeMate USB handset, fill up your personal account, launch your QIP Infium and simply start communicating”.

“The appearance of such a multi-services project like QIP Infium indicates that IP-telephony market has been developing in the right direction, when QIP Infium, SIPNET operator and Skypemate equipment suppliers combine their efforts to achieve one single goal. And our common goal is to make internet communication easier and more popular”, - Mikhail Usov, UMD Project, Means of Radio Communication Joint-Stock company Director General, the exclusive supplier and Skypemate trade mark owner in Russia says.

“There are many difficulties for an ordinary Russian resident to work with new technologies and equipment for such technologies (SIPNET, Skype, etc.), when using hardware to simply start working (registration, filling up the accounts)”, - Vladimir Tyurin, Euroset operators supervision Project Manager says. According to Mr. Tyurin, SkypeMate USB handsets are successfully sold and SIPNET payments are carried out in the Euroset retail outlets. QIP Infium allows people to start operating the sold equipment in short time and that is a great plus. The "switch on and start working” principle might interest many users and that will all ow to make even more people use the new equipment, distributed by our retail outlets”, - Vadim Tyurin thinks.

QIP Infium designers do not intend to stop and plan to add a great number of new possibilities to their product as well as to finish off the already existing products in the near future.

“Infim has combined those advantages that were collected during QIP 2005 creation and testing, - Ilkham Zyulkorneev, the projects author says. – Redesigning a wheel once again one already knows where the hidden problems might appear and tries no to repeat the former mistakes. Most of all we wanted not only to realize the possibility of supporting several profiles and different ways of transferring instant messages in Infium, but also to eliminate all the possible inconveniences and to create such an interface that is practically the same as the usual qip2005. Using Infium, you might now switch to different networks to send instant messages, beginning with Jabber and finishing with Mail.ru agent, everything depends on your desire and choice”.

Developers of Mail.ru Agent - QIP Infium messenger competitor – positively estimate the new product prospects on the market: “I can say that I appreciate QIP designers, who managed to create such an easy product that it has won over the greatest market share of the ICQ original product. I think they will be able to win over a part of Skype users in Russia, but not to replace the mentioned product”, - Anna Artamonova, Mail.ru Marketing Director believes. Finam agrees with Mrs. Artamonova but adds there is practically nothing to win from Skype. “According to our estimates not more than 300 thousand users exploit Skype at present. The market is far from being saturated. Those who are more serious in promoting their products might gain a worthy marketplace rather quickly”.

Download QIP Infium 3,33Mb

Download QIP English
Download QIP Deutsch

SIP vs VoIP

2007-06-25T05:23:00.000-07:00

Too many people think that SIP = VoIP. Then some say: hey, what about video?! But most people stop there.

So, the consequence is that the blog world on VoIP now is starting to question whether the steam has run out of VoIP. Thomas Anglero blogs about this in Telecom's Tsunami. However, as I wrote to him in a private email:
"I have been looking forward to this. Let the tourists go home and the real innovation begin...

but I didn't write about the basic tenet underlying why I bother: That SIP together with established web-technologies is the missing real-time, two-way communication layer required to move Web 2.0 beyond the mini-apps we see today and towards the web as a real-time network of small web applications communitating and acting on behalf of users. That SIP was used to implement VoIP is good because it gets the traction, but VoIP is just a small piece of what SIP soon will do."

I realized that my comments to Thomas belonged here on the blog. So, all SIP people out there, get started innovating using SIP without VoIP, think sessions, anything that belongs in a session will benefit from SIP!

Jiri Kuthan talked about SLAMP (SER, Linux, Apache, mySQL, Perl/PHP) at the SER developers' meeting in Prague. He may be more right than he thinks. SIP Express Router is well positioned to become the apache of everything that is sessions-based on the Internet. Do you have an Ajax-based web application that needs to subscribe to a user's continously updated contact lists in order to provide that extra functionality? The answer is SIP. Do you want to track a GPS unit's movements? SIP. Do you want to push a map on a user's mobile phone while you are the phone because he calls you and cannot find your offices? SIP again.

So, get started with SLAMP innovation. Create that SER plugin and connect it to your web application. Create SIP user agents that have nothing to do with VoIP. Let us show the tourists that real innovation is on the Internet, not in the old telco world!

Mobile-to-SIP Free Client

2007-06-25T04:54:00.000-07:00

Offering free stuff can surely bring you into limelight but if you would like to survive in the long term you need to be highly competitive in order to create a mark. Fring, a company from Israel is out with a mobile phone client which offers free VoIP calls to fixed line VoIP clients such as Skype, GogleTalk and MSN besides user of their technology. The company is using viral marketing as a weapon to create a mark in the VoIP word which is already populated with a number of companies offering free calls.

With a number of companies doling out free stuff I wonder for how long they will be able to sustain it? It seems that in the long term a number of companies would lose steam and only those who have a large user base and attractive rates will stay in the race.

VTraveller Skype Phone

2006-11-29T11:05:00.000-08:00

There are several ways to communicate with friends and colleagues when you're travelling. Shouting very loudly is one option, but it's pretty useless unless the person you're trying to contact is within spitting distance. So what do you do? No, you don't pack two tin cans and a really long bit of string. You use the phone. The problem is, contrary to the popular saying, talk is not cheap. But it is if you use the ingenious Skype-friendly VTraveller. That's because this chic little phone allows you to make quality VoIP calls wherever you may roam.

For anyone who's spent the past couple of years with their head in a barrel of tar, Skype is a simple bit of software that utilises VoIP (Voice over Internet Protocol) technology. This lets you use your computer to make free calls to other Skype users and obscenely low cost calls to regular landlines and mobiles all over the world. How? Well, in a nutshell it digitally converts your voice through your internet connection (broadband is best) and on to its final destination. Clever, eh?

But what's so good about the VTraveller, we hear you ask. Well, this stylish mobile-lookalike is a fully functional USB-powered phone that works in exactly the same way as our other VoIP handsets. The difference is it's a compact one-piece unit that you can take anywhere. Just pop in the Skype CD, plug the VTraveller into your computer's USB socket, and Bob's your uncle. It's easier than, er, picking up the phone!

Despite its supreme portability the VTraveller is crammed with features including an LED message/connection indicator, one click voicemail, voice recognition technology and wideband circuitry for crystal clear sound. Best of all you can navigate all of Skype's services via the VTraveller's keypad, so there's no need to faff around with fiddly laptop mouse thingies.

Apart from a James Bond briefcase full of flat-packed supermodels, we can think of no better travelling companion than the VTraveller, as it allows you to natter away for nothing wherever you may roam. But even if you're not going anywhere the VTraveller is a brilliant investment, because it's so gorgeously compact you can shove it in a drawer when you're not using it - which won't be very often, because once you start Skype-ing on the VTraveller you'll be hooked.

link

Mobile Skype Really Sucks

2006-11-03T13:15:00.000-08:00

I just tested Skype for Pocket PC again, and got my best results ever: call quality that's only slightly worse than talking into a tin can at the bottom of a well, relayed via satellite from Iraq.

I've used the Pocket PC version of Skype on three or four devices now, and I'm always amazed at what a foul dog this program is. Echoes? Check. Static? You bet. Device slowdowns? De rigeur. Clear conversations? Not on your life. Never mind that the thing only works through wired headsets, killing off twenty-five years of the trend of holding up phones to your ear, but neglecting to anticipate Bluetooth.

Go onto the Skype support forums and you'll find a bunch of Skype fans desperately trying to sew a silk purse out of this thing: "It works using Build 42, but not Build 41, as long as you open Internet Explorer first, propitiate the rain gods, and hold your phone upside down." Forget beta - is there a letter before alpha?

Skype's attraction is low cost international calling. But there are other solutions which leverage the ease of use and quality of service of the actual voice cellular networks, yet lower your per-minute international charges, such as drugstore calling cards and Rebtel. I imagine there are also a few desperate cheapskates out there who want to make ordinary domestic calls without signing up for as many cellular minutes as they need to use. But the cost is just too high in terms of installation headaches, battery burn and awful, awful, miserable quality.

I'm not against all VOIP. Vonage gets the formula right: make it look and work just like the good ol' phone network. I've never cottoned on to those PC softphones like regular Skype -- do you really want to rely on the legendary stability of Microsoft Windows for your phone service? But I see analyst Davis Janowski chatting on his PC all the time, so I assume it works OK in terms of voice quality.

But not the Pocket PC version. It may be free, but it's not worth the price.

link

2006-09-23T07:30:00.000-07:00

JAJAH Conference Calling Limit Up To 10 Users

The NEW way to place a conference call ...
JAJAH Conference Calling is a new powerful feature that allows you to schedule and initiate cheap conference calls with your friends and co-workers in an easy-to-use online set up.

The benefits

Save up to 95% compared to other conference call solutions
Total cost transparency. The exact rate for the conference call is always shown next to the "Call" button
No costs at all will arise for the other conference call participants
Missing conference call (dial-in) is a thing of the past, JAJAH is actively ringing the participants (dial-out)

JAJAH

2006-09-18T23:10:00.000-07:00

SKYLOOK 2.0 (SKYpe & outLOOK)

Skylook 2.0 includes a major new feature called “Alerts and Forwarding”. This feature allows you to be alerted about important events that occur when you are away from your PC. These events include: The receipt of an important Email; the receipt of a Skype instant message, an important contact coming on-line, a missed Outlook reminder; a missed call or voice message. When any of these events occur, Skylook can inform you via phone or SMS. Depending on the type of event, the contents of the relevant message are forwarded to the phone via audio or SMS (e.g. voice mail messages are replayed over the phone call).

Here are some interesting scenarios where we see this feature being of use:

You use Outlook for all your appointments but you are often away from your PC, and you either don’t have an Outlook compatible PDA or you aren’t great at keeping it synchronized. Hence you often miss meetings. With Skylook, if a reminder pops up when your Skype status is Away or Unavailable, or if a reminder pops up and is not dismissed within a couple of minutes, you will receive an SMS so that you do not miss that important meeting or other appointment.
You desperately need to speak to Fred, but can’t get in touch with him. His Skype status is offline and you cannot reach him on his mobile phone. You want to know as soon as he “resurfaces”. You use Skylook to set up an alert that will call you on your mobile phone as soon as he comes on-line. Later, your phone rings - it’s

Skylook - it tells you that Fred is now online and tells you that you can press “1″ to call him now. You press “1″ on your phone and Skylook calls Fred on Skype and connects him through to your phone.

You are sweating on an important email from your boss, but you have to go out. You set up a Skylook alert to forward any emails from your boss to you via your phone. Later, your phone rings - it’s Skylook with your email message. Skylook reads the email to you using text-to-speech technology. After hearing the email, you are given the option to call your boss via Skype by pressing “1″ on your phone.

You use Skype a lot for instant messaging, but you are often away from your PC. You check a box in the Skylook options dialog to tell it to forward all instant messages to you via SMS. From that point, if your Skype status is “Away” or “Unavailable” when you receive a Skype text chat message, the message is automatically forwarded to you via SMS.
You don’t want to be available all the time, so you don’t set up Skype call forwarding, but you would like to know if anyone leaves you a voice message while you are away from your PC. That way you can choose which calls to return. You check a box in the Skylook options dialog to tell it to forward all voice messages to your mobile phone. Later, when you are away from your PC, your phone rings - it’s Skylook with a Skype Voice Mail message. After hearing the message, you are given the option to call the sender via Skype by pressing “1″ on your phone.

Skylook 2.0

2006-09-18T14:00:00.000-07:00

Intel + Siemens = Next Generation VoIP

Intel Corporation and Siemens Communications today announced the establishment of a technology relationship between the two companies that includes an agreement to work collaboratively to create open unified communications solutions based on Voice over Internet Protocol (VoIP). The two companies have agreed to jointly fund and cooperatively conduct research focused on secure wireless networks and real-time open unified communications in the short term as well as work together longer term to create comprehensive vertical industry collaboration solutions to drive business process optimization for key market segments such as telecommunications service providers, financial services and digital healthcare.

Additional components of the agreement call for the two companies to establish joint market development efforts focused on the enterprise and service provider sectors. The overall goal of the joint effort is to demonstrate real time communications solutions for business process optimization using Intel architecture such as Intel® dual-core technology and carrier class Rack Mounted Servers (RMS) from Intel and the HiPath 8000 and OpenScape™ from Siemens.
The first step is to demonstrate the OpenScape platform running Applications such as Personal Portal, Unified communications, Video and Voice Conferencing and Mobile Clients. Intel and Siemens expect to present findings and display the first wave of technology solutions developed at an Intel lab in the US to selected customers by the end of 2006.

“Intel’s experience as the leading provider of silicon technology to the enterprise market world wide and established track record of industry collaboration puts it in a great position to shape the enterprise of the future’” said Gordon Graylish, vice president of the sales and marketing group and general manager of Europe, Middle-East and Africa (EMEA). “As a result of this collaboration, enterprise customers will be able to avail themselves of unified communications based on the industry’s highest performance and most cost effective architecture.“

“Siemens has a strong commitment to this joint development effort and to ongoing collaboration with Intel to unlock the potential of our open unified communication enterprise solutions to a global ecosystem of third party independent software providers and systems integrators,” said Thomas Zimmermann, president, Enterprise Systems, Siemens Communications. “Growth potential for the open unified communications market is significant. Our own customer base for the HiPath 8000 solution has grown substantially, and Siemens is adding a new HiPath 8000 customer nearly every week.”

read more

2006-09-18T13:47:00.000-07:00

WOW! Adobe launches Acrobat 8 version for VoIP service

US-based software company Adobe systems has launched Acrobat 8 version for online conferencing and VoIP services. It has been integrated with its updated Creative Suite 2.3 Premium.

Available for both Windows and Mac, the new version provides Acrobat Connect that facilitates Internet users to discuss and edit documents or other material during the real-time conference.

The Acrobat 8 also features an Acrobat Connect professional that offers VoIP, video, whiteboard, screen sharing to any flash-enabled user.

The company said the Acrobat Connect would be marketed from early next year in English for monthly fee of $39 and $395 per year.

Sent VoIP Central

2006-09-16T08:08:00.000-07:00

Windows Real-Time Communications Client API SDK (75,8Мб)

Source RTC Client SDK:

RTC Information Agent MS Visual C++

VideoFone Application Visual Studio .NET 2003 (Visual C# .NET)

VideoFone Application Visual Studio .NET 2003 (Visual Basic .NET)

2006-09-16T07:48:00.000-07:00

O'Reilly's VoIP

Practical VoIP Using VOCAL, 07 2002

Switching to VoIP, 06 2005

Asterisk: The Future of Telephony, 09 2005

VoIP Hacks (Tips & Tools for Internet Telephony), 12 2005

Skype Hacks Tips & Tools for Cheap, Fun, Innovative Phone Service

2006-09-16T07:46:00.000-07:00

Lists of VoIP/SIP software

VoIP Phones by VoIP-Info.org
+ Wiki by VoIP-Info.org

List of SIP software by Wikipedia.org

2006-09-16T07:28:00.000-07:00

VoIP's Terms in Wikipedia

VoIP - Voice over IP

Voice over Internet Protocol, also called VoIP, IP Telephony, Internet telephony, Broadband telephony, Broadband Phone and Voice over Broadband is the routing of voice conversations over the Internet or through any other IP-based network.

Protocols which are used to carry voice signals over the IP network are commonly referred to as Voice over IP or VoIP protocols. They may be viewed as commercial realizations of the experimental Network Voice Protocol (1973) invented for the ARPANET.

VoIP - Voice over IP

Mobile VoIP - Mobile Voice over IP

Mobile VoIP or 'mobile voice over Internet Protocol' is an extension of the voice over IP technology and service. It puts wings on the classic approach of VoIP.

Mobile VoIP is more than Voice over WiFi or VoWiFi. Using any broadband IP-capable wireless network connection mobile VoIP will be an application over other networks such as EVDO rev A (which is synchronously high speed - both high speed up and down), HSDPA or potentially WiMax. Mobile VoIP will enable further economic and mobility tradeoffs. For example, Voice over WiFi offers free service but is only available within the coverage area of the WiFi Access Point. High speed services from mobile operators using EVDO rev A or HSPDA with probably have better audio quality and capabilities for metropolitan-wide coverage including fast handoffs from mobile base station to another, yet it will cost more than the typical WiFi-based VoIP service.

By mid-2006, there are an estimated 70 million users of Skype - a PC to PC service for voice communications over the Internet Protocol and some 20 million users of gateway-to-gateway voice over IP services such as Vonage, and there are a billion users of mobile phone users around the world.

Mobile VoIP - Mobile Voice over IP

VoWLAN - Voice over Wireless LAN

VoWLAN (Voice over Wireless LAN) is the use of a wireless broadband network for the purpose of vocal conversation. In other words, it's just like VOIP but over a Wi-Fi network. VoWLAN can be conducted over any internet accessible device, including a laptop, PDA or the new VoWLAN units which look and function like cellphones. VoWLAN's chief advantages to consumers are cheaper local and international calls, free calls to other VoWLAN units and a simplified integrated billing of both phone and Internet service providers.

At this point VoWLAN is not popular because of a lack of wireless coverage area. So, many VoWLAN service providers have been distributing cellphones with dual wireless and GSM modes.

Although VoWLAN and 3G have certain feature similarities, VoWLAN is different in the sense that it uses a wireless internet network (typically 802.11) rather than a cellphone network.

VoWLAN - Voice over Wireless LAN

SIP - Session Initiation Protocol

The Session Initiation Protocol (SIP) is an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include Internet telephone calls, multimedia distribution, and multimedia conferences." (cit. RFC 3261). It was originally designed by Henning Schulzrinne (Columbia University) and is specified in the RFC 3261 of the IETF SIP Working Group. In November 2000, SIP was accepted as a 3GPP signaling protocol and permanent element of the IMS architecture. It is widely used as signaling protocol for Voice over IP, along with H.323 and others.

SIP clients traditionally use TCP and UDP port 5060 to connect to SIP servers and other SIP endpoints. SIP is primarily used in setting up and tearing down voice or video calls. However, it can be used in any application where session initiation is a requirement. These include, Event Subscription and Notification, Terminal mobility and so on. There are a large number of SIP-related RFCs that define behavior for such applications. All voice/video communications are done over separate session protocols, typically RTP.

A motivating goal for SIP was to provide a signaling and call setup protocol for IP-based communications that can support a superset of the call processing functions and features present in the public switched telephone network (PSTN). The SIP Protocol by itself does not define these features, rather, its focus is call-setup and signaling. However, it has been designed to enable the building of such features in network elements known as Proxy Servers and User Agents. As such, these are features that permit familiar telephone-like operations: dialing a number, causing a phone to ring, hearing ringback tones or a busy signal. Implementation and terminology are different in the SIP world but to the end-user, the behavior is similar.

SIP - Session Initiation Protocol

2006-09-11T22:50:00.000-07:00

Google Talk Portable

This is a version of Google Talk which has the ability to run from a USB key.

It's done up the same as all of the rest of the PortableApps.com solutions. One thing to note though is that in order to comply with Google Talk's Terms and Services, I had to leave googletalk.exe out of the zip file. So all you have to do is drop googletalk.exe into the "googletalk" folder and it should run properly.

I haven't had a lot of time testing it, but everything should run properly. Tell me if there are any problems. Comments, questions and feedback welcome.

link

by portableapps.com

Download Google Talk Portable from robloach.net
Download Google Talk Portable from skypeclub.ru

2006-09-01T04:57:00.000-07:00

RhlymtYco3ok1rME7Bzx4Tx5NdFaHlYOqYgslrfsiw4=

2006-09-01T04:41:00.000-07:00

IBM Lotus Sametime

Enterprise instant messaging and Web conferencing
Real-time Right now

Instant messaging
Spell check, smiley faces, built-in VoIP and more. Chat with Mac and Linux users, too.

Web conferencing
Simpler to schedule and use. Automatic reconnections.

Social networking
Find an expert, discussions in open forums, instant polls with instant results.

Download IBM Lotus Sametime 7.5 Demo

2006-08-29T21:51:00.001-07:00

Skype Alliances

Skype and Intell

Skype* Internet calling starter kit: make free Skype to Skype calls, plus free calls within the US and Canada to mobiles and landlines until 12/31/06. For details, visit www.skype.com.

Skype VoIP demo

Hear the difference: Realize your competitive advantage, lower communication costs, and increase productivity with Skype* and Intel® dual-core technology.

Play the Video (12Mb)

(link)

Skype and Google Talk

Talking with Skype
Google Talk's got a new gig. Google and eBay have signed an agreement around text-based advertising and "click-to-call" advertising, in which Google Talk and Skype will power voice calls between customers and merchants. (Read the full press release here.)

Just as exciting are our plans to explore interoperability between Google Talk and Skype, making it easier for our users to chat with one another. This is just another step in our commitment to interoperability via open, industry standards.

Lewis Lin
Product Marketing Manager, Google Talk

(link)

2006-08-29T00:36:00.000-07:00

Linphone (GNU oSIP, OpenSource)

Linphone is a web phone: it let you phone to your friends anywhere in the whole world, freely, simply by using the internet. The cost of the phone call is the cost that you spend connected to the internet.

Here are the main features of linphone:

Works with the Gnome Desktop under linux, (maybe some others Unixes, but this has never been tested). Nevertheless you can use linphone under KDE, of course !
Since version 0.9.0, linphone can be compiled and used without gnome, in console mode, by using the program called "linphonec"
Works as simply as a cellular phone. Two buttons, and one more to chat.
Linphones includes a large variety of codecs (G711-ulaw, G711-alaw, LPC10-15, GSM, SPEEX and iLBC ). Thanks to the Speex codec it is able to provide high quality talks even with slow internet connections, like 28k modems.
Understands the SIP protocol. SIP is a standardised protocol from the IETF (http://www.ietf.org), that is the organisation that made most of the protocols used in the internet. This guaranties compatibility with most SIP - compatible web phones.
You just require a soundcard to use linphone.
Other technical functionnalities include DTMF (dial tones) support though RFC2833 and ENUM support (to use SIP numbers instead of SIP addresses).
Linphone is free software, released under the General Public Licence.
Linphone is documented: there is a complete user manual readable from the application that explains you all you need to know.
Linphone includes a sip test server called "sipomatic" that automatically answers to calls by playing a pre-recorded message.

How to use linphone ?

To call somebody, you must provide to linphone a SIP URL:
It is something like toto@machine.com, where toto is a linux user that runs linphone, and machine.com is the name of a host on a network. If you don't know the machine's name you can specify simply an IP address in dot notation (as 192.0.0.1)

If you want more information about linphone, read first the user manual.
For documentation on the internals of linphone, see section « developers » of this web site.

Compatibility:

Linphone is mostly sip compliant. It works successfully with these implementations:
eStara softphone (commercial software for windows)
Pingtel phones (with DNS enabled and VLAN QOS support disabled).
Hotsip, a free of charge phone for Windows.
Vocal, an open source SIP stack from Vovida that includes a SIP proxy that works with linphone since version 0.7.1.
Siproxd is a free sip proxy being developped by Thomas Ries because he would like to have linphone working behind his firewall. Siproxd is simple to setup and works perfectly with linphone.
Partysip aims at being a generic and fully functionnal SIP proxy. Visit the web page for more details on its functionalities.
Linphone may work also with other sip phones, but this has not been tested yet. So if you want to complete the list by testing with your own sip phone, contact me.
Linphone uses the SIP protocol to establish calls, for that reason it cannot work with H323 phones, because SIP and H323 are different and opposite protocols. H323 phones are Netmeeting (for windows), Gnome-meeting (Unix), OpenPhone...

2006-08-28T20:56:00.000-07:00

Skype Started Wifi program

Skype is a little piece of software that lets you make free calls over the Internet. We are revolutionising the telecoms world by allowing our users to call other Skype users anywhere in the world for free. In less than 2 years we’ve had over 180 million downloads (they’re free too) and we have a growing community of 55 million registered users.

Making free calls over Wifi is pretty nifty and our Wifi partners who have already opened their networks to Skype users agree. Making your network Skype-friendly means providing our users with free Skype-only web access. It’s a great way for new users to try Wifi and gives them an easy path to upgrade to full Internet access.

Here’s how to do it.
Marketing

Your Wifi start page should include a Skype banner, which you can get here.

You can check Skype’s promotional and distributional terms here.

There should also be a prominent space on your start page that lets people know they can use Skype for free from your hotspot. If you want to limit the time so that users sign-up for full Internet access, that’s fine, but we do request a minimum of 15 minutes free Skype-only access. We recommend “Talk for free! Click here to get free access to Skype.” Clicking on this should lead to a joint landing page.

The landing page should have a Skype banner, a link to download Skype and a button to allow Skype-users to connect to your Wifi network. Simple really.

Check out Skype’s affiliate program here as well. Using any of the banners provided in the affiliate program will allow you to earn commissions on sales that result from your clicks.
Technical

In the broadest possible terms, we consider an ideal network configuration to be one that’s set up according to the rules shown here:
Ideal network configuration for Skype

1. Outgoing TCP connections should be allowed to remote ports 1024 and higher.
2. Outgoing TCP connections should be allowed to remote ports 80 and 443.
3. Outgoing UDP packets should be allowed to remote ports 1024 and higher.
4. For ideal performance, the NAT translation should be stateful, meaning that translations are remembered and reused for subsequent packets. The state must be kept for at least 30 seconds after the most recent translation. (Skype recommends that the translations be maintained for as long as an hour, if possible.)

Many peer-to-peer applications, including Skype, rely heavily on UDP packets to help maintain the best possible quality of connection among peers because UDP packets can be transmitted quickly and require very little overhead to manage. However, for UDP communications to work properly for Skype through NAT, the translation rules for UDP packets must be consistently handled, meaning that UDP packets set from one external network address and port number must be consistently translated to an internal network address and port number without varying either the network address or port number.

Although the use of UDP is optional - meaning Skype will work fine without the ability to transmit UDP messages - the call quality experienced by Skype users will be much better, on average, if the caller is able to send UDP packets to the called party and receive UDP answers in reply.
Tip: Checking your network for P2P friendliness

Many of our customers have told us that they use a freeware program called “NAT Check”, written by Bryan Ford, to see if their network’s UDP translation is compatible with P2P protocols including Skype. The NAT Check program is available for free download from the program’s website at http://midcom-p2p.sourceforge.net/ and is available in a precompiled form for platforms running Microsoft Windows, Mac OS X and Linux. (NAT Check is not Skype software.)

Skype-friendly partial results from NAT Check

UDP RESULTS:

UDP consistent translation: YES (GOOD for peer-to-peer)
UDP loopback translation: YES (GOOD for peer-to-peer)
UDP unsolicited messages filtered: YES (GOOD for security)

In the results of NAT Check shown above, we see that the network’s UDP translation is applied consistently (“consistent translation”), that the input and output ports are identical except in the event of a conflict (“loopback translation”) and that unsolicited UDP packets sent to the network are discarded (“unsolicited messages filtered”).

Although not strictly necessary, it is preferable for the network’s firewall or NAT gateway to support IP packet fragmentation and reassembly. In addition, the firewall must not block an attempt to send parallel UDP packets or TCP connection attempts to multiple ports at the destination address. Some firewalls misclassify such behavior as port scanning and therefore block the host altogether. Such behavior could not only impact the ability of Skype to run but would likely impact other legitimate network applications running on the same host computer.

by Skype.com

2006-08-28T20:27:00.000-07:00

SIP Over Asterisk

How do I get SIP endpoint to talk through Asterisk?

I have two SIP endpoints provisioned on the SIP PBX and I can see SIP registers correctly. However, the two SIP endpoints can not call one another (get a "404 Not Found" SIP error message in response to an INVITE). I want to add more SIP endpoints to the PBX but need to get this simple case working first.

I have configured the sip.conf file as follows:

1000
type=user
host=dynamic
context=default
username=1000
dtmfmode=rfc2833

1001
type=user
host=dynamic
context=default
username=1001
dtmfmode=rfc2833

The sip clients are registering in the format:

Any ideas on what is wrong? Do I need to do anything with the extensions.conf file (or is the default settings sufficient to get one sip phone to call the other?)

by voip-info.org