Showing posts with label SIP. Show all posts
Showing posts with label SIP. Show all posts

C2Call - New browser-based Java VoIP app

C2Call (C2Call) lets you make web-based P2P VoIP phone calls direct from your browser whether you're using Linux, Windows, or an Apple Mac. The software is a Java widget that is loaded temporary from the C2Call web page into your browser each time you use C2Call. Using the C2Call Java widget I spoke with Martin Feuerhahn, Director of Actai Networks Pte Ltd Taiwan Branch. The voice quality was excellent and the latency wasn't too bad either considering I'm in Connecticut and he's in Taiwan! In Internet Explorer when you first load the widget you'll see that the application has a digital signature that has been validated by a trusted source: Actai Networks GmbH, the company mentioned in the above screenshot is the parent company of www.c2call.com. In Firefox the app just ran without showing the digital signature verification message. The client is just 1.1MB in size so it's fairly small. The beauty of C2Call is that you can send a call invitation to anybody you want to talk to simply by sending an email invite directly from the www.c2call.com applet. The recipient can click on the link and can instantly talk to you. This is a really great viral feature that is especially good for non-technical people. For instance, you can invite your grandma to talk simply by emailing her. Unlike Skype which would require her to install the software, register a username, configure Skype, etc., grandma can just click a link and speak to you almost instantly. Here's a sample email invite being sent out to my boss and industry-leading VoIP pundit, Rich Tehrani. The widget is SIP-based and importantly it uses the Speex codec, which gives the software its very good voice quality. So you're probably wondering how a Java-based VoIP app fares versus a Flash-based VoIP app, such as the Flashphone app I've written about. Well, for one if you decide to use an Adobe Flash-based solution, you have to use the Nellymoser codec, which is not a free codec. Second, Adobe Flash requires the use of a media server, which Martin claims is not as scalable as their true P2P (peer-to-peer) VoIP solution. Another feature of note is that C2Call can traverse about 90% of firewalls using standard SIP technology. The official launch isn't for another couple weeks, but if anyone wants an invite give me a shout (Contact link above).

Skype and other VoIP Apps on Mobile Phones

It seems new software comes out daily that enables you to run a VoIP application, such as Skype, on your mobile phone natively, through a bridged gateway, or using the Skype API. While Skype is certainly the most popular VoIP application, other software solutions exist that use the SIP standard. Still others provide SIP-to-Skype bridging capabilities. I thought it would be useful to have a rundown of the various solutions out there. SoonR Talk - I broke the news about this first software application to enable VoIP on the Apple iPhone. It features a nice clean interface that emulates the Skype client look and feel, as seen in the screenshot to the right. The way it works is that the AJAX interface acts as a remote control for your computer where the SoonR service is installed along with Skype. There is nothing to install on your phone because SoonR Talk uses a browser that supports AJAX. You simply login into SoonR website. Shape Services' IM+ for Skype - lets you run Skype on the iPhone with their IM+ for Skype application. IM+ for Skype, also works on the iPod touch. The beauty of IM+ for Skype is that it doesn't even require Skype to be running on your PC - it leverages the Skype API and servers run by the IM+ for Skype folks (Shape Services). You can find out more info in my Skype on the Apple iPod touch article. SIP to Skype calls - Check out my article here for a how-to step-by-step guide that lets you deliver SIP calls directly to Skype. Essentially you can have a SIP DID number ring your Skype client. The workaround uses Net2Max.com's One Click Contact number (1CC number) to make this possible. SippySkype - Java software that allows you to make and receive Skype calls from your SIP/VoIP adapter or SIP softphone. Basically it's a Skype/SIP Bridge/Gateway/Proxy. You'll need a SIP client on your mobile phone for this to work. But assuming your mobile phone can run a SIP client, you can make SIP-to-Skype calls. iPhoneGnome, enables PhoneGnome calling from your iPhone. You can call other PhoneGnome numbers as well as anyone using SIP-based services, FWD, MSN, Yahoo, Google Talk, and any other numbers that you can call from your PhoneGnome account. I should stress that there are non-Skype VoIP apps that run on mobile phones, many of which target the Apple iPhone, but not all. This next example, Jajah does support the iPhone, but they actually support any mobile phone. Jajah - They have several mobile phone-to-VoIP solutions available. There's the JAJAH Apple iPhone and iPod Touch App. There's Jajah Direct, which allows you to simply dial a local access number to talk globally - no Internet or data access needed and you can skip having to dial from your PC using VoIP software (i.e. Skype). There's also the Jajah native SIP client. And of course there's Jajah Mobile Web (http://mobile.jajah.com) which gives you one-click access to their free or low-cost global calling service, directly from the browser, which works with any mobile phone with web access since no application download is needed. truphone - truphone has native VoIP applications for several mobile phones. It currently works on many of the popular Nokia N-series (eg, N95) and E-series phones (eg, E90). truphone leverages the SIP standard for all of their softclients. Also, truphone supports the Apple iPhone and was the first to offer a true 100% native VoIP application on the iPhone. You have to jailbreak your iPhone to install it though. Sipcall - This is an exciting new offering currently in beta. They offer a free Skype and SIP calling service called Hipsip that works from any mobile phone's web browser. You simply install their bridging software on your PC and then from your mobile phone you can access Hipsip and call any Skype or SIP user. The software will make a cell phone call to a local Hipsip number automatically. Hipsip converts the call to VoIP and bridges the call with the Skype or SIP user you dialed. (Hat tip to Markus Göbel for discovering this software and for the screenshot. Go check out his recent write-up!) Talkplus - unique mobile phone service that leverages VoIP, identity management, scheduling/contact rules, and other advanced business/personal processes that make your mobile experience more productive while enabling privacy controls. TalkPlus gives cell phone users a 2nd virtual phone number with powerful accessibility and privacy rules while leveraging the cost savings of VoIP termination. TalkPlus's patented technology is network agnostic, and supports standard cell phones that support the BREW and JAVA platforms. They also support native clients for Nokia phones, Windows Mobile, Blackberry, and Palm. TalkPlus is also WAP compliant which virtually all mobile phones support. One interesting thing I talked about at ITEXPO with Talkplus CEO Jeff Black is that TalkPlus has built their own Skype gateway. In fact, when pressed further, Jeff mentioned they actually reverse engineered Skype's protocol. Although the Skype gateway isn't part of TalkPlus's launch today, Jeff explained that they have tested it in their labs and it's working very well. iSkoot - lets you call your buddy list from your mobile phone without needing your PC. iSkoot software is simply a software download to your mobile phone to enable you to make Skype calls. Supported phones include Nokia Series 60 handsets: 6600, 6620, 6630, 6670, 6680, 6682, 7610, N70. They also support Motorola RAZR, Treo, and Nokia Series 40 handsets. Windows Mobile and Blackberry phones are also supported. iSkoot Mobile Network allows consumers to place and receive Internet calls with their mobile phones without the need for PCs or Wi-Fi hot spots. Mobivox - Mobivox allows you to make free calls to other Skype users as well as free calls to other Mobivox users. What's unique about Mobivox is they offer 3 ways of initiating a call using ANY mobile phone. First, you can call a local access number and VoxGirl asks you which number you want to dial. Using speech-recognition it will connect the call. Second, you can use web call back. Logon to your Mobivox account and click who you want to call and Mobivox will connect you both. Third, using SMS you can 'text' MOBIVOX the name of the person or number you wish to call. VoxGirl will call you back and connect you. fring - It's a thin client that allows you to talk & IM via your handset's Internet connection to other mobile phones and PC-based services such as Skype, Google Talk, MSN Messenger, SIP and Twitter for free. It also unifies all of your various contacts across all of these IM/VoIP networks so you can use a single app to view presence and initiate chat sessions or voice calls. It works on Nokia and Windows Mobile phones. You can also share files using fring. eqo mobile-to-Skype solution - Essentially, eqo acts as the "glue" betwen online communities such as LinkedIn, MySpaces, orkut, or Skype with mobile carriers such as T-Mobile, Cingular, Vodaphone, etc. Their architecture is a patent pending core IP that uses proprietary call signaling, that is P2P-based and cross device/network domain. It leverages mobile data (SMS) to control the signalling and the mobile voice network for the media stream to the mobile phone, so you don't need the Skype client running on your cell phone. Essentially, their software client is a J2ME application that resides on your cell phone and which keeps your buddy list synched with Skype. eqo did a good job at keeping the application small since it's only a 121k JAR file, most of which is the graphics for the user interface. Their latest client, which still runs on Java compatible phones, lets you make VoIP calls, as well as instant message across networks (MSN, Yahoo, AIM, GoogleTalk, ICQ, Jabber). And there you have it. Several ways to VoIP on your mobile phone using both Skype and non-Skype VoIP solutions. What's your favorite mobile VoIP application?

SippySkype SIP-to-Skype Gateway

Last month I wrote about my strong disagreements with a guest blog post on Skype Journal that was titled "A SIP/Skype Gateway Is NOT In The Forecast". I disagreed with the premise that a SIP/Skype gateway isn't in the forecast. Well, Skype Inc. may not care about offering a SIP-to-Skype gateway, but that doesn't mean VoIP enthusiasts are going to sit around and not build their own SIP-to-Skype gateways! I've blogged on a few occasions where I discussed the desire for Skype users to have SIP connectivity and various home-brewed solutions. Well, we can add another SIP-to-Skype gateway solution to the mix. I just discovered SippySkype today, which is an open source & free SIP-to-Skype gateway. Check out the features: SippySkype is Java software that allows you to make and receive Skype calls from your SIP/VoIP adapter. Basically a Skype/SIP Bridge/Gateway/Proxy Call Skype Users using speed dial or use Skype out. SIP to Skype authentication/denial mappings via SIP caller ID and IP blocks - 1.1 or higher Skype to SIP authentication/denial mappings via incoming Skype User ID - 1.1 or higher Support RFC2833 touchtone decoding (DTMF) - 2.0beta or higher Could be used as an endpoint with Asterisk Auto play pre-recorded file(s) to SIP callers - 2.0beta or higher Incoming SIP Pin number authentication and dialing - 2.0beta or higher Open Source - You can modify/fix it if you like. It's free System Requirements: Skype Client Working Java 1.6.0 or better runtime mjsip/mjua 1.6 http://www.mjsip.org/ - Use those included with SippySkype as some bugs have been fixed. Skype4Java 1.0 https://developer.skype.com/wiki/Java_API - Unmodified SIP/VOIP adapter such as a spa-3102 to make and receive Skype calls or register with a provider or Asterisk. Should work where Skype4Java works (windows/linux/osx). (I'm using it on Windows XP)

Talkonaut 4.0 native Symbian S60 edition released

The GTalk2VoIP dev team just released their VoIP+chat application for Symbian S60 phones that runs as a native (.sis file) application. I should stress that several Nokia smart phones come with a SIP stack, however they are limited to WiFi use only and won't work over your cellular network. You can thank the carriers for that one. Talkonaut on the other hand is not bound to WiFi only, allowing you to make calls over GPRS, EDGE, 3G or WiFi. Talkonaut has essentially developed their own VoIP SIP stack that allows Talkonaut users to make voice calls over most data connections your mobile phone might have, such as GPRS, EDGE, 3G and WiFi. The application also sports the ability to make free VoIP calls to Google Talk users, to SIP phones, to MSN, Yahoo, AIM and ICQ voice capable IM clients. I believe you can even chat with MSN Messenger, Yahoo, AIM, and ICQ users. Here's a list of the new features in the 4th release according to the release: - Talkonaut 4.0 was entirely rewritten in C/C++ and now runs on Symbian S60 3rd edition based Nokia smart-phones. Moving from Java to native platform allowed to reduce memory usage, improved speed and to add some functionalities that were not previously available for Jave applications, like VoIP, access to file local system (improved file transfer) and interaction with other applications (Web Browser for opening URLs right from chat window). - Using a set of narrow-band Speex codecs and relying on Jingle Audio extension to XMPP protocol (same as implemented in Google Talk), Talkonaut now brings to the world a very powerful combination of IM chat, Presence and VoIP calls made over data connection. - Talkonaut 4.0 allows to make free VoIP calls to Google Talk users, to SIP phones, to MSN, Yahoo, AIM and ICQ voice capable IM clients, as well as to other Talkonaut fellows. - Talkonaut 4.0 allows to receive free calls from SIP phones (or any other VoIP networks), from Google Talk, MSN, Yahoo, AIM and ICQ users. - Talkonaut 4.0 allows to make cheap VoIP calls to any mobile or landline phone number in the world. - Talkonaut 4.0 allows to define any number of SIP accounts and use them to make free or cheap calls over third-party VoIP/SIP carriers. A flexible Dialing Plan feature is helpful for choosing routes to destinations between different carriers. - Talkonaut 4.0 is fully equipped with all the features previously available in J2ME version of Talkonaut 3.0, inherits the same graphical user interface and menu structure. Talkonaut 4.0 runs on the following sets of Symbian S60 based Nokia smart-phones: o 3rd Edition: 3250, 5500, 5700, 6110, 6120, 6121, 6290, E50, E51, E60, E61, E61i, E62, E65, E70, E90, N71, N73, N75, N76, N77, N80, N81, N81 8GB, N82, N91, N92, N93, N93i, N95, N95 8GB Talkonaut 4.0 can be downloaded over-the-air using mobile web browser from http://get.talkonaut.com/, or via PC from http://www.talkonaut.com/download.shtml

Junction Networks Adds Inbound Bridge to onSIP Hosted PBX

Junction Networks, provider of the hosted onSIP PBX service, has launched Inbound Bridge, an accessory service that ties in third-party providers of international and domestic DID numbers. Inbound Bridge saves money for Junction Networks’ business customers by allowing them to find the best per-minute price for inbound VoIP calling minutes in their chosen geographic regions. Essentially this bridging application lets hosting customers use third-party providers of DID numbers for best LD price in desired footprint. According to Junction Networks, "Inbound Bridge solves a vexing problem for companies that want local numbers and 800 numbers in foreign countries or specific domestic regions—numbers typically rented from other VoIP network providers. Many of these providers simply deliver calls from one end point to the other; they don’t implement the key function of the SIP VoIP signaling protocol that allows callers to navigate phone menus, transfer calls, put callers on hold and trigger other events. Without this function (specified in the IETF’s RFC 3515), inbound callers are disconnected when they “press 1 for sales,” or try to transfer to extension numbers for specific people. Or at the least the DTMF touch tones are not recognized. Junction Networks' Inbound Bridge performs the missing SIP function, maintaining the two-way SIP signaling needed for interactivity after a call is already established. It can therefore pass the entered touch-tone digits to the auto attendant and other PBX applications. It can integrate onSIP with any network vendor having an open SIP implementation (permitting communication with other SIP networks). “This is a perfect opportunity if you’re a company that wants a toll-free service from Europe or Asia, or you have some other reason to use a third-party phone number provider,” said Rob Wolpov, president, Junction Networks. “But you do want to use our hosted solution, because it works very well, you have company extensions in one or more sites, and you like the price of that.” ($39.95 a month for core voice applications and an unlimited number of SIP extensions, with free calling between them.) “You can get your toll-free origination from any provider that offers SIP delivery of inbound calls, and just have it go to us through the Inbound Bridge. We charge $1.95 a month per DID, and half a cent a minute to cover our cost of the Bridge. If you can get your toll-free for one or one and a half cents a minute, you wind up paying two cents a minute for toll-free, as opposed to paying almost four cents a minute to Junction Networks. “That’s completely OK with us,” Wolpov stressed. “Go somewhere else to get your minutes and then come to us for the auto attendant, the ACD queue, the voice mail, all those sorts of applications.” To date, Junction Networks has tested the Inbound Bridge with international DID provider Voxbone. Other third-party DID providers are to be tested and added in the coming months. “We’re happy to supply onSIP customers with our core value—locally dialed phone presence in more than 40 countries around the world,” said Rodrigue Ullens, Voxbone CEO. “With our intercontinental voice-only backbone, Junction Networks’ Inbound Bridge and onSIP platform, these enterprises can get the best value in DID numbers and international VoIP transport, plus all the convenience and flexibility of hosted IP PBX.”