Posts categorized "VoIP"

T-Mobile Rolling Out HD Voice (Wideband) In US Mobile Network

T mobileMarking a huge step toward moving beyond the limitations of the legacy phone networks, at the Consumer Electronics Show (CES) this week in Las Vegas T-Mobile announced that HD Voice is now available nationwide on its US network. This will give people the richer, fuller voice experience similar to what many of us have gotten used to experiencing while making Skype calls.

There is, of course, the caveat that HD voice (also called "wideband audio") is only available using specific smartphones:

To experience HD Voice, both parties on the call must use capable T-Mobile 4G smartphones such as the HTC One™ S, Nokia Astound and Samsung Galaxy S® III

TheNextWeb also suggests that the iPhone 5 should support HD Voice when T-Mobile makes it available on their network sometime this year.

Over on AnandTech, Brian Klug dives into a bit more detail about T-Mobile's HD Voice, specifically naming the AMR-WB codec, and relays some of his own testing that confirmed that it is live now.

This is an excellent step forward, even with the caveat that it only works on T-Mobile's 4G network and only with specific smartphones. As more and more people get used to the richer and better quality of wideband audio, expectations will rise and continue to push the ongoing migration of all telecom over to IP-based solutions.

Kudos to the technical teams at T-Mobile for making this happen!

P.S. I'm also personally pleased to learn about this because T-Mobile supports IPv6 across their mobile network, too. Now if only they could improve their coverage in southwestern New Hampshire, I'd be able to actually consider switching to them.


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Facebook Rolls Out VoIP In Canada on iOS!

FacebookToday, Facebook apparently began testing of true voice-over-IP (VoIP) calling from its iOS app for all Facebook users in Canada. If you have an iPhone and are in Canada, you can update to the latest version of the Facebook Messenger app and start making free phone calls to your friends on Facebook. Two articles have more details:

I was alerted to this by (appropriately) a Facebook post from Tris Hussey, author of the iPhone Hacks article.

Since I'm not in Canada, I can't test it myself... an update to the Messenger app for me will only get me the ability to leave "voice notes". But I'm looking forward to learning more from my friends in Canada.

If this rolls out to users outside of Canada, this has the potential to be huge and a major disruption to telecom. Yes, there is Skype on mobile phones, and a dozen other apps like Viber and Voxer, but...

... Facebook has the directory and the eyeballs!

You have your friend connections already in Facebook. Plus, people are already spending a significant amount of time in the Facebook app. This just makes it simple to move into real-time communications with someone.

I'm looking forward to learning more from friends up north... and to hopefully trying it out at some point!

UPDATE: Here's the iOS update message for Facebook Messenger:

Facebook v2 1 iphone

So the way I read that, we should all be getting this capability in the next few weeks.


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The Fascinating Interest in Using Google Voice With SIP Addresses

Why are so many people interested in using Google Voice with SIP? Is this a sign that people really want to use SIP-based services for VoIP? Is this all hobbyists or people looking to play around with Google Voice? Or is it people trying to solve real interconnection issues? What are people trying to do with Google Voice and SIP?

All these questions came to my mind today when I dipped into Google Analytics and noticed that for the month to date in November 2012, my old (March 2011) post about Google Voice and SIP addresses continues to receive a large amount of traffic:

Ga googlevoiceandsip

Slightly over 3,000 pageviews in the first 13 days of November - and if I go back a bit I see over 71,000 pageviews since January 1, 2012. In fact, it's had about 232K pageviews since I wrote it over 1.5 years ago, and has accounted for almost 25% of all traffic to this site in that time.

And this particular article was just one in a series of articles I wound up writing about Google Voice and SIP as we all collectively tried to figure out what was going on.

Digging into the traffic sources to the page, almost all of it this month comes (somewhat predictably) from search. The search terms, at least the ones we can see (since Google now shows "Not Provided" for all searches done over SSL), show a range of interest in SIP:

Ga googlevoiceandsip search

And all of this for a service from Google Voice which seemed to be a temporary service and subsequently stopped working... kinda, sorta... and then did work... and then didn't work. (And I just checked... and it doesn't work for me right now.)

I find all this interest fascinating. I hope it's a good sign that people out there do want to see more usage of SIP addresses.

And I do hope that at some point Google will open up the connection again and let us connect in to Google Voice numbers using SIP URIs. It would be a great move.

Meanwhile, I'll continue to be fascinating by all the traffic still coming to those old articles...


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Today's VUC Call - Setting Up A Cellular Network In The Desert For Burning Man

TimpantonToday's VoIP Users Conference (VUC) call at 12:00 noon US Eastern should be quite an interesting one. Tim Panton from Voxeo Labs and Tropo will be joining the call to talk about his experience setting up a mobile network in the middle of the desert for this year's Burning Man event.

Tim recently described the experience in a guest post at TechCrunch: "What We Learned Running A Mobile Network At Burning Man" and on the VUC call will talk more about what he did. The FAQ from the Papa Legba camp at Burning Man makes for quite an interesting read. I'm looking forward to hearing more from Tim... and the call is open for anyone to join in.

You can join the live call via SIP, Skype or the regular old PSTN. There is also an IRC backchannel that gets heavy usage during the call. It will be recorded so you can always listen later.


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Slides: How The Hidden Secret of TCP/IP Affects Real-time Communications

Recently at Voip2day + ElastixWorld in Madrid 2012, Olle E Johansson gave a great presentation outlining where we are at with telecom and VoIP in 2012 - and where we need to go! Olle is a long-time, passionate and tireless advocate for the open Internet, IPv6, SIP and standards and interoperability. I've known Olle for years via Asterisk-related issues, via the VUC calls and via work on SIP over IPv6.

This presentation (slides available) really hits a number of key points about where we are at now:

In particular I was struck by his slides 24-28 that strike the same theme I've been writing about across multiple blogs, namely the way we are reversing the "open Internet" trend and retreating back inside walled gardens of messaging:

This is what customers wanted to avoid

He goes on to walk through what happened with SIP and how the protocol evolved - and evolved away from interoperability. His conclusion is that we as customers need to take back control, avoid vendor lock-in and demand interoperability.

He also points people over to his "SIP 2012" effort where he is undertaking to compile a list of what really defines "SIP" in 2012, i.e. more than just RFC 3261. (I'll note he's looking for feedback on these ideas.)

All in all an excellent presentation... and yes indeed we all collectively do need to "WAKE UP" and demand better solutions!


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Video: What Is WebRTC/RTCWeb All About? How Does WebRTC Work?

Do you want to understand what WebRTC / RTCWEB is all about and why so many people are passionate about its potential for extending real-time communications (voice, video, chat, data-sharing, etc.) into web browsers?

I recently wrote about some of the larger issues of how WebRTC will disrupt telecom, but in this video, "RTCWeb Explained", Cullen Jennings, one of the co-chairs of the IETF's RTCWEB working group, dives down into the technical details to explain how it all works and what the various different components of of the solution are. I particularly like how Cullen covered some areas like "identity" that I haven't seen stressed as much in other pieces about WebRTC. The video comes in at about 39 minutes and is well worth viewing:

For more information, I've put together a page about the broader WebRTC / RTCWEB initiative with links to relevant resources.


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How WebRTC Will Fundamentally Disrupt Telecom (And Change The Internet)

Old phoneIf we step back to before 1993, publishing and finding content on the Internet was a somewhat obscure, geeky thing that a very few people cared about and very few knew how to do. It involved gopher servers, ftp sites, archie, veronica, WAIS, USENET newsgroups, etc., and this "World-Wide Web" service primarily demonstrated via the server at info.cern.ch. It was an amazing period of time for those of us who were there, but the number of users was quite small.

Then NCSA released Mosaic in 1993 ... and suddenly everything changed.

Anyone could create a web page that "regular" people could see on their computers. Anyone could download Mosaic and use it. Anyone could share their sites with the installation of server software.

The Web was truly born into public consciousness... the creation of Web-based content was democratized so that anyone could do it... the creativity of developers was unleashed... a zillion new business models were thought of... and the Internet fundamentally changed.

Fast-forward to today...

... and the "Web" is still predominantly a document-based system. You make HTTP queries to retrieve pages and send HTML and XML documents back and forth between web browsers and web servers.

Separately, we have a world of telecommunication apps that have moved to IP. These are not just voice, but they are also video, instant messaging, data-sharing. They have moved so far beyond what we traditionally think of as "telecommunications". The apps use wideband audio, HD video, white boarding, sharing and so many capabilities that cannot have even been remotely imagined by the creators of the PSTN and all the legacy telcos and carriers. They are "rich communications" applications that have severely disrupted the traditional telco world.

The problem is that creating those rich, real-time communications apps is somewhat of a black art.

It is the realm of "telephony developers" or "VoIP developers" who can understand the traditional world of telcos and can interpret the seven zillion RFCs of SIP (as all the traditional telcos have glommed all sorts of legacy PSTN baggage onto what started out as a simple idea).

Deploying those rich communication apps also involves the step of getting the application into the hands of users. They have to download an application binary - or install a Flash app or Java plugin into their browser. Or on a mobile device install an app onto their mobile smartphone.

The world of rich communication experiences is held back by development problems and deployment problems.

Enter WebRTC/RTCWEB

Suddenly, any web developer can code something as easy as this into their web page:

------
$.phono({   
   onReady:   function()   {
       this.phone.dial("sip:[email protected]")
 } } );
------

Boom... they have a button on their web page that someone can click and initiate a communications session ... in ANY web browser. [1 - this is not pure "WebRTC" code... see my footnote below.]

Using JavaScript, that pretty much every web developer knows... and using the web browsers that everyone is using.

And without any kind of Flash or Java plugins.

Boom... no more development problems. Boom... no more deployment problems. [2]

WebRTC is about baking rich, real-time communications into the fabric of the Web and the Internet so that millions of new business models can emerge and millions of new applications can be born.

It is about unleashing the creativity and talent of the zillions of web developers out there and turning the "Web" into more than just a document-based model but instead into a rich communications vehicle. It's about moving these apps from an obscure art into a commonplace occurrence.

We really have absolutely no idea what will happen...

... when we make it as simple for ANY developer to create a rich, real-time communications experience as it is to create a web page.

But we're about to find out... and done right it will fundamentally change the Internet again.

If we think the legacy telco crowd are upset now about how "VoIP" has screwed them over (from their point of view), they haven't seen anything yet. WebRTC/RTCWEB doesn't need any of their legacy models. It bypasses all of that in ways that only the Internet enables. It is NOT shackled to any legacy infrastructure - it can use new peer-to-peer models as well as more traditional models. And it goes so far beyond what we think of as "communication" today. [3]  The potential is there for so much more than just voice and/or video... it's about establishing a real-time, synchronous "communications" session between two (or more) endpoints - what media are used by that session is up to the apps: voice, video, chat, data-sharing, gaming...  we really don't know what all people will do with it!

I see it as the next stage of the evolution of the Internet, disrupting to an even greater degree the business models of today and changing yet again how we all communicate. The Internet will become even more critical to our lives in ways we can't even really imagine.

THAT is why RTCWEB (in the IETF) and WebRTC (in W3C) are so critically important ... and so important to get deployed.


[1] The code I'm showing is for a library, "Phono", that in fact will sit on top of the WebRTC/RTCWEB protocols. It is an example of the new apps and business models that will emerge in that it makes it simple for JavaScript developers to create these apps. Underneath, it will use the rich communications protocols of WebRTC/RTCWEB. Someone else will come up with other ways to do this in JavaScript... or python... or ruby... or whatever language. But because they will all use the WebRTC/RTCWEB protocols, they will interoperate... and work in any browser.

In full disclosure I should also note that Phono is a service of Voxeo, my previous employer.

[2] And BOOM... there go the heads exploding within the legacy telco crowd when they start to fully understand how badly the Internet has rendered them even MORE irrelevant!

 

[3] Note that a WebRTC app certainly can communicate with the traditional PSTN or other legacy systems. My point is that it is not required to do so. One usage of WebRTC will, I'm sure, be to "web-enable" many VoIP systems (ex. IP-PBXs) and services. But other uses will emerge that are not connected at all to the PSTN or any legacy systems.

Image credit: dmosiondz on Flickr


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Today's VUC Call All About The "FreeSWITCH Cookbook" - Noon US Eastern

Today at noon US Eastern on the VoIP Users Conference (VUC) Call for Friday, April 27th, the group will discuss the brand new "FreeSWITCH Cookbook"[1] published by PACKT Publishing. The four authors of the book, who are also leaders of the FreeSWITCH project, will apparently be joining the call.

While Asterisk generally gets most of the "open source VoIP" buzz, the folks at the FreeSWITCH project have been working away on their own solution. As they will say, FreeSWITCH performs a different role than Asterisk and is used in different contexts.

FreeSWITCH has become quite a powerful platform and I'm looking forward to learning more about what is going on with the project right now.

You can join the live call via SIP, Skype or the regular old PSTN. There is also an IRC backchannel that gets heavy usage during the call. It will be recorded so you can always listen later.

As noted on the VUC page for today's call, the show will also be simulcast in video using Google+ video and YouTube. If you are interested in joining the video side of the call, please follow the instructions on the page.

[1] In full disclosure, this is an affiliate link with Amazon and if you actually purchase the book I receive a tiny amount of money. If you think this influences what I write here, you clearly haven't been reading my site. :-)


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WebRTC (real-time VoIP in web browsers) On April 13th VUC Call - Join In!

Want to learn about how voice and video calls will take place right in your web browser? WITHOUT a Flash or Java plugin?

The "WebRTC" initiative is making this a reality through efforts of the major browser vendors, VoIP industry companies and standards working groups within both the IETF and W3C. On the VoIP Users Conference (VUC) Call on Friday, April 13th, the group will have a discussion of what exactly is happening with WebRTC... and then some live demos from the Voxeo Labs and Phono teams who have been working on this topic for some time now.

This is, to me, an incredibly important area of work as we have the opportunity to really bake real-time communications (RTC) into the fabric of the tools we use every day to work with the Internet.

I'm looking forward to the VUC call ("tomorrow" as I write this, but probably "today" when most of you read it) and would encourage you to join in to listen and/or participate in the conversation.

You can join the live call via SIP, Skype or the regular old PSTN. There is also an IRC backchannel that gets heavy usage during the call. It will be recorded so you can always listen later.


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Are They Crazy? Digium Enters The Phone Game With Asterisk IP Phones

DigiumphonesWhen I first saw the news today, my immediate reaction was:
Seriously? Digium is coming out with phones???
In a rather fascinating move in an already extremely crowded market, Digium announced today that they will be producing "Digium Phones", a new line of IP phones specifically targeted at users of Asterisk and Switchvox (both Digium products). They tout among the benefits:
  • Crystal clear HD Voice
  • Simple setup and installation
  • Tightest integration with Asterisk
  • Built-in & custom applications
  • A built-in "app engine" JavaScript API

There will be three models available:

  • D40—An entry-level HD IP phone with 2-line keys. Priced at $149.
  • D50—A mid-level HD IP phone with 4-line keys and 10 quick dial/BLF keys with paper labels. Priced at $179.
  • D70—An executive-level HD IP phone with 6-line keys and 10 quick dial/BLF keys on an additional LCD screen. Priced at $279

The news release indicates they will be available in April and are currently on display at ITEXPO this week down in Miami. A datasheet is available

Application Platform

What is perhaps most interesting to me is the "app engine" included in the phone. From the news release:

Digium phones include an app engine with a simple yet powerful JavaScript API that lets programmers create custom apps that run on the phones. They aren’t simply XML pages; Digium phone apps can interface directly with core phone features.

Many IP phone vendors have tried various systems like this to let developers build more apps into the phone with varying degrees of success. What makes Digium different, though, is that it comes from the developer community. The history of people working with Asterisk is the history of tinkering and hacking away on the systems. In fact, in the early days, that was all you could do. No fancy GUIs... just configuration files and cryptic APIs. As a result, Digium has a very strong developer community (they claim 80,000+ developers) who just may be able to make use of this new API.

What remains to be seen is what kind of applications you can really build with these phones - and how easy it is to install and or use these apps.

Are They Crazy?

But are they crazy for entering the already insanely-crowded IP phone market? Particularly at a time when enterprise smartphone usage is increasing - and may often be the preferred communication medium? And when people are becoming increasingly comfortable with softphones, courtesy largely of Skype and "Unified Communications" desktop apps like Microsoft Lync and similar apps from Cisco, Avaya, Siemens, IBM and more?

I completely understand that Digium would want to make the Asterisk "user experience" much easier and simpler. Particularly as Digium continually seeks to move beyond their traditional more developer-centric audience into businesses and enterprises. Many of those folks want a system that "just works." If they can order a system from Cisco or Avaya that comes complete with the IP PBX, IP Phones, etc. and it all just works, they may choose that over a less-expensive but harder-to-put-together solution using Asterisk.

As these new Digium IP phones are "designed exclusively for Asterisk and Switchvox," they should remove that pain and make it much simpler to get an Asterisk solution up and running. (Side note: Does this "designed exclusively" phrase mean they won't work with other systems? Or just that they work better with Asterisk? UPDATE: Digium's Kevin Fleming answered in the comments - the phones are SIP phones that will work with any system for basic features.)

Still, the IP phone space is incredibly crowded. One vendor of VoIP products, VoIPSupply.com, lists 382 results for IP phones. A quick scan of that list will show you names like Polycom, Snom, Grandstream and Aastra, all of whom have been typical phones used with Asterisk-based systems. (As well as Cisco, Avaya and other more "traditional" telecom players.)

What will these new direct-from-Digium IP phones do to the relationships with those other IP phone vendors?

Much of Digium's early business was with PSTN gateway cards that you could install into your computer. With much of that market moving entirely over to SIP trunking or SIP-based gateways, is the IP phone line designed primarily to replace that fading revenue line? Or to simply provide another revenue source for the company - perhaps at the expense of partners?

And what is the state of the market for IP phones, anyway? Analyst firm Frost and Sullivan says the market for SIP phones will continue growing and NoJitter's Eric Krapf has reported that IP phone vendors are seeing strong growth.

Still, with the "consumerization of IT" and the "bring-your-own-device" movement as people want to use their iPhones, Android phones, iPads, tablets, etc., it seems a curious move to launch a brand new line of IP phones.

However, Digium - and Asterisk - hasn't gotten to where it is by following the conventional wisdom. If anyone can carry off the launch of a new IP phone line, they may be able to do it. It will certainly be interesting to see where this takes them.

A new IP phone line... in 2012?

I would never have thought I'd be writing about that.

What do you think? Crazy move? or smart?


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