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December 2007

Posts from November 2007

Truphone embeds an IAX softphone into Facebook that lets you make calls to regular phones for free

200711301329The major product Dean Elwood has been working on now that he has moved to Truphone is the Facebook application that Truphone announced two days ago. Their blog provides a link to the Facebook application and, of course, in true Truphone style, offers us a video with cows:

I've not yet had a chance to do more with it than install it and play a bit with the configuration options:

but I'm very much looking forward to giving it a try. There are several interesting aspects to this app for me:

  • It is an embedded softphone (Java-based). No extra software you need. Just click the button and you can call the person who has it on their Facebook profile. To my knowledge this is the first time we've seen this in a Facebook app.
  • From the user side, you can link that button to any of the following:
    • Your Truphone number.
    • Any regular landline or mobile phones in the US or Canada.
    • A SIP address.
    • A Google Talk address.
  • A GrandCentral phone number.
  • The Facebook app uses the IAX protocol used primarily by Asterisk. This gets around all of the firewall/NAT traversal issues that plague SIP.

All of that makes for an interesting new app inside of Facebook. Now, there are already a number of "click-to-call" Facebook apps out there (some of which I've covered here) but in his announcement of moving to Truphone, Dean talks about what is different:

There are several click to call/callback/speak type applications already on Facebook. The differentiator here, and the interesting part about this application (and also the hardest part) is that we’ve embedded a JAVA based softphone right into the heart of Facebook. This makes the experience from a user point of view seamless with the Facebook environment. The user never leaves Facebook, they speak into Facebook. Additionally, the "call me" button for this application is not restricted to your own profile page - it functions as a Facebook attachment, which means it can be dropped onto a friends Wall, or added to a Facebook mail message or any other attachment-accepting application which exists on Facebook now or might do in the future.

So now Facebook users can put this "Call Me For Free" button in other locations within Facebook... and Facebook users can use this as a way to stay inside of Facebook but yet new mix in voice communication to people outside of Facebook. Now I can look up someone in Facebook and then simply click the call button to reach them by voice directly.

I look forward to experimenting with the application in the next week or two. Those of you who are Facebook users and want to try it out can simply install the application.

What do you think? Do you think people will use this app? Does voice have a role mixed into a social network like Facebook?

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Large-scale conference calls with *better* voice quality than the PSTN (using Skype)

skype_logo.pngOne interesting development in the world of Skype last week which I've seen little mention of is the fact that the folks at have rolled out a Skype Extra that lets Skype users have large-scale conference calls. Like most such large conference bridges, they have moderation/"hand-raising", call recording, email invites, etc. However, the key point to me is that their conferencing bridge uses the wideband audio supported by Skype! That is the key. You now have conference calling with audio quality that is far better than the PSTN! This is where we start to get into the space where VoIP can offer a truly different - and better - user experience than traditional telephony. The Skype blog touches on this:
HighSpeed Conferencing is the only audio conferencing service available to Skype users that offers high-definition (HD) voice quality. There’s no degradation of audio quality, no matter how many Skype users participate in a conference call. And with unlimited usage during a conference call, you can talk as much as you want. Some people stay on the conference bridge all day.
I've not yet used the service as right now I'm not involved with large conference calls, but at the point that I do I will very definitely check it out. (On a tangent, I wonder if Polycom has a trademark on "HD Voice"?) Have any of you tried it out?

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Introducing "Speaking of Standards", a new Voxeo blog about industry standards, IETF, W3C, SIP Forum, etc.

200711292028A large part of why I have NOT been writing here all that much in the past few weeks is that I've been busy in my new role with Voxeo working on a corporate blog portal. I've been covering a bit of that odyssey over on my Disruptive Conversations blog as well as in my weekly reports into the For Immediate Release podcast. It's been a great amount of work but also a lot of fun - I've been very lucky to have a colleague who does amazing things with CSS and graphics, and so the sites look a whole lot better than they would if I were left to my own devices.

I'm very pleased to say, now, that we've reached the point where I'm willing to link to our work and talk a bit about what we are doing. The main blog portal is the predictable "" but the weblog that we're really starting to use and could be of interest to readers of this blog is our "Speaking of Standards" blog found at:

I've obviously been very occasionally writing here about standards and some of that may continue, but I expect most of my writing on the subject will now occur over on this new Voxeo weblog - and I'll naturally be writing more on the subject. We'll be writing about the IETF and SIP standards, but also the W3C and standards such as VoiceXML and CCXML that I've never covered at all here. We'll be linking to events and tutorials we find and generally providing whatever information we can about standards affecting our industry, as well as Voxeo's views and implementations of those standards.

Why would Voxeo sponsor a weblog about standards? Primarily because the company and our products are all about open standards - which was one of the things that attracted me to the company after they first approached me. I've since learned that they've been leading the IVR industry in adopting open standards. As the products page says in the "Fast Facts" sidebar:

  • 100% Standards based IVR
  • Supports W3C VoiceXML 2.0
  • Supports W3C CCXML 1.0
  • Supports W3C SRGS 1.0
  • Supports W3C SSML 1.0
  • Supports CallXML 3.0
  • First platform with XML call control
  • First platform with XML conferencing
  • First shipping CCXML implementation
  • First SIP/VOIP IVR platform

Not bad, eh? Add to that the fact that our CTO (my manager), RJ Auburn, chairs the W3C's Working Group on CCXML and we've hired other folks involved with standards efforts... all of that is why we added a weblog on standards.

So if you would like to see our view on industry standards, find tutorials about various standards or learn about standards-related events we may be attending, I would invite you to come on over and check out "Speaking with Standards" - or subscribe to the RSS feed. While I (and others) will still be working on improving the site, it's mostly done and I'm delighted to be able to return to writing more. Let us know what you think!

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Congrats to Dean Elwood for joining Truphone!

200711291428Congratulations to Dean Elwood for joining Truphone as their Director of Platform Operations! I've known Dean for a couple of years now through Blue Box: The VoIP Security Podcast where he's commented from time to time and also provided us the SIP-based comment line (sip:[email protected]) through his involvement with the web site. We had a chance to meet a year or so ago at the first Blue Box dinner we had in London and Dean also hosted a dinner at VON Fall Boston a few weeks ago. He's a great guy with tremendous talent and I'm sure he'll be a great resource for Truphone. Speaking of Truphone, they are also a fascinating company to watch and I've come to know a good number of folks involved over time. I'm looking forward very much to seeing what comes out of their work and I wish Dean all the best in his new role.

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New Facebook voice app: VoxCall lets you do free calls between SIP phones/numbers

200711200958By way of my Facebook NewsFeed this morning, I learned that several friends had installed a new Facebook app called "VoxCall" (must be logged into the walled garden of Facebook to see the link). A quick Technorati search brought me to Alex Saunders' blog post on the subject which clued me in to the fact that this was from the folks at Voxalot, some of whom I'd met down at Fall VON in Boston.

[Side Rant: This also shows the inherent weakness and stupidity of Facebook's current implementation of "groups". The Voxalot folks had posted info about this app in both the news and wall of their Facebook group, but of course I would never see it unless I just randomly happened to go there. Had they sent a message to all group users, I would have seen it in my Facebook Inbox, but it would be nice if instead Facebook had some way to notify you that you had new info in the groups to which you subscribe.]

The VoxCall app is basically a "click-to-call" app that makes use of Facebook's directory. You simply click on the name of someone else who has the app installed and, like many click-to-call apps, you are called first and then the other party is called and the connection is made.

An interesting aspect is that VoxCall works with SIP URIs (addresses). When you install the app you have to enter your SIP URI at which point you then receive a call on that URI where you are asked to enter the PIN displayed on the screen:
It's actually a pretty nice way of authenticating the endpoint. Given that Voxeo's a VoIP application platform company, we naturally all have SIP URIs for our extensions (sip:[email protected] for me) so it was easy for me to sign up. Users of Gizmo would likewise have a SIP address, as would users of many other VoIP services. If you don't have a SIP URI, Voxalot has a suggested path to get one on their VoxCall FAQ. (One thing I don't completely understand is why you would need to do their step #2, Register for a VoxPremium account, if you already get a SIP URI from the Voice Service Provider you signed up with in step #1. But maybe the point is that some of those VSPs won't give you SIP URIs... ?)

Once registered, the process is quite simple. You have a "Call Friends" tab that is shown below (complete with some advertisement being blocked by the local proxy server that I run that blocks ads from typical ad-serving sites):
You simply click on the person's icon and the call process starts. First it calls you, then it calls the other party. No charges incurred by anyone outside of whatever inbound connection fees we would normally pay (in my case, none). I called Alec and so my page changed to show his picture and the fact that I was calling him:
Alec and I had a good chat with surprisingly good audio quality given the convoluted path our call was taking. I was on a Polycom IP phone connected across the Internet to Voxeo's SIP servers in Florida. The call went across some network cloud to Alec's TruPhone number (which has a SIP URI) which wound up ringing his mobile as he was driving along the 401 somewhere in southern Canada. Audio quality was quite good and didn't seem to have any real issues in the 5 or 10 minutes we chatted.

The VoxCall app also has an Echo Test number you can call to hear the latency and has some conference rooms that I have not yet tried.

Overall, it's an interesting app, although I guess my basic question is simply this: will I use it? As I wrote earlier, the phone is no longer as critical of a communication tool for many people, myself included. When I think of Facebook, I think of it as a place for email-ish communication. If I need to reach someone urgently, I have used Facebook as a place to get a phone number from in the past. Will I think to use to it place a call in the future? I don't know.

There are a couple of barriers to that, really. First, the app only works with people who have it installed. Second, to install it you need a SIP URI and the whole concept of SIP addresses is only really now starting to come to people's attention (outside the early adopter crowd). Third, initiating the call requires going into the VoxCall application page inside Facebook to click on the person's icon to call. It would be nice if it could be done simply from the list of friends that you have. (Having said that, it's actually easier to simply go into the app page than it is to search through Facebook's friend list and then go into their profile to then click on a link below their picture.)

The nice thing about the app, though is that it does use the Facebook directory. As Alec puts it:

Perhaps the biggest differentiator for Voxcall is simply that it hooks into a directory that a lot of people know and use.

As Facebook continues its climb in popularity and moves onward toward the goal of being your definitive "portal" to the Internet, this VoxCall app (and others like Alec's own Free Conference Call app) help connect in voice to the communications mix (for those who still want/need to use it).

In any event, kudos to Voxalot to bringing out another voice app on top of Facebook. It's good to see the platform being used for voice. As a advocate for SIP and open standards, I applaud apps that promote the use of all things SIP. Give it a try. What do you think of it? (Feel free to give me a call if you are a Facebook friend of mine.)

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I'll be out in Vancouver Dec 2-7 for the 70th meeting of the IETF.

200711191406Just confirmed travel plans today - I will be heading out to the 70th meeting of the Internet Engineering Task Force (IETF) in Vancouver, British Columbia, Canada, from December 2-7. If any readers will be out there (either for the IETF or in Vancouver in general), please do drop a note and let me know. This will be my first meeting in my new role with Voxeo and I'm very much looking forward to renewing old acquaintances and also getting more directly involved with the work of the IETF.

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Verizon brings in 40 Gbps IP circuits... OC-768, anyone?

200711191352Having been online now since the early 1980s and having watched/used the ever-increasing amounts of bandwidth we have available, it still made me pause to read that Verizon Business has launched 40Gbps connections on its backbone using Juniper Networks routers.

40 Gbps?

Pretty mind-blowing, considering where we have come from. I'll spare you all the tugging on my beard (that is now gray in spots) and reminiscing about how we all had to connect at 110 baud using acoustic couplers... I will say that back in the late 1990's when I wrote the Networking Essentials Exam Guide, I did cover the OC-n naming convention for connections, but I don't think we would ever have imagined that some day there would be "OC-768". Pretty cool!

Not that we as consumers will necessarily see that bandwidth anytime soon... but it is nice to know it is there in the backbone. (Also interesting that they note that the growth of their VoIP offerings is one of the reasons for being interested in this backbone.)

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Did you know RFC 4733 had replaced/obsoleted RFC 2833 for DTMF signaling in SIP?

Did you know that RFC 4733 replaced/obsoleted RFC 2833? I just learned this myself through a SIP Forum mailing list exchange the other day. For those not aware, RFC 2833 and now 4733 define methods of carrying DTMF signals (and other similar signaling) in RTP streams separate from the main audio component of the RTP stream. A typical example of use might be where you were using a highly-compressed audio codec for audio between two SIP endpoints where the high degree of compression might make it challenging for the DTMF tones to be correctly interpreted on the receiving end. Using "RFC 2833 compliant" signaling, the sending SIP endpoint would send those DTMF tones as separate packets within the RTP stream.

My key takeaway from learning about RFC 4733 is that we should really be talking about "RFC 4733 compliant" signaling... but given that the industry is really only now starting to really talk about "RFC 2822 compliant" signaling, I'm not sure I expect to see that happening anytime soon.

Anyway, here's the abstract from RFC 4733 - you can naturally read the rest of the document to understand more:

This memo describes how to carry dual-tone multifrequency (DTMF) signalling, other tone signals, and telephony events in RTP packets. It obsoletes RFC 2833.

This memo captures and expands upon the basic framework defined in RFC 2833, but retains only the most basic event codes. It sets up an IANA registry to which other event code assignments may be added. Companion documents add event codes to this registry relating to modem, fax, text telephony, and channel-associated signalling events. The remainder of the event codes defined in RFC 2833 are conditionally reserved in case other documents revive their use.

This document provides a number of clarifications to the original document. However, it specifically differs from RFC 2833 by removing the requirement that all compliant implementations support the DTMF events. Instead, compliant implementations taking part in out-of-band negotiations of media stream content indicate what events they support. This memo adds three new procedures to the RFC 2833 framework: subdivision of long events into segments, reporting of multiple events in a single packet, and the concept and reporting of state events.

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Want to see the people I work with? - Voxeo's office and people... as seen via Flickr

Originally uploaded by voxeophoto
As most of you know by now I'm now employed by Voxeo and the folks down in the Orlando office recently started using some camera's to upload pictures to company Flickr stream. We did this largely because we're hiring (more job openings to be posted soon) and we want potential candidates to see what a fun place it is to work... but it will also factor into some of the other blogging and other work we'll be doing.

Anyway, you can check it out. No photos of me there, yet, since I wasn't around when the camera's were being passed around. (I'll upload a picture of me to the site, though, soon.)

Jeff Pulver on the status of VoIP-related legislation in the USA...

I haven't written much here about the state of VoIP-related regulations in the USA, but Jeff Pulver just did yesterday on his blog with his post, "VoIP in America: The State of VoIP". I would encourage you to give it a read, even if you don't live in the USA. It's great that we all are building useful technologies... but we also need to make sure that government regulations do allow us to use those technologies.

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