Posts categorized "VoIP"

Remote Working: the Benefits, Disadvantages, and some Lessons Learned in 15+ years

Djurdjica-boskovic-G8_A4ZWxE3E-unsplash-776pxWith so many people now having to learn to work remotely due to restrictions related to COVID-19, what information can people share who have been working from home? Back in October 2019, I realized it was 20 years ago when I started working remotely, and so I sent out some tweets asking for opinions about the benefits of working remotely, the challenges / disadvantages, and then the lessons people have learned. I subsequently recorded podcast episodes on each of those three topics.

The links to the Twitter threads and podcasts are below.At some point I may turn them into longer articles themselves, but in the meantime, I hope they will help some of you with ideas for how to get adjusted to this new way of working.

And… I would suspect many of you might just want to jump directly to the lessons learned… 

Benefits

Many of the benefits were about no commute, the ability to be present with family, freedom to work and live wherever, flexibility, caring for family, and more.  (Note that a good number of the benefits mentioned (such as working from "anywhere") are currently NOT possible because of the self-isolation / quarantine imposed by the COVID-19 situation.)

Disadvantages

Loneliness, isolation, and the lack of social connections with colleagues topped the list of disadvantages, along with the lack of physical activity, home distractions and more.

Lessons Learned

Some of the key lessons that I have learned in over 15 years of working remotely, and that were common in other comments include:

  • Create a separate space (ideally, a separate room) - this is critical if you can do it.
  • Invest in a good chair and other office equipment - since you are going to be sitting in it so many hours of your day! (Or some people now have desks that let you stand, too.)
  • Make time for physical activity - get OUTSIDE if you can! Go for a walk. Go for a run. Or work out in a home gym. Multiple people suggested dogs being a great way to force you to do this.
  • Make a schedule - and STICK to that schedule - it is super easy to work many hours at all different times. Figure out a schedule that works for you,  your employer, your team, and your family - and then try to stick to that schedule.
  • Use collaboration tools - things like Slack are critical for your own sanity so that you are “connected” to other people in your organization. (Granted, you may need to figure out how to not be too connected to everyone and spend your day drowning in notifications!)
  • Take actual lunch breaks - step away from your computer and your home office. Get up and move around.
  • Sit with your face toward natural light, if possible - it looks better than artificial light… and you’ll get some Vitamin D, too. 🙂
  • Lighting IS important, particularly for video calls - you do want to have light shining on you in a way that works well for video. You may want to experiment with different lamps around you or on your desk.
  • Have video calls with other remote workers - make time to connect with colleagues, ideally over video calls. Even if it is just to chat for 5 or 10 minutes. It can help ease the sense of isolation - and they may like it, too! Sometimes if I have a question that I’m going to write in email or Slack, I’ll ask myself, “would it be faster if I just ask them in person?” And if so, I’ll ping them via a message to see if they are available for a video call.
  • Work in different locations - Try sometimes to get out of your home office and work in other parts of the house. Take a laptop and work in another room, or on a deck or yard if you have one. (Granted, this might be hard if you have many people in your household all working in the same building.)

On this last point, you’ll see in the Twitter thread and hear on the podcast all the comments about working from other locations. For example, working at cafes with WiFi, etc. That IS a critical lesson many of us have learned. Successful remote working can involve getting outside the walls of your home office - and outside of your home. Obviously this is currently NOT possible with the COVID-19 situation, but something to definitely think about if you continue working remotely once we are past all of this.

Other remote workers… what other lessons learned would you add?

Best wishes to you all as we all try to navigate this new world of social distancing and working remotely over the next weeks and months!


UPDATE #1 - over on Twitter, someone I know pointed out that this is NOT regular "working from home" (WFH). His text: "I've WFH 11 years. current situation is not normal WFH. you can't go to a coffee shop to interact w people, work out or take advantage of all sorts of WFH perks like normal.
self-quarantine != WFH
"

I definitely agree, Paul, this is NOT regular "working from home".


Photo by Djurdjica Boskovic on Unsplash. - No, that’s not MY desk… far too clean! 😏


The Publishing of RFC 8496 Concludes the 10-year Saga of P-Charge-Info

Rfc 8496 p charge info

October 31, 2018, was a special day for me. Not because it was Halloween, but because after 10 years a small little document I co-authored about the "P-Charge-Info" header for SIP-based Voice-over-IP (VoIP) was published as informational RFC 8496. You can see it at either:

Ultimately, all this document does is register the Session Initiation Protocol (SIP) Header Field of "P-Charge-Info" within the "SIP Parameters" registry maintained by IANA at:

But the story of getting that registration to happen is a long one!

In the beginning...

The short version is this. Back in around 2007 or so, I was working for Voxeo and we were using the "P-Charge-Info" header in our large SIP-based application server to pass along billing information. Essentially, when someone made a call on our system, we wanted to pass a billing identifier that was often different from the source phone number (i.e. "CallerID"). This quote from RFC 8496 was pretty much Voxeo's use case:

As another example, a hosted telephony provider or hosted voice application provider may have a large SIP network with customers distributed over a very large geographic area using local market PSTN numbers but with only a very few actual PSTN interconnection points.

The customers may all have local phone numbers yet outgoing calls are actually routed across a SIP network and out specific PSTN gateways or across specific SIP connections to other SIP service providers. The hosted provider may want to pass a billing identifier to its SIP service providers either for the purpose of simplicity in billing or to obtain better rates from the SIP service providers.

While we at Voxeo were already using P-Charge-Info extensively, we wanted to use the P-Charge-Info header with more SIP service providers, and needed some form of documentation for how to use the header. We also were concerned about the profileration of more P-headers and wanted to register "P-Charge-Info" with IANA so that more people might find and use that header rather than inventing their own. (We were happy with P-Charge-Info and didn't want to have to support more SIP headers related to charging identifiers.)

So I began the process of submitting an Internet Draft way back in February 2008 documenting P-Charge-Info and requesting its addition to the IANA registry.

Almost immediately Tolga Asveren, then at Sonus Networks, contacted me to let me know they were using P-Charge-Info in the SIP equipment they were selling. He had some great suggestions, supplied some additional text, and was interested in working on the document with me. So I published a -01 draft 3 days after the first draft, and Tolga and I began our 10-year journey through the process of getting this document published.

Square pegs in round holes - slamming ISDN info into a SIP header

Along the way, others approached us from traditional PSTN telecom companies indicating that they were interested in using P-Charge-Info as a way of passing the "ISUP Charge Number" that was part of ISDN signaling. Though it was not something that either Tolga or I worked directly with, we were okay adding this in, and so two new parameters got added:

  • Numbering Plan Indicator (NPI)
  • Nature of Address (NOA)

... along with a substantial amount of new text.

A 2011 version of the document can show what this was all about.

Lost in limbo

Meanwhile, the document had gotten caught up in the need to wait while RFC 3427 was replaced by RFC 5727, which defined the whole SIP Header Field registration process.

And then I wound up leaving Voxeo to join the Internet Society (my current employer). And so my attention was no longer focused on VoIP and so this draft was truly a "back burner" kind of thing that I just worked on in random moments.

And unfortunately, it turned out that slamming legacy PSTN signalling into a SIP header caused a whole number of challenges, both with getting agreement - and also with some of the internal IETF processes. It turned out that to do this registration, we were going to have to do some other registrations - and more.

I also published a version that changed some of the parameters values in a way that was not backwards-compatible, which caused some friction.

By 2015, both Tolga and I were ready to ... just... let... it... die...

Returning from the dead - and returning to the SIP roots

And then in early 2017, Henning Schulzrinne and Richard Shockey contacted me to let us know that the US FCC was interested in the status of P-Charge-Info. The FCC had some billing issues between carriers where the carriers were in some cases already using P-Charge-Info, but the carriers really wanted an actual RFC versus an expired draft. There was also some potential interest in having the header around as one of many different tools in the FCC's efforts to combat robo-calling / spam phone calls.

So Tolga and I worked with Henning and the IETF area directors to come up with a plan to resuscitate the document. In doing so, we stripped out all the legacy PSTN signaling. Specifically we removed the NPI and NOA parameters, and all the mentions of the ISUP Charge Number.

We returned it to the simplicity of the original document way back in 2008.

The original goal had been to simply document existing usage of the P-Charge-Info header, as it was being used by various SIP providers and vendors. We returned it to that root.

Success!

After a number of further drafts, an expert review by Adam Roach, and more feedback from Area Director Ben Campbell, the draft finally entered the RFC Editor queue by way of the Independent Series Editor (ISE). Many thanks are due to Ben to sponsoring the draft. He, Jean Mahoney, and Adam were all critical in helping get it across the finish line. Thanks, too, to Henning and Richard who provided the spark for us to revive the document.

It would not have happened, either, if Tolga had not taken the lead over the past year in making edits, answering all the questions from people, proposing solutions and continually asking how to move the draft forward. I may have been the one editing the XML and submitting the new draft versions, but he was the one driving the text revisions.

It's great to see the document finally published as Informational RFC 8496. Looking back on the journey, there were:

  • 16 revisions of draft-york-sipping-p-charge-info (2008-2012)
  • 6 revisions of draft-york-dispatch-p-charge-info (2012-2015)
  • 9 revisions of draft-york-p-charge-info (2017-2018)

Now, granted, some of those were a simple "update to keep the document from being 'expired'", but still it was all a great amount of work.

I learned a HUGE amount along the way. When the journey began in 2008, I didn't really understand much about IETF processes. Now I do - and now understand how we could have done this quite differently along the way.

Thanks to everyone who provided feedback and support over the years.

P-Charge-Info is finally registered in that IANA registry! :-)


Celebrating 10 Years of Blogging at Disruptive Telephony

DisTel Dec2006

Ten years ago today, on December 18, 2006, I launched this blog with a very short 1-paragraph post:

Welcome to Disruptive Telephony! For a number of years, I have been blogging about VoIP as part of my personal blog, "Blog.DanYork.com". However, I'm now in the process of splitting out some parts of my writing into separate blogs. This is one of those blogs. Right now... I'm just setting it up, so don't expect to see much here. Stay tuned, though... much will be happening soon.

At the time, I was living in Burlington, Vermont, and working remotely for the Office of the CTO at Mitel Networks back up in Ottawa, Ontario (where we lived from 2000-2005). Dave Edwards, a friend from Ottawa, left the only comment on that post.

In 2006, the "VoIP blogging" world was quite small - and we all pretty much knew other. Om Malik was writing on his own site (it was yet to become GigaOm). Andy Abramson had VoIPWatch. Jeff Pulver was writing on his sites. Tom Keating at his "VoIP and Gadgets blog" on TMC. Martin Geddes had his "Telepocalypse" site. Alec Saunders had "Saunderslog". And there were a few others...

This was back in the day when we read each others blog posts, commented on them, excerpted each other's posts, etc. And "social media" was not yet a big thing.

It's been a crazy 10 years since... being "restructured" out of a role at Mitel in 2007 after their merger with Inter-Tel, finding a role with Voxeo through this Disruptive Telephony blog (they read this post about telephony not mattering, and then my post about the role I was seeking)... moving to Keene, NH, in 2008... joining the Internet Society in September 2011... it's been a wild ride!

Along the way, I wrote a ton of articles about Skype, SIP, Google and many other VoIP technologies.  MANY relating to security. At one point I seemed to have become Skype's receptionist since no one could find a phone number on Skype's (pre-Microsoft) web site. I wrote about startups that showed great promise, and also about when those promises faded. Many articles on many different topics...

I learned a huge amount and met many great people and made great connections from the writing on this site.

Over this decade of writing, TypePad gives me these stats:

  • 1,209,851 Lifetime Pageviews
  • 331.10 Pageviews/Day
  • 800 Total Posts (including this post)
  • 924 Total Comments

Very appropriately - and with no plan whatsoever - this is the 800th post on this site.

I started using Google Analytics on the site in October 2007 and it tells me I've had 1,817,045 pageviews since time, proving, once again, how difficult it is to track viewers, since the stats are different. More interestingly, GA shows me the top posts that have attracted interest over the years:


1. Google Voice Now Offers SIP Addresses For Calling Directly Over IP (March 2011)

2. Understanding Today's Skype Outage: Explaining Supernodes (December 2010)

3. Did Google Hang Up On Calling Google Voice Via SIP? (March 2011)

4. Will iOS 9 Make My iPad2 Usable Again? (June 2015)

5. What is an Over-The-Top (OTT) Application or Service? - A Brief Explanation (July 2012)

6. How To Set A Skype Chat So That New Arrivals See (Some) Chat History (March 2011)

7. You Can Now Call Into Google+ From Regular Phones - Google Connects Google Voice To Hangouts (May 2013)

8. UPDATE: Will iOS 9 Make My iPad2 Usable Again? (Reports after the upgrade.) (September 2015)

9. Why Is Skype Forcing A Software Upgrade On All Of Us? (Plus The Community Outrage) (August 2014)

10. Did Google REALLY Kill Off All XMPP/Jabber Support In Google+ Hangouts? It Still Seems To Partially Work (May 2013) 


 No real surprises there... my post about Google Voice and SIP addresses STILL receives a significant volume of interest, even though that capability died long ago. For a while, back in maybe 2009-2012, I was one of the main people writing about Skype, and so many of my posts of that era were highly viewed.

A few of my own favorite posts that aren't on that list include:


A. The Directory Dilemma - Why Facebook, Google and Skype May Win the Mobile App War (June 2015 and December 2014) - one of my longer pieces diving into what I see as the prime challenge for new entrants into VoIP / messaging. (The link is to the updated version on CircleID, but the original version was here on this site.)

B. Why The Opus Codec Matters - Even If You Don't Care About Audio (July 2013) - my thoughts on why people need to care about audio codecs.

C. Moving Beyond Telephone Numbers - The Need For A Secure, Ubiquitous Application-Layer Identifier (May 2013) - After SIPNOC 2013, I dove into the whole area around "What do we use as an application-layer identifier for Internet-connected devices?"

D. A Brief Primer on the Tech Behind Skype, P2PSIP and P2P Networks (November 2010) - I kept needing to explain peer-to-peer (P2P) networks to people, and Skype's setup in particular, that I felt compelled to do a deep dive and explain how P2P systems worked. Fun to write!

E. Hypervoice - The Fundamental Flaw In The Proposal (October 2012) - this piece analyzing a proposal from Martin Geddes and the ensuing comment trail make for good reading about different viewpoints on the future of telecommunications.

F. Ch-changes - Taking A New Job At The Internet Society To Join The Fight For The Open Internet (September 2011) - this one is of course a favorite as it explains why I am doing what I am doing now with the Internet Society.

There were many other favorites, like my rant about WebRTC and who we were building it for, but these were the main ones.


Of course, if you look at both of those lists you can notice that with the exception of the two iPad / iOS9 posts and the updated "Directory Dilemma" in 2015, all of these are older posts.

This shows, though, the decline I've had in posting here.  Look at these numbers:

  • 2016 - only 7 blog posts (including this one)
  • 2015 - 25 blog posts
  • 2014 - 28 blog posts
  • 2013 - 30 blog posts
  • 2012 - 40 blog posts
  • 2011 - 154 blog posts
  • 2010 - 90 blog posts
  • 2009 - 52 blog posts
  • 2008 - 110 blog posts
  • 2007 - 234 blog posts

Clearly my velocity has decreased in a serious way, mostly as a result of new responsiblities with the Internet Society and a decreased amount of time for writing here.

I have a loooooonnnnnnnggggg queue of articles I want to write here. The reality is that while some things have changed over the 10 years, many of the same issues are still here.

We'll see where I go with this in 2017. I have a great amount of focus I'd like to give to messaging... let's see if I can make that happen!

Meanwhile... today I will just say THANK YOU TO ALL THE READERS OVER THE 10 YEARS! I'm glad to have helped many people along the way - and I'm glad to have been challenged by many people as well.

I'm looking forward to the next 10 years of writing here... because one thing is definitely for certain: telephony will continue to be disrupted!


Facebook Messenger Launches Group Conference Calls (Audio-only)

Continuing their efforts to be THE communication platform you use, the Messenger team at Facebook rolled out "group calling" this week within the Messenger app on iOS and Android. The new feature was announced by David Marcus, head of the FB Messenger team. Right now this is audio-only (i.e. not group video) and per media reports is limited to 50 participants.

I had to go to the AppStore and upgrade the Messenger app on my iPhone to the latest version, but once I did, I suddenly had a phone icon in the upper right corner of a group chat:

FB groupcalls 1

Tapping that phone icon brought me to a screen where I could choose which of the group members I wanted to bring into the group call:

FB groupcalls 2

After tapping "Call" in the lower right, Messenger launched the call and gave me feedback about who it was connecting, etc:

FB groupcalls 3

It then connected those who were available and four of us were in a group conference call:

FB groupcalls 4

As you can see in the screen captures, I had the standard buttons to mute my microphone and to activate the speakerphone.

AUDIO QUALITY - The audio quality was quite good. I couldn't find any technical info about what they are doing "under the hood" but one of the folks on the call understood that it was WebRTC-based, which would then imply the use of the excellent Opus audio codec. We experienced a couple of audio hiccups but nothing outside the normal VoIP experience and nothing that really detracted from the call. It certainly sounded like a rich, wideband-audio connection.

We didn't stay on the call for long as I didn't want to take their time (or my own), but exiting the call was simple and brought us right back into the group chat to continue our communication.

MOBILE-ONLY - One concern noted by a couple of folks was that the incoming audio call only rang on their tablet or phone, i.e. the iOS or Android app. It did not ring inside of Facebook in a desktop web browser or in the Messenger.com website.

Beyond that, though, it seemed a very straightforward and positive experience.

Now, Facebook Messenger is not the first to do this, of course. Skype has had group audio and video calls for years. As Venturebeat noted, in March of last year Line launched group calling for up to 200 people and WeChat added group audio and video calls in September.

Still, this is Facebook Messenger, with its 900 million users, providing yet another reason to NOT use traditional audio conferencing solutions.

I would suspect, too, that video conferencing can't be too far off, either, given that Facebook Messenger currently does let you do 1:1 video calls - and also that competitors offer group video calls.

It continues to be an absolutely fascinating time to watch the severe disruption of traditional telecommunications... and this move by Facebook is yet another example of how the ways we are communicating are changing.

What do you think? Will you use the group calling within Facebook Messenger?


Audio Recording: My SIPNOC 2014 Talk - "Is It Time For TLS For SIP?"

Is it time to use Transport Layer Security (TLS... essentially what we used to call "SSL") to add a layer of trust and security to Voice-over-IP (VoIP) that uses the Session Initiation Protocol (SIP)?

Way back in June 2014, I gave a talk on this topic at the SIP Network Operators Conference (SIPNOC) in Herndon, Virginia. I recorded the audio of the session... but then lost track of the recording. I recently found it and, since much of it is (sadly) still relevant, I decided to release the recording as one of my The Dan York Report audio podcast episodes:

The slides that go with the presentation are available on SlideShare:

You'll see in the slide deck that I also provide some tutorials around DANE and DNSSEC along the way.

Coincidentally, I learned on Facebook over the weekend that my friend Olle Johansson was speaking on this exact topic at the FOSDEM 2016 conference in Brussels this weekend. His slides about SIP & TLS are also available on SlideShare, and he has more recent information - and also the conclusion that we need to use "SIP Outbound" for any of this to work:

Olle's last slide about what we need to do hits on the key points - and I agree with his conclusions.

Let's look at how we can get more TLS used within SIP to bring about a more secure and trusted VoIP infrastructure!


Talko's Purchase By Microsoft Shows The Challenge Of The Directory Dilemma

Today Microsoft announced that they acquired the technology of Talko, a communication app created by Ray Ozzie and launched back in September 2014. Fortune has an article on the acquistion, as do a good number of other media sites.

After Talko first launched, I wrote about my initial experience - and the problem I had of Talko working through my home firewall. But I was intrigued by the possibilities laid out in a Medium article about how Talko could change communication and integrate voice, chat and messaging in interesting ways.

The reality, though, was that Talko was a classic case of suffering from the Directory Dilemma - as I said in that article:

People will only USE a communication application if the people they want to talk to are using the application.

And that was true for me... I tried out Talko, as I try out many apps. I used it for a while. And then... I stopped.

The people with whom I communicate were not regularly using Talko.

You can see the recognition of this dilemma in today's front page of Talko's web site:

However, as engaged as many of you have been, the reality is that the broad-based success of communications apps tends to be binary: A small number of apps earn and achieve great viral growth, while most fall into some stable niche.

For all the value and enjoyment it's delivered, and for all the team's listening and perseverance, Talko was largely on the path to filling a (passionate) niche. We're in this to have great impact, so it's time for a change.

and:

We deeply appreciate the commitment that so many of you made in betting on Talko. You invested your time and your reputation to convince your friends and co-workers to use the product with you.

This is the reality that messaging / communication apps have to face today. Either somehow build that massive directory - or be happy (and financially stable) within the certain niches and communities in which your product can thrive.

What's next for the Talko team (minus Ray Ozzie, who has said he will not be re-joining Microsoft) isn't 100% clear. Both the Microsoft and Talko posts today are vague, with the latter saying:

As part of the Skype team, we'll leverage Talko’s technology and the many things we’ve learned during its design and development. We'll strive to deliver the best of our product’s innovations far more broadly than on our current path.

and:

Looking forward, we hope to hear from you again as we find ways to deliver the best of Talko in Skype.

We'll have to see what pieces of Talko they bring into Skype.

Congrats to Ray Ozzie and the Talko team - and to Microsoft - on this acquisition. I hope it does work well for all involved.

Meanwhile, we can look and wonder which of the zillion new messaging apps out there will be the next to fold into a larger player...

P.S. There's a thread on Hacker News about today's announcement and there was a really long thread on HN back in 2014 when Talko was announced that may still be of interest.


Video and Slides Now Available For My AstriCon 2015 Keynote: Open Source and The Global Disruption of Telecom

If you're interested in what I said last month at AstriCon 2015 in my keynote on "Open Source And The Global Disruption of Telecom: What Choices Will We Make?", the video and slides are both available.

As I wrote about previously, the context for this discussion was to talk about the changes that are happening all around us in terms of the ways in which we communicate. Here was the abstract:

There is a battle raging for the global future of telecommunications and the Internet. Taking place in networks, board rooms and legislatures, the battle will determine how we all communicate and what opportunities will exist. Will telecom support innovation? Will it be accessible to all? Will it give us the level of security and privacy we need to have the open, trusted Internet? Or will it be restricted and limited by corporate or government gatekeepers?

The rise of voice-over-IP has fundamentally disrupted the massive global telecommunications industry, infrastructure and policies. Open source software such as Asterisk has been a huge driver of that disruption and innovation.. but now what? What role do platforms such as Asterisk play in this space? And what can be their role in a telecom infrastructure that is now mobile, increasingly embedded (Internet of Things) and more and more using proprietary walled gardens of communication?

How well I delivered on that will be up to you to decide... but I felt good about how it all came out and received many great comments and feedback throughout the rest of the event and afterwards. And, as a speaker I could see from the crowd (about 500-ish people) that they were NOT looking down into their smartphones or laptops... which is always a good sign! ;-)

A key point of what I aimed to do was to bring people up to a higher level to think about how their own actions fit into the broader context of what is happening in the world today.

It was fun to do! And I loved all the questions I was getting after that. My goal was to make people think... and it seemed that at least for some I did.

My part of the video starts after 15 minutes of introductory items (this was the opening of the event), so if you watch in the embedded video below you'll need to move forward to the 15:00 mark. You can also follow this direct link to the start of my segment with an introduction to me from Mark Spencer, the creator of Asterisk.

(And yes, this was the first time I had ever given a presentation wearing a ponytail in the long hair experiment I've been trying this year... I'm still not 100% sure I'm going to keep this style. This may be the first and only presentation you see with me like this.)

Unfortunately, the video only shows me talking on stage and doesn't show the slides I was using... so you don't understand what I'm talking about when I reference the slides.

I've posted the slides to my SlideShare account but as you'll see without the video or audio they aren't of much value. This was a wonderful opportunity for me to present in the very minimalist style I prefer where I only use images or a few words - and I thoroughly enjoyed doing so.

However, syncing the slides to the video is not something you'll probably find easy. At some point perhaps I'll create another video showing both my speaking and the slides... but I don't know that it will happen anytime soon.

Meanwhile, here they are...

Some of the links I reference in the presentation include (in the order of their appearance):

If you enjoyed this presentation and would like to have me potentially speak at your event, please do contact me. I've been speaking for many years and very much enjoy giving these kind of presentations at all types of events.


Keynote at AstriCon on Oct 14: Open Source And The Global Disruption Of Telecom - What Choices Will We Make?

Astricon danyork 660px

Two weeks from today I'll be in Orlando giving the opening keynote address at AstriCon 2015. The abstract of the session is:

Open Source And The Global Disruption Of Telecom - What Choices Will We Make?

Wednesday, October 14th, 2015 - 9:00 am to 9:45 am - Pacifica Ballroom 7

There is a battle raging for the global future of telecommunications and the Internet. Taking place in networks, board rooms and legislatures, the battle will determine how we all communicate and what opportunities will exist. Will telecom support innovation? Will it be accessible to all? Will it give us the level of security and privacy we need to have the open, trusted Internet? Or will it be restricted and limited by corporate or government gatekeepers?

The rise of voice-over-IP has fundamentally disrupted the massive global telecommunications industry, infrastructure and policies. Open source software such as Asterisk has been a huge driver of that disruption and innovation.. but now what? What role do platforms such as Asterisk play in this space? And what can be their role in a telecom infrastructure that is now mobile, increasingly embedded (Internet of Things) and more and more using proprietary walled gardens of communication?

Join the Internet Society's Dan York in an exploration of what the future holds for telecom infrastructure and policy - and how the choices we make will determine that future.

Sounds great, eh?

Now I just have to deliver on that lofty rhetoric! :-)

Seriously, though, I'm very much looking forward to giving this presentation and I'm delighted that the folks at Digium asked me to speak. We're at a critical time in the evolution of our global communications infrastructure... with everything moving to IP and also moving to mobile, there are incredibly important choices we have to make for our future.

In the talk, I'll be speaking about the scenarios we have for what our future Internet could look like. I'll be talking about the role of open source. I'll be challenging the audience with some questions to ponder. I'll touch on some of the incredibly important - yet hard to understand - global policy issues such as the upcoming WSIS+10 Review in December - and why an open source developer should even remotely care! I'll of course hit on security issues and the rise of mobile... and more...

I'm excited!

I'm also excited to finally attend an AstriCon event. I used to write about Asterisk a good bit and for a while was running my own server in my home office for VoIP... but in all that time I never was able to work in attending an AstriCon!

If you are going to be there in Orlando, please do say hello! (There's still time to register!)

P.S. And yes, Olle Johansson, I'll be sure to work in at least one reference to IPv6! And TLS, too! Don't worry! :-)


Firechat Enables Private Off-The-Internet (P2P) Messaging Using Mobile Phones

Firechat mesh network

There was a fascinating article posted on Medium this week by the CTO of messaging app Firechat:

In the text he outlines how they do decentralized "off-the-grid" private messaging using an ad hoc mesh network established between users of the Firechat app. It sounds like the app instances join together into some kind of peer-to-peer (P2P) network and then do normal "store-and-forward" messaging.

Of note, the apps do NOT need an Internet connection, or even a cellular network connection - instead they can use the Bluetooth and WiFi radios in the mobile phones to create a private mesh network and connect to other users of the Firechat app.

Naturally, having spent some time exploring P2P networks back when I was playing around with P2P SIP and distributed hash tables (DHTs) and other technologies, I immediately jump into the techie questions:

  • How are they routing messages from one user to another?
  • How is the "directory" of users in P2P mesh maintained?
  • What addresses are they using for the communication? Is this still happening over IP addresses? Or are they using some other kind of addressing?
  • How do users join and leave the mesh network?
  • How do user get authorized to join the private mesh? (Or is it just open to all?)
  • How secure is the communication between the parties?
  • Is the message encrypted or private in any way? Or is it just plain text?
  • How well do smartphone batteries hold up if multiple radios are being used? What is the power impact of joining into a mesh network like this?

None of that is covered in this article, of course... this piece is more about the theory of how this can work given a particular density of users. It introduces the phrase "percolation threshold" and provides some background and research into how these kind of networks can be created.

I've always been fascinated by P2P networks like this sounds to be. The beauty of the Internet... the "Internet Way", so to speak... has been to support distributed and decentralized architectures.

If you think about mail or web servers, they are (or at least were) massively distributed. Anyone could set up a mail or web server - and millions upon millions of them bloomed. While we've certainly seen a great amount of centralization due to market dominance (ex. Gmail), the architecture still is distributed / decentralized.

Except... of course, the directory is still centralized. Mail and web servers rely on the central directory of DNS to resolve domain names into IP addresses so that connections can occur. Most other applications rely on DNS for this as well.

Hence my curiousity about how Firechat is handling the directory and routing issues.

I'm also intrigued by how the article hints at integrating Internet-connected users into the P2P mesh. So you really have a hybrid network that is part P2P and part connected out to cloud-based servers.

(And all of this brings me back to those early days of Skype 8-10 years ago when so many of us were captivated by the P2P mechanisms they created... most all of which is now gone in the post-Microsoft-acquisition as Skype has moved from P2P to server/cloud-based - with one big reason being given that mobile devices apparently had speed and battery life issues participating in true P2P networks.)

A key challenge Firechat faces, of course, is the "directory dilemma" of building up the quantity of users where P2P mesh networks like this can happen. This is the same dilemma facing basically all over-the-top (OTT) messaging apps. "Percolation theory" requires a certain user density for a mesh like this to work.

That will be their struggle.

And in some urban areas I can see this working quite well. Perhaps not so much out in the woods of New Hampshire where I live!

But I wish them well with this. I love to see new explorations of potential new architectures for communication. And I can certainly see instances when ad hoc, distributed/decentralized P2P meshes like these could be quite useful.

And I'm definitely looking forward to some more technical articles that dive down into some of these questions.... I do hope they'll write more soon!


Photo credit: Stanislav Shalunov's article about Firechat


WebRTCHacks Publishes Analysis of Facebook and WhatsApp Usage of WebRTC

WebrtchacksThe team over at webrtcH4cKS (aka "WebRTCHacks") have been publishing some great articles about WebRTC for a while now, and I thought I'd point to two in particular worth a read. Philipp Hancke has started a series of posts examining how different VoIP services are using WebRTC and he's started out exploring two of the biggest, Facebook and WhatsApp, in these posts:

Those articles are summaries explaining the findings, with much-longer detailed reports also available for download:

Both of these walk through the packet captures and provide a narrative around what is being seen in the discovery process.

A common finding between both reports is that the services are not using the more secure mechanism of DTLS for key exchange to set up encrypted voice channels. Instead they are using the older SDES mechanism that has a number of challenges, but, as noted by the report, is typically faster in enabling a call setup.

All in all the reports make for interesting reading. It's great to see both Facebook and WhatsApp using WebRTC and I think this will only continue to help with the overall growth of WebRTC as a platform. As an audio guy, I was pleased to see that Facebook Messenger is using the Opus codec, which is of course the preferred codec for WebRTC... but that doesn't necessarily mean that it has to be implemented by companies using WebRTC within their own closed products. Kudos to the Facebook team for supporting Opus!

Thanks to Philipp Hancke for writing these reports and I look forward to reading more in the series!