News from Industry

WebRTC vs Zoom? WebRTC is actually quite good. But you knew that already – didn’t you?

They say quality is in the eye of the beholder. So behold.

We’ve all been told once and again that this video conferencing vendor or that video conferencing vendor work great. They offer the best quality. The best experience. They work in conditions that others don’t.

I even had a call once with an entrepreneur that explained to me how he is going to offer a service that is better in its 1:1 video quality than Skype and Google Hangouts. And he is going to do it with WebRTC. I spent the better part of that call to get him off that idea (something about his logic was off there).

But I am digressing.

As many others, I’ve been told time and again how Zoom is great. How in spite of the fact that it doesn’t work in the browser and forces you to download its client (some even refer to it as a virus), it gets traction and adoption. It feels like it is the best game in town. And then they mention the reasons:

1. It’s free (until it isn’t, which is a great business model if you can make it work, and Zoom is making it work)
2. It has better video quality than the competition. Especially WebRTC

I am not the only one who needs to listen to it, and even believe it to some extent. The guys at Jitsi got curious – why not put it to the test?

So they took a Mac device, placed it on a WiFi network, added a network limiter so they can fiddle with the network configuration, and did a 1:1 call. Once with Zoom. And once with WebRTC.

Idea is this – start with as much bandwidth as the video call wants. Then limit it to 500kbps. Check how much time it takes to adapt. Remove the limit and change how much time it takes it to adapt back. More about it in Jitsi’s blog.

Essentially – testing for this network conditions:

The longer that marked areas, the worse the experience is going to be for the users.

And guess what? Zoom faired worse than WebRTC. Not a little, but a lot worse.

Full adaptation to limiting the bandwidth took WebRTC 20 seconds. It took Zoom 156 seconds (!).

Ramp up back to 2mbps took WebRTC 32 seconds. It took Zoom 62 seconds.

Now here’s my analysis of this.

WebRTC Rocks

Yap. it really does.

The screen capture from that Zoom blog post that was pasted by Jitsi?

Stating that “web-RTC is a very limited solution that would not allow us to provide all the excellent features that our users have come to expect from us”?

That’s from 2015.

A lot have been improved in WebRTC since then, if that explanation was even correct in 2015 to begin with.

Without the need for most of us to do anything, we’re getting updates to a top notch media engine in the form of WebRTC inside the browsers we use. The code used in Chrome are open sourced, so they are accessible to all to embed it in their own applications as well.

Security fixes? New codecs? Improved media algorithms? They just “happen”. Out of thin air. For most of us.

Defending Zoom

If I look at it from Zoom’s point of view, besides the fact of being a dominant player in the market with or without WebRTC, here’s the challenges with such a test scenario:

• It was done once, or a few times. But it is still only one scenario
• It wasn’t a real life scenario. Just something concocted for this. Jitsi could have rigged it and tweaked it so that WebRTC would shine, but in real life, that doesn’t happen, and at Zoom we’re optimizing for real life scenarios
  • (that isn’t really so. From my experience and knowledge of the Jitsi team, I’d estimate they tried to be VERY careful here to not fall into that trap)
  • (and what’s real life scenarios anyway?)
• The network limiter used changes behavior in ways that aren’t close enough to reality
  • (that I can understand and live with. We see faster uptake of the same type of scenarios for WebRTC at testRTC – more on that later)
• Zoom might be working through external remote servers for that same session while WebRTC is going peer to peer on the local network. Servers behave differently than clients, so the results seem somewhat “off”
• In other scenarios, Zoom might actually be better than WebRTC

Which leads us to the fact that more tests are needed to know which one is best and in which scenarios.

This starts to sound like the VP8 vs H.264 quality comparisons of the past (I never could tell the difference).

It’s the Infrastructure Stupid

With WebRTC, it all boils down to the infrastructure. The one with the better deployment wins the quality game.

1. Do you peer to peer for 1:1 sessions and seamlessly switch to SFU architecture when more participants join?
2. Where are your media servers located?
3. Do you cascade the session across media servers to improve quality?
4. Do you provide feedback to the user about the network conditions?
5. Do you switch video off when there’s not enough bandwidth?
6. How are you managing things like FEC, simulcast, SVC, … ?
7. What about mobile and native app support?

And the list goes on.

With vendors who use proprietary codecs and transport protocols, this is doubly so, as they need to cater for the browser once they reach WebRTC. So while their native apps might be optimized, it might all go down the drain once they transcode or just “translate” to reach the browser using WebRTC.

Need to understand WebRTC and how to design and architect real world solutions with it? A first step is to understand the servers used to connect WebRTC.

Join a free video course on WebRTC servers

Which brings us to why someone like Zoom should use WebRTC and thing about the quality issues once connecting to it:

You Need WebRTC

Zoom already supports WebRTC. I just found out when I searched for stuff to write this article: there’s a Zoom Web Client

It runs on Chrome and enables using audio in Chrome when joining meetings. No video, probably because transcoding the proprietary video codec Zoom uses to the ones in WebRTC is too complicated, but using G.711 or Opus in the browser and transcoding or using the same in Zoom is way simpler.

Zoom is going through the same phases that Amazon did with Chime:

• Amazon Chime started with a downloadable client
• They then added limited browser support that enabled users to view the screen shared in the browser and connect via the phone without the need to download the client
• Later on, audio support was added to the web client
• And recently, video got supported
• Screen sharing and remote desktop control still doesn’t work. I’d say it is a matter of time

This exact same path has been happening to other vendors in one way or another.

Why not Check Your Own Service?

While writing this article, it dawned on me, that this is one of these scenarios that is ridiculously easy to simulate using testRTC, so I went ahead and created a script that does just that:

• Loads up Jitsi with 2 participants. That should cause them to work peer-to-peer
• Run the call for 1 minute unhindered
• Limit bitrate to 500kbps and run for 2 more minutes
• Remove bitrate limit and run for 2 more minutes

Here’s how the main part of the script looks like:

   // Wait for 1 minute client    .pause(60*sec)    .rtcScreenshot('ALL GOOD');    if (probeType === 1) {    client        .rtcEvent('Start limit', 'global')        .rtcSetNetworkProfile('custom', 'bandwidth', 500000, 'both', 'both')    }    // 2 minutes with bandwidth limits client    .pause(60*sec)    .rtcScreenshot('LIMITED')    .pause(60*sec);    if (probeType === 1) {     client        .rtcSetNetworkProfile('') // back to pristine network conditions        .rtcEvent('Stop limit', 'global');    } client    // 2 more minutes unlimited    .pause(60*sec)    .rtcScreenshot('BACK TO NORMAL')    .pause(60*sec);

 

The .rtcEvent() calls are there to place a vertical lines on the graphs while the .rtcSetNetworkProfile() is there to fiddle around with the network conditions.

There were two probes here, each one a participant in the call. The first one is the one I limited while the second one was left “untouched”.

Here’s what the graphs look like on the second probe:

The above graph shows the outgoing birate. Within a span of 5 seconds, WebRTC finds out the new effective bitrate and adapts to it. Ramping back up takes some 20 seconds.

The above graph shows the incoming frame rate. You can see how frame rate reporting in WebRTC takes a bit of time to get back to its usual self – also some 20 seconds or so.

I wanted to check how the Jitsi SFU would behave, so I tweaked the test URL for that. The results? Still better than the Zoom one. 20 seconds to hit 30 frames per second and around 50 seconds to get back to full bitrate.

If you want to try it yourself, just import the JSON file in this Google Drive folder to your testRTC account and modify it to fit your needs.

Where to now?

WebRTC is more than good enough.

Making it better is usually about thinking your way through the best possible architecture, along with media servers that take care of network conditions properly.

As for Zoom… please make sure your next call with me is on something that has WebRTC. The machine I regularly use for call is Linux. Zoom doesn’t work there… it doesn’t really support Chrome or Linux. Yet.

The post WebRTC vs Zoom. Who has Better Video Quality? appeared first on BlogGeek.me.

Here we mark the freezing of development for Kamailio v5.2 series.From now on, for few weeks, no new new features will be pushed in the master branch. Development can still be done, but should be hold in developers’ GIT personal branches.Once the branch 5.2 is created (expected to happen in 3-4 weeks), the master branch becomes again open for new feature. Meanwhile the focus has to be on testing current code.Work on related tools (e.g., kamctl) or documentation can still be done as well as getting the new modules in 5.2 in good shape, plus adding exports to KEMI interface (which should not interfere with old code).The entire testing phase is expected to be 4 to 6 weeks, then the release of v5.2.0 will be out.Thanks for flying Kamailio!

Kamailio SIP Server v5.1.6 stable is out – a minor release including fixes in code and documentation since v5.1.5. The configuration file and database schema compatibility is preserved, which means you don’t have to change anything to update.Kamailio® v5.1.6 is based on the latest source code of GIT branch 5.1 and it represents the latest stable version. We recommend those running previous 5.1.x or older versions to upgrade. There is no change that has to be done to configuration file or database structure comparing with the previous releases of the v5.1 branch.Resources for Kamailio version 5.1.6Source tarballs are available at:

• https://www.kamailio.org/pub/kamailio/5.1.6/src/

Detailed changelog:

• https://www.kamailio.org/pub/kamailio/5.1.6/ChangeLog

Download via GIT: # git clone https://github.com/kamailio/kamailio kamailio
# cd kamailio
# git checkout -b 5.1 origin/5.1Relevant notes, binaries and packages will be uploaded at:

• https://www.kamailio.org/pub/kamailio/5.1.6/

Modules’ documentation:

• https://www.kamailio.org/docs/modules/5.1.x/

What is new in 5.1.x release series is summarized in the announcement of v5.1.0:

• https://www.kamailio.org/w/kamailio-v5-1-0-release-notes/

Thanks for flying Kamailio!

By david drexler – Flickr, CC BY 2.0, Link

 

Back in August, Reuters reported on a “secret legal fight” between the FBI and Facebook about wiretapping Messenger calls. The Verge as they found our old post about reverse-engineering Messenger from 2015 and had a number of follow-up questions on it for a Messenger wiretapping article they ran. Technical details on the case are quite hard to find so I was not able to dig deeper into the specifics around wiretapping.

Continue reading Messenger was not forced to wiretap but… at webrtcHacks.

Astricon 2018, the Asterisk users’ conference, is just several days away. It takes place again in Orlando, FL, USA, during October 9-11, 2018.Kamailio is very well represented at this edition, besides having a stand in the expo floor, there will be presentations by Alex Balashov, Daniel-Constantin Mierla and Fred Posner.You can find the schedule and more details about the event at:

• https://www.asterisk.org/community/astricon-user-conference

We hope to see many of you in Orlando!Thanks for flying Kamailio!

Kamailio SIP Server project is organizing a meeting of its developers during September 27-28, 2018, hosted by sipgate.de in Dusseldorf, Germany.The event is intended to facilitate the interaction between Kamailio developers and to offer a convenient environment for working together on several topics of high interest for the project, including writing code for Kamailio and its tools, improving documentation, or discuss about future development.Everyone from the community is welcome to join. Please note we have a limited capacity of 20 seats in the meeting room. Also, very important to be aware that this is not an event to learn how to use Kamailio.More details about the event, the venue, how to register, are available at:

• https://www.kamailio.org/w/developers-meeting/

Looking forward to those two intensive hacking Kamailio days in Dusseldorf!Thanks for flying Kamailio!

As a result of collaborative work at Kamailio Developers Meeting, we succeeded to merge the two existing module exports interface (one for Kamailio modules and the other one from SER modules) in a single one.All public modules were updated, but if you have any private module then you have to update as well in order to get it compiled with the latest master. Just look at one of the modules (e.g., sl module is a good option) and all the fields in mod exports structure have comments with their meaning.In short: we removed unused fields for statistics, mi commands and extra-processes (from Kamailio old interface) and oncancel (from SER old interface), kept the RPC exports from SER interface and PV exports and dlopen flags from kamailio interface. The other fields were common in both interfaces, but be aware that order was also changed.If you have a module implementing the SER old interface, you also need to add the free fixup field in functions exports structure — you can just set it to 0.With this we have now a single mod exports interface in all modules, the core is also cleaner as we got rid of the various compatibility layers.Thanks for flying Kamailio!

There will be some maintenance work done to deb.kamailio.org server these days in order to move to a better infrastructure, so it may not be available for a while. The DNS needs to be updated as well, this also can affect you if your router/provider does caching.This is affecting the APT repository for Debian and Ubuntu packages.Update Sep 28: migration has been completed, if you encounter any issue with the apt repositories, contact us.Thanks for flying Kamailio!

The autumn of 2018 has plenty of events where Kamailio developers and community members will participate. Among them:

• Kamailio Developers Event, during September 27-28, in Dusseldorf, Germany
• Kamailio On Tap – Free Community Meetup on the evening (19:00) of the 27th of September, in Dusseldorf, Germany
• Astricon 2018, during October 9-11, in Orlando, USA
• TADHack Global, during October 13-14, Worldwide
• #Berlin5GWeek, during November 12-16, in Berlin, Germany
• Kamailio Advanced Training, during November 12-14, in Berlin, Germany
• TAD Summit, during November 13-14, in Lisbon, Portugal
• VoIP2Day, during November 14-15, in Madrid, Spain
• FUSECO Forum, during November 15-16, in Berlin, Germany

If you attend any of these events or are around their location at their dates, get in contact with Kamailio’s community via sr-users mailing list and let’s meet, greet and discuss latest news about the project and RTC market.Enjoy the autumn!Thanks for flying Kamailio!

About 15 Kamailio developers are traveling to Dusseldorf for the Kamailio Developers Meeting during September 27-28, 2018, giving the opportunity to organize an open social networking event in the evening of September 27, 2018, at 19:00!That’s “Kamailio On Tap“, an informal meeting of Kamailio developers and VoIP community/RTC industry, to be hosted at:Pub: Hafenquelle (front of the street)
Gladbacherstrasse 74
40219 DusseldorfIt is not an event only for Kamailio developers, anyone from the VoIP community/RTC industry that wants to join us is welcome! It is a free to attend event, you just need to register in order to be sure we do not exceed available seats and dimension properly what is offered during the event.To register, send an email to  with the subject: “Kamailio On Tap” until September 26, 2018. Feel free to pass these details to friends, colleagues or business connections that you have in the area of Dusseldorf and you think they may be interested to attend.Beer and pizza are sponsored for everybody!Looking forward to meeting many of you in Dusseldorf!Thanks for flying Kamailio!

An updated WebRTC FAQ for those who wish to understand this tech somewhat better.

It is 2018, and it seems like there’s no good FAQ for WebRTC. Nowhere. They’re just not up to date. That, coupled with my own need to be the best source of information on the web about WebRTC (and the fact that my last few articles were more about CPaaS and messaging than WebRTC), got me to write this one.

What is WebRTC?

WebRTC is both a standard specification and an open source project.

WebRTC allows sending and receiving of real time voice, video and arbitrary data across browsers and other devices. This means we now have an easy way as users to conduct voice and video conferences from a browser or from our mobile devices. WebRTC can do a lot more than that, but voice and video in real time is the basis of what you get out of it.

There’s a short video explaining What is WebRTC on my site.

Who is behind WebRTC?

WebRTC originated from Google. It started by an acquisition of a few companies, whose technology was then repackaged and released as open source under the name of WebRTC.

Google is still the main vendor behind WebRTC. That’s because its own WebRTC engine is the main WebRTC open source project out there and it is also the one that gets integrated into the Chrome browser.

Mozilla, Microsoft and Apple all contribute to WebRTC and have their own implementations of WebRTC in their browsers (some of these implementations are derived from the Google code).

Other vendors and individuals contribute to the specification through the IETF and W3C, where the standardization process of WebRTC takes place.

My own contribution to WebRTC is this site, which publishes a lot of free information around WebRTC as well as the Kranky Geek event, WebRTC Index and WebRTC Glossary.

Is WebRTC ready for commercial use?

Yes.

WebRTC is used today by commercial services (here are 10 such examples).

Some complain and gripe that WebRTC isn’t ready for commercial use. This stems due to the many changes that the codebase and specification is undergoing. It also means that if you plan on using WebRTC, either do that through a third party managed service (a CPaaS vendor – list here) or make sure to have a team of savvy developers that can keep up with the pace.

The changes introduced to the WebRTC codebase itself oftentimes breaks backward compatibility and features, probably by sticking to a “move fast and break things” motto to some extent.

Why should I use WebRTC?

If you don’t need real time voice and video then you might not need to use WebRTC at all.

If you do, then it is a matter of capability, resources and time to market:

• If you want your service to work inside a web browser, then WebRTC is your only way of getting real time voice and video into a browser
• If you want it elsewhere, then in almost all cases, using WebRTC will cost you less and get you there faster than the alternatives

What codecs are used in WebRTC?

For voice, the mandatory codecs are G.711 and Opus. Out of these two, be sure to use Opus (G.711 is old and crappy).

For video, the mandatory codecs are VP8 and H.264. Apple’s Safari browser doesn’t support VP8. And on Android, Chrome won’t support H.264 on *some* devices (I’ll let you go figure out on which ones). More about that in this video mini-series.

VP9 is supported by Chrome and Firefox. AV1 seems to be the future.

What browsers support WebRTC?

All of them. Almost. But not exactly. And there are differences.

• Chrome is where most developers focus. It isn’t 100% aligned with the specification yet (none of the browsers are)
• Firefox is the next that gets focus from developers. Close enough to Chrome in its implementation
• Edge doesn’t support data channels. And many skip it when it comes to testing due to is low market adoption
• Safari is what everyone wants (Apple you know), but it is still buggy and doesn’t have support for VP8. Most need Safari support for iOS but are fine with not supporting Safari on Mac. Read this webrtcHacks post for more

There’s a devices cheat sheet on my website.

And then there’s adapter.js which you should definitely use.

Can I use WebRTC on mobile devices?

Yes.

On Android, on official Chrome and Firefox browsers, WebRTC is available.

On iOS, Safari offers something usable if you are willing to invest the energy to get it working well.

On both Android and iOS you can take the WebRTC source code and integrate it inside your native application. Google even releases prebuilt packages for both Android and iOS.

If you want to use a Webview inside your app, then this is easy with Android, restrictive with iOS for now (you won’t be able to access the camera or the microphone there).

Do I need special servers to run WebRTC?

Yes.

You definitely need a signaling server. And STUN/TURN server. You might need a media server.

WebRTC is said to be peer-to-peer. It is when it comes to the media as much as possible. But developers can make use of it in server centric environments. And there are some scenarios where it makes no technical sense to use peer-to-peer (for example if you want to broadcast something to a million people or conduct a video conference with 20 participants).

There’s a free video mini series explaining WebRTC servers on this site.

Can WebRTC be used to create large conferences?

Yap.

Think of WebRTC as a basic building block that gives you superpowers. With it you have the ability to send and receive voice and video in real time virtually on every device and browser.

Now what you do with this superpower, how you interact with it, architect your solution around it – that’s up to you.

There are vendors offering video conferencing that uses WebRTC and gets to 10’s of participants. Webinars with 100’s of live viewers in the audience.

You can read more about scale and size of WebRTC.

Is WebRTC posing a security threat for me?

No.

And yes.

Depending who you are and what are your needs.

I wrote a lot about WebRTC security in the past. It gets tiring.

WebRTC comes with security in mind. It encrypts everything. Can’t remove that encryption. And browsers get security updates faster than any other software you have.

The one sticking issue is probably the fact that it exposes the local IP address of your machine when it is used. VPNs that are implemented properly solve that as well. More about that over at webrtcHacks and VPN leaks.

What does WebRTC 1.0 mean?

WebRTC 1.0 is the first time that WebRTC will have an official specification.

Up until now, we had drafts and browser implementations that were an approximation of the drafts. Now we have an approximation of the WebRTC 1.0 specification and approximations of implementations to it in browsers.

Confused?

Don’t be. Assume WebRTC is good to go commercially (check that part of my FAQ) and just go read Jan-Ivar’s explanation @ Mozilla’s Advancing WebRTC blog.

Oh – and be sure to use adapter.js.

How much does WebRTC cost?

It doesn’t. And it does.

WebRTC is freely available in browsers.

The source code is also freely available.

The servers you will need to use it – someone will need to pay for them. That payment can be to a managed service, or to a cloud vendors and developers who will develop, install and maintain them. Up to you to decide.

Oftentimes, developers assume everything should be free with WebRTC, whereas reality is different. And for some reason, most perceive development  costs as free or sunk costs (they will call it investment) as opposed to paying a third party for doing the hard stuff for you.

A bit more on this here.

How can I learn more about WebRTC?

If you are into free, then try reading the specs, playing with the official samples, reading this blog and webrtcHacks.

There are a few courses on coursera, pluralsight and elsewhere. Never tried them, but read their agendas. Take a look for yourself and decide what’s for you.

There are books, but none of them is up to date with the specification.

Best place? Hands down? My paid course. Advanced WebRTC Architecture Course

Can I help you?

Maybe.

There’s my course. There’s testRTC where I am a co-founder (we do testing and monitoring of WebRTC apps).

I also consult. Around architecture, vendor selection, defining requirements, setting roadmaps, working on differentiation and doing pure marketing related work. What can I say?

I like the variety.

You can reach out to me here.

–

Got a question about WebRTC that needs to go into this FAQ? Add it below in the comments.

The post WebRTC FAQ: The 2018 Version appeared first on BlogGeek.me.

Social messaging is killing RCS in all the places that matter.

When looking at messaging in the context of communications and people, we can probably split the story into 3 distinct models:

1. Consumer centric
2. Business centric
3. Businesses to consumers (and vice versa)

I’ll quickly sift through the first two and focus on the third.

Consumer Centric

Consumer centric is easy. That’s where Apple iMessage, WhatsApp, Facebook Messenger, Telegram, WeChat and a bunch of others are competing. The approach there today is to deliver a rich messaging experience that includes text, images, video, voice and video calling, location, groups, … – the list goes on. And on. And on.

They have won the war against SMS. We still have SMS. Some mistakenly call it ubiquitous (on my phone it is used for spam and 2FA messages only). They won the war against RCS that never really started.

To give you a clue – Israel is a WhatsApp country. If you don’t have WhatsApp you don’t exist. It is true from the age of 8. I just purchased the first smartphone for my 8 year old boy. Not so he can play or call with the phone – just so he can send messages to his classmates and stay part of the social fabric of his class. It happened to my daughter when she reached that age. I am now a part of multiple WhatsApp groups: family, close friends, parents of my kids’ classes and after classes, work related, etc.

How easy would it be to move people in Israel from entrenched groups that hold history, images and videos? And to what end? How would RCS be any better in its experience?

Business Centric

Business centric is Slack. It used to be all about calling and the PBX. Slack changed the game. Everyone is talking about “team messaging” today. I used the term enterprise messaging years ago.

What Slack did was find a good balance between functionality and user experience that no other player has been able to copy properly so far, but everyone is after.

• Cisco wrote Spark from scratch, then rebranded it as WebEx Teams
• Then there’s Microsoft Teams, which gobbled up Skype for Business
• And there’s Facebook for Workplace
• Google had Google Talk, and Voice, and then Hangouts, and then Meet, and now Google Hangouts Chat
• Did I miss anyone? Maybe: Twist, Flock, Discord, Mattermost, ChatWork, Ryver, Zoho Cliq, RingCentral Glip, Semaphor, Troop Messenger, Redbooth, Flowdock, and the list goes on – each with its own shtick

WhatsApp is unlikely to penetrate businesses in a meaningful way. Facebook built Workplace instead of trying to introduce Facebook or Messenger directly.

Where’s SMS in this orgy of messaging? Meaningful conversations happen in IP messaging services and not over SMS anymore. Some solutions, like VonageFlow offer a seamless experience that encompasses both messaging as we know it today and SMS, though I’d argue that capability is a business to consumer one.

For all intent and purpose, SMS is non-existent when it comes to business centric messaging.

Business to Consumer

Back to RCS. RCS was supposed to be the future of SMS when we all move to IP based packet networks. Guess what? We’re all on IP based packet networks, and RCS isn’t really here yet in any meaningful way.

In the past couple of years, RCS got a new tune by its proponents. The strategy changed from getting consumers back from social networks towards being the one ubiquitous network – the ring to rule them all. Here’s the idea: you get RCS on all smartphones worldwide. Now carriers have the ubiquity they had with SMS. And businesses would pay for such access to customer’s phones.

Not going to happen.

Why? Because Apple and Facebook have other plans for us.

Apple now has Apple Business Chat. It is built into the iPhone, making businesses discoverable and reachable over iMessage from the Safari browser, Spotlight search, Siri assistant and Apple Maps. I’ve written extensively about it when it was introduced on SearchUC: Apple Business Chat looks to polish customer messaging

WhatsApp came out with their own offering called WhatsApp Business API. Similarly to Apple Business Chat, it offers the ability for businesses to communicate with consumers. Apple does that by focusing on contact center vendors while Whatsapp partners with CPaaS vendors. The goal? Get higher exposure and not working directly with longtail developers in the initial release.

What drove me to even start writing this article? This title of a TechCrunch post: Wish, Netflix, Uber and ~100 others testing WhatsApp’s new Business API

Businesses aren’t waiting for RCS. They are trying to figure out how to communicate with their customers via WhatsApp.

They had Line, WeChat, Facebook Messenger. And they’re still aiming for WhatsApp – a messaging service that isn’t even a US-thing.

Which brings me to the main thing – business to consumer is now a social messaging realm. Carriers have lost that domain as well.

1 Billion Defines the Moat

Remember ubiquity? Here’s what it takes to be interesting:

1 Billion Monthly Active Users

Who has that number today?

Facebook (WhatsApp + Messenger), Apple Business Chat and WeChat. WhatsApp being the biggest one are redefining this market. You hear a lot about how customers still phone businesses and chat isn’t catching up with contact centers. That might be true, but only partially.

Today’s chat solutions usually require being on the company’s website. SMS hasn’t proven itself in a large scale for anything other than notifications to customers on orders and transactions. Whatsapp can change that – and to that extent, any of the other 1B+ MAU social messaging apps.

RCS? With what billion users exactly?

With the large social networks, a 100 million monthly active users seem like a rounding error.

Focus is on Customer Care – Not Marketing

Another interesting aspect (and difference) is that social networks are keeping user identity and access close to their chest. While WhatsApp is using phone numbers for identity, piggybacking on carriers in a way, they are not allowing anyone access to a user without the user’s permission. This means:

1. Businesses can’t “spam” users by sending them unsolicited messages just because they know their phone number or user name
2. A user must first approach the business. Inbound use cases are the focus here, which lends itself nicely to support and purchasing activities
3. Outbound marketing campaigns, ads, promotions – these aren’t something that are encouraged at the moment

What these networks are trying to do is to get businesses and consumers off their SMS communications and shift it to their network. To do so, they plan on offering a superior experience. They are doing that not only by adding richness over the limited 160 character experience of SMS, but they are also making sure this will be a useful service to their user base and won’t be considered spammy.

Will there be other avenues opened to businesses on social networks to interact with users through marketing campaigns and outbound messaging? Sure. But it isn’t the first priority. The market needs to be created first.

Where Can We Go Next?

We are headed towards an omnichannel interaction model.

To me that means that a business will meet a customer wherever it is comfortable for the customer in the context of that specific interaction.

A customer may prefer a phone call at one interaction, but a chat over WhatsApp on another.

The challenge here is that different customers may prefer different social networks. Or aren’t even approachable on some of the social networks. This isn’t going to change any time soon either. The number of social networks is still growing, and while we have a few huge players, others are important to specific populations.

Businesses will need to rely on multiple such channels if they want to reach out to a larger target audience of potential customers.

Back to RCS

It is coming. In some carriers. On some devices. In some form.

Is it going to take back ownership of the interactions from social networks? No.

What it can be, is just another channel. Right next to the rest. It will only become important if it can make that 1 billion monthly active users mark.

Oh, and it will need to succumb to the rules of engagement laid out by social networks today, around business-to-user permissions.

The post Social Messaging != Carrier Messaging (the stories of Whatsapp Business API & Apple Business Chat) appeared first on BlogGeek.me.

I has been more than a year since Apple first added WebRTC support to Safari. My original post reviewing the implementation continues to be popular here, but it does not reflect some of the updates since the first limited release. More importantly, given its differences and limitations, many questions still remained on how to best develop WebRTC applications for Safari.

I ran into Chad Phillips at Cluecon (again) this year and we ended up talking about his arduous experience making WebRTC work on Safari.

Continue reading Guide to WebRTC with Safari in the Wild (Chad Phillips) at webrtcHacks.

The warm welcome note for Tsvetomir Dimitrov joining the Kamailio Developers Team, who recently contributed the ims_ipsec_pcscf module. In the past, he submitted fixes and improvements to the code for other existing modules related to IMS/VoLTE and SMS operations.We are glad to have the team expanded and we are looking forward to more contributions from Tsvetomir!Thanks for flying Kamailio!

Visual design tools in CPaaS are now a part of the offering.

In October 2017, almost a year ago, Twilio announced Studio. I wrote at the time a lengthy article about my thoughts on Twilio Studio and CPaaS. My closing paragraph then was this one:

It will be interesting to see how competitors would react to this in the long run, and even more interesting to see what will Twilio Studio grow into.

Then in January 2018, I wrote about the 7 CPaaS Trends to Follow in 2018. The ones I zeroed in on:

1. Serverless – a few more CPaaS vendors now offer serverless
2. Omnichannel – more about that in one of my next articles
3. Visual/IDE – guess why I wrote this article?
4. Machine learning and Artificial Intelligence – Got a whole new report covering AI in RTC if you are interested
5. AR/VR – planning to write about this one a bit later
6. Bots – they’re already everywhere, directly linked to both omnichannel and AI
7. GDPR – everyone covers that now in CPaaS

Not sure which CPaaS vendor to use? Check out my free CPaaS Vendor Selection Matrix. It will give you the KPIs to look for.

Download the CPaaS Vendor Selection Matrix

Guess what happened since with Visual/IDE?

Messagebird introduced Flow Builder: “The power of our Voice and SMS solutions at your fingertips, without writing a single line of code.”

Plivo announced PHLO on August: “A whole new visual way of integrating communications that would empower developers to design collaboratively, build visually and deploy instantly.”

 

Voximplant came out with Smartcalls: “a smart and flexible tool that helps you create outbound call campaigns in no time”

All of these CPaaS players invested into a Twilio Studio-like tool.

Let’s check out what each player did and why.

Twilio Studio

Where it all started (even if there were tools before or in parallel to it).

Studio’s entry point is either an incoming message, an incoming call or a REST API call. From there, the actions include things you do with messages and phone calls, along with the ability to execute generic functions.

A nice touch to Studio is its revision control system – it saves past changes made to the flows you built, allowing switching back and forth between revisions. It would be nice to have named revisions, some automated verbose explanation of changes made, etc.

Messagebird Flow Builder

Messagebird Flow Builder is focused around SMS. The inputs you can use for it are either an incoming SMS or an incoming webhook API call. Once in the “flow”, you can branch the flow based on the time and date or other conditions related to the contents of the message. The end result? An outgoing SMS, email or webhook. There’s a bit more to it than that, like the ability to manage subscriptions in Messagebird or wait for certain replies inside the flow.

What I like about the Messagebird Flow Builder is that it is rigid in how it outlines the boxes and their connections – it doesn’t let you move boxes around (a cool feature that got tiresome rather quickly on me in other tools here – Studio and PHLO).

Plivo PHLO

Plivo PHLO is a me-too Twilio Studio tool.

It has the same entry points, node types and capabilities, assuming you’re interested in SMS and voice calls that is. Where Twilio Studio offers more generic “Messages”, Plivo has only SMS. This is probably fine for most users.

The only thing I couldn’t find in PHLO is the ability to execute an arbitrary JS function. There’s also no revision control as of yet. Other than that, PHLO is a rather straightforward too to use.

Voximplant Smartcalls

The Voximplant Smartcalls service is different in nature. Where the rest of the pack here is focused on incoming events that trigger action, Smatcalls is all about campaigns. And all about voice.

You can create a scenario. Scenarios in Smartcalls is a visual decision tree of what to do with an outgoing call. You dial, someone answers, you play a specific recording, maybe ask them to click on digits, etc.

You can do things like send email or call a REST webhook, but the purpose of it all is to drive an automated outbound voice campaign: once you have a scenario, you create a campaign. A campaign is a time window, a scenario and a list of phone numbers to dial out to. Smartcalls does the rest to automate the scenario created across all phone numbers at the specified time window.

On Pricing

Here things get somewhat murkier.

Do you pay for using the designer tool itself when it gets invoked? (you do with Twilio Studio)

Do you need to pay for the communications used within the flows created? (you don’t with Voximplant Smartcals).

Plivo, being the shadow of Twilio for voice and SMS, decided not to price the use of PHLO at all, and make that an important part of their announcement as well:

“That’s why, in addition to bringing in 100% Plivo-API support out-of-the-box, we are also making it FREE to build using PHLO. This is not just a commercial decision. This is our stake in the ground — as we truly believe this is how the communication capabilities of the future will be built.”

Here’s the visual from the product page:

Will this create pressure on Twilio? I doubt it, but who am I to say?

A Comparison Table

I put these tools in a table, to see where each one is focused:

 

Twilio Studio Messagebird Flow Builder Plivo PHLO Voximplant Smartcalls Focus Inbound Inbound Inbound Outbound Medium Voice, SMS, Omnichannel messages SMS Voice, SMS Voice Cool factor Revision control Really easy to use Campaign management Flow pricing Per flow invoked Free Free Per minute charges Communications pricing Not included Not included Not included Included A Word about iPaaS

Maybe a few paragraphs…

iPaaS stands for Integration Platform as a Service. The poster child service here is probably Zapier, allowing the connectivity of one service to another. I use it daily in my own business to power many of the integrations on this website.

Many of the CPaaS players have been working on enabling their use via Zapier, so a user doesn’t need to be a developer to send a message for example. Being able to build more complex communication flows using a visual builder sits well with this approach.

What will be interesting to see is how the two play out with each other, if at all. Will these visual builders get integrated into Zapier? Will these visual builders include easier integration points to other services besides what they themselves offer and a rudimentary capability of invoking a REST call?

Welcome to Visual CPaaS

CPaaS is more than making communication API calls or offering github repositories. In the past two years we’ve seen some interesting movements in this space and innovations coming out.

I can’t wait to see what will come next.

Not sure which CPaaS vendor to use? Check out my free CPaaS Vendor Selection Matrix. It will give you the KPIs to look for.

Download the CPaaS Vendor Selection Matrix

The post The CPaaS Version of iPaaS: MessageBird & Plivo Join the Twilio Studio Bandwagon appeared first on BlogGeek.me.

WebRTC isn’t the only cool media API on the Web Platform. The Web Virtual Reality (WebVR) spec was introduced a few years ago to bring support for virtual reality devices in a web browser. It has since been migrated to the newer WebXR Device API Specification.

I was at ClueCon earlier this summer where Dan Jenkins gave a talk showing that it is relatively easy to add a WebRTC video conference streams into a virtual reality environment using WebVR using FreeSWITCH.

Continue reading VR Video Calling with WebRTC and WebVR (Dan Jenkins) at webrtcHacks.

A new module has been contributed recently by Tsvetomir Dimitrov extending Kamailio capabilities on IMS and VoLTE.The module is named ims_ipsec_pcscf and contains the methods for IPSec initialisation and deinitialisation needed for using Kamailio as a Proxy-CSCF. The documentation of the module is available at:

• https://kamailio.org/docs/modules/devel/modules/ims_ipsec_pcscf.html

Enjoy!Thanks for flying Kamailio!

On the 28th of August 2018, Daniel-Constantin Mierla will join the ClueCon Weekly Conference to talk about what’s new lately in the Kamailio ecosystem.ClueCon Weekly is a live 1-hour video conference session organised every Wednesday by the FreeSwitch project team, everyone can join with a modern web browser or dialing in from PSTN. Details about how to connect are available at:

• https://freeswitch.org/confluence/display/FREESWITCH/ClueCon+Weekly+Conference+call

You are welcome to join and discuss or ask questions about Kamailio as well as its usage together with FreeSwitch.If there is a topic that you would like to be approached, email to and we will try to plug it in.Thanks for flying Kamailio!

Kamailio SIP Server v5.1.5 stable is out – a minor release including fixes in code and documentation since v5.1.4. The configuration file and database schema compatibility is preserved, which means you don’t have to change anything to update.Kamailio® v5.1.5 is based on the latest source code of GIT branch 5.1 and it represents the latest stable version. We recommend those running previous 5.1.x or older versions to upgrade. There is no change that has to be done to configuration file or database structure comparing with the previous releases of the v5.1 branch.Resources for Kamailio version 5.1.5Source tarballs are available at:

• https://www.kamailio.org/pub/kamailio/5.1.5/src/

Detailed changelog:

• https://www.kamailio.org/pub/kamailio/5.1.5/ChangeLog

Download via GIT: # git clone https://github.com/kamailio/kamailio kamailio
# cd kamailio
# git checkout -b 5.1 origin/5.1Relevant notes, binaries and packages will be uploaded at:

• https://www.kamailio.org/pub/kamailio/5.1.5/

Modules’ documentation:

• https://www.kamailio.org/docs/modules/5.1.x/

What is new in 5.1.x release series is summarized in the announcement of v5.1.0:

• https://www.kamailio.org/w/kamailio-v5-1-0-release-notes/

Thanks for flying Kamailio!

A web survey says… that you need to join in to learn more about real time video technology.

I’ve partnered up with Vidyo on a survey they are working on with Hanover Research. This one is focused on how real time video technology gets used in different industries, as well as how decisions are made when choosing the technology stack to use.

Fill out the survey

I worked as a programmer during my time at school. It was fun, but it is hard to call it professional work (although the last place was a startup focused on medical patient records in the Israel healthcare system). My first “grownup” job as a developer was at a video conferencing company. You can say I’ve been spending my time in front of a webcam for more than half of my lifetime, communicating with peers and colleagues.
In the last several years, as a consultant, much of my work is conducted online. At times with customers that I have never met face to face – only through a video conference.

At testRTC, almost all of our sales are done through video conferencing. Recently, we had a conference call conducted on one of the web conferencing platforms that was selected for use by our customer (we tend to use Google Meet by default, but flexible to use whatever the customer is comfortable with). People from that company always join with their video turned off. I forgot mine on for a couple of seconds, which allowed me to use it as an excuse to ask the person who I had working relations with for several months now to see her as well. She obliged, and for a brief few seconds it felt more human. Now it is a lot easier for me to have a mental image of that person when she speaks. This adds volumes to the connection between us humans.

For me video isn’t a gimmick. It is a critical tool.

Are all my calls video calls? No. Just like I use messaging but still use voice calling. Different tools for different jobs.

 

When Vidyo asked me to join them for the survey, I automatically said yes. As someone who uses video on a daily basis, I am always interested in understanding how others are making use of video if at all.

The survey Vidyo is doing comes to answer one main question: How (and why) video gets embedded into different businesses?

For me, one of the more interesting questions relates to the applications businesses develop, and if they don’t plan on adding communication functions into them, then why. Understanding what barriers and challenges people see in these technologies can help us as an industry decide where to put our focus.

If you are reading this blog and want to help me out in understanding the industry better, would you be so kind as to fill out this online survey? If you do, you’ll have my thanks as well as a copy of the research findings.

Fill out the survey

The post Understanding video tech in the enterprise: a web survey appeared first on BlogGeek.me.