News from Industry

Siremis v4.4.0 is out – the web management interface for Kamailio SIP Server.

There were few changes to mark the last release compatible with Kamailio v4.x series. Future development will focus on compatibility with Kamailio v5.x.

Step by step installation tutorial, screenshots and demo are available on the web at:

• Installation Tutorial
• Screenshots (shared with v3.2.1)
• ChangeLog

Siremis is used during Kamailio Advanced Training classes for management of SIP server, a good opportunity to learn about Siremis itself, see more details at:

• https://www.asipto.com

Time to start another ongoing project. This time – my Monthly Virtual Coffee sessions about WebRTC, CPaaS, APIs and comms in general.

Some time in 2015-2016, I decided to host Virtual Coffee sessions. Once a month, I’d pick a subject, create a presentation and host a meeting with my customers. All of them. It was open for questions and it was fun. It stopped because… I don’t know. It just did.

Ever since then, I wanted to do something similar. I found I like talking and interacting with people, and I want to do it more.

Which is why I am now announcing the new Virtual Coffee with Tsahi.

Here’s how it will go down:

• The live sessions are free to join. For anyone
• Recording of the session will be available to my customers only
• Topic will be selected and announced a week or more in advance
• This will happen once a month. Hopefully

How will this be announced?

I won’t be using this blog to publish future sessions – sorry.

The sessions will be announced through Crowdcast (the service I started using for such events lately), so follow me there. And through my newsletter, so if you’re not subscribed – do it now.

What topics will I cover?

I really don’t know…

If you want something specific – drop me a line.

Our 1st Virtual Coffee together

The first topic I want to tackle?

CPaaS, WebRTC, Differentiation and M&A

When? May 23 @ 15:30 EDT

There are over 20 different CPaaS vendors out there, and that number is growing and shrinking at the same time:

• AT&T closing their Enhanced WebRTC APIs
• Cisco acquiring Tropo
• Vonage acquiring Nexmo
• CLX Communication acquiring Sinch
• RingCentral adding APIs to its UCaaS
• TeleSign announcing self service CPaaS, only to get acquired
• …

I want to take the time to review some of this M&A activities, as well as show how different vendors are trying to differentiate themselves from the rest of the crowd.

Join me for this Virtual Coffee with Tsahi

Oh – if you you have questions for this already – just ask them on Crowdcast once you register.

See you there!

The post Announcing my Virtual Coffee sessions appeared first on BlogGeek.me.

WebRTC establishes peer-to-peer connections between web browsers. To do that, it uses a set of techniques known as Interactive Connectivity Establishment or ICE. ICE allows clients behind certain types of routers that perform Network Address Translation, or NAT,to establish direct connections. (See the WebRTC glossary entry for a good introduction.) One of the first problems is for […]

The post Am I behind a Symmetric NAT? appeared first on webrtcHacks.

What is WebRTC and What is it Good For? This 7-minute video provides a quick introduction to WebRTC and demonstrates why it is growing in importance and popularity.

Covered in this video:

• What is WebRTC?
• Current state of adoption of WebRTC
• Why is it so much more than just a video chat enabler
• The power of “Open Source” in WebRTC
• How WebRTC works
• Five reasons to choose WebRTC

What is WebRTC?

WebRTC is an HTML5 specification that you can use to add real time media communications directly between browser and devices.

Simply put:

WebRTC enables for voices and video communication to work inside web pages.

And you can do that without the need of any prerequisite of plugins to be installed in the browser.

WebRTC was announced in 2011 and since then it has steadily grown in popularity and adoption.

By 2016 there has been an estimate from 2 billion browsers installed that are enabled to work with WebRTC. From traffic perspective, WebRTC has seen an estimate of over a billion minutes and 500 terabytes of data transmitted every week from browser communications alone. Today, WebRTC is widely popular for video calling but it is capable of so much more.

A few things worth mentioning:

• WebRTC is also completely free
• It comes as open source project that has been embedded in browsers but you can take and adopt it for your own needs
• This in turn has created a vibrant and dynamic ecosystem around WebRTC of a variety of open source projects and frameworks as well as commercial offerings from companies that help you to build your products
• WebRTC constantly evolving and improving, so you need to keep an eye on it

So, how does WebRTC work?Code and API

It is important to understand from where we are coming from: If you wanted to build anything that allowed for voice or video calling a few years ago, you were most probably used C/C++ for that. This means long development cycles and higher development costs.

WebRTC changes all that: it takes the need for C/C++ and replace it with a Javascript API.

WebRTC comes with a Javascript API layer on the top that you can use inside the browser. This makes it far easier to develop and integrate real time communications anywhere. Internally, WebRTC is still mostly implemented using C/C++, but most developers that use WebRTC won’t need to dig deep into these layers in order to develop their applications.

Availability

WebRTC today is available in most modern browsers. Chrome, Firefox and Microsoft Edge support it already, while Apple is rumored to be in the process of adding WebRTC to Safari.

You can also take WebRTC and embed it into an application without the need of browser at all.

Media and access

What WebRTC does is allow the access to devices. You can access the microphone of your device, the camera that you have on your phone or laptop – or it can be a screen itself. You can capture the screen of the user and then have that screen shared or recorded remotely.

Whatever WebRTC does that does in the real time, enabling live interactions.

WebRTC isn’t limited to voice and video. It allows sending any type of data any arbitrary data

There are several reasons WebRTC is a great choice for real time communications

1. First of all, WebRTC is an open source project
   1. It is completely free for commercial or private use, so why not use it?
   2. Since it is constantly evolving and improving, you are banking on a technology that would service you for years to come
   3. WebRTC is a pretty solid choice – It already created a vibrant ecosystem around it of different vendors and companies that can assist you with your application
2. WebRTC today is available in browsers and most modern browsers today support it
   1. This has enabled and empowered the creation of new cases and business models
   2. From taking a Guitar or a Yoga lesson – to medical clowns or group therapy – to hosting large scale professional Webinars. WebRTC is capable of serving all of them and more
   3. WebRTC not limited to only browsers because it is also available for mobile applications
      1. The source code is portable and has been used already in a lot of mobile apps
      2. SDKS are available for both mobile and embedded environments so you can use WebRTC to run anywhere
   4. WebRTC is not only about for voice or video calling
      1. It ss quite powerful and versatile
      2. You can use it to build group calling service, add recording to it or use it only for data delivery
      3. It is up to you to decide what to do with WebRTC
   5. WebRTC takes the notion of communication service and downgrades it into a feature inside a different type of service. So now you can take WebRTC and simply add communication in business processes you need within your application or business

So what other choice do you really have besides using WebRTC?

The idea around WebRTC and what you can use it for are limitless. So go on start building whatever you need and use WebRTC for that.

Embed this video on your own site for free! Just copy and paste the code below…

<iframe src="https://player.vimeo.com/video/217448338" width="640" height="360" frameborder="0" webkitallowfullscreen mozallowfullscreen allowfullscreen></iframe>

 

The post What is WebRTC and What is it Good For? appeared first on BlogGeek.me.

Recently I discovered a tutorial written on Medium blogging platform by Igor Olemskoi, published about two months ago, covering Kamailio and Siremis installation, named “Kamailio SIP proxy — installation and minimal configuration example“.It is a good reading for anyone fresh to Kamailio, providing guidelines to get started with a SIP proxy installation, along with Siremis web management interface.Click here for the link to the blog article.Should you write any tutorial involving Kamailio, we are more than happy to share it with the community via the website. Just contact us via mailing lists! Ping us also if you are aware of any other resource that worth sharing.Thank you for flying Kamailio!

Contact centers are the main adopters of WebRTC still. This is clearly reflected by my infographic of the WebRTC state of the market 2017.

Motto:“This ‘telephone’ has too many shortcomings to be seriously considered as a means of communication. The device is inherently of no value to us.”

Western Union telegraph company memo, 1877.

Think you know how WebRTC fits in a contact center? Check out with The Complete WebRTC Contact Center Uses Swipefile

Get the swipefile

Recently, Jaroslav from iCORD, told me the stats they now see from the contact center deployment they have in O2 Czech Republic, who also happen to be their parent company.

How is O2 CZ making use of WebRTC in their Contact Center?

What they did isn’t the classic approach you will see to WebRTC in contact centers, but rather something slightly different. If you are a customer of O2 CZ and you are thinking of making a purchase on their website, you have the option to leave a number for them to immediately get back to you:

And yes – there is also an “exit intent”  on that sales page, so if try to leave this page, it will appear as a popup.

How is a phone call related to WebRTC you ask? Well… it isn’t. Unless you factor in the fact that we now know what web page the user is on.

What happens next, is that a contact center agent will call back to the user, and the user will see something new on his browser – a shared space between him and the agent that just called him.

This shared space will enable the agent to browse the same page the customer was on, and move on from there elsewhere. It also includes annotations – the agent can draw or mark things on the screen. One last thing – the user will see the video of the agent, but will not share his video.

See? They even haggle and write down discount prices right on the webpage.

Now, if the interaction started with a phone call, the agent in the contact center can instruct the customer to go to the O2 CZ website and enter a PIN code there – and magically get to the same experience.

Here’s a diagram to show the communication channels we now have between the customer and the contact center agent:

Why this approach?

1. Customers continue using the phone. Same way they did in the past
2. There’s no reliance on the customer’s camera, microphone or muted speaker volume
3. It got stitched right into the existing contact center O2 CZ already had in place

The results

But was this effective? Was it worth the effort?

O2 CZ have been running this contact center service throughout 2016, and took the time to analyze the results. They did so only for sales related calls – the money makers.

Here’s what they found out:

Using this approach is much more efficient than a simple phone call.

Let’s stop right here for a second and soak that statement.

We’re talking about a contact center.

Of a mid-sized European carrier (4 million subscribers).

The type of those where I am told over and over would NOT adopt WebRTC because it does not support Internet Explorer 4. Oh. And this specific service falls back to Flash if the customer’s browser doesn’t support WebRTC and even decreases further in feature set to static screenshot and PDF file sharing for those who don’t even support Flash.

And they are already doing it for a full year.

Successfully.

In production.

In front of live customers.

Who would have thought a non-startup company that isn’t located in Silicon Valley and operated by 16-year olds would be able of doing such a ridiculous thing like deploy WebRTC in production directly to where money gets negotiated with customers.

— end of rant —

Back to the results.

Call length on average dropped

It takes 30% less time to negotiate and close a deal than a regular phone call and considerably shorter than text chat. This may seem a bit backwards – the fact that chat takes the longest and a video session the shortest, but that’s the experience of this contact center.

How about succeeding to close a deal and make a sale? WebRTC gets closed deals 25% more than regular phone calls. Chat is slightly less successful than WebRTC but more successful as phone. These values were measured on session landing at sales agents’ desk once those irrelevant and redirected were filtered out.

And the customer satisfaction? Over 20% rate the service 5 stars at the end of the interaction and 7% left positive textual evaluation of the service. Compared to the traditional IVR system that’s really high.

Where does this lead us?

1. There are many ways to deploy WebRTC in a contact center
2. WebRTC is already being used by contact centers – successfully
3. Done right, there’s huge value in adding WebRTC to your customer engagement. It adds efficiency and improves customer satisfaction, resulting with higher value to both sales and care

Want to learn more of the various ways WebRTC fits in with contact centers?

Get my free WebRTC Contact Center Uses Swipefile

And if you are looking for more information about the O2 CZ deployment details – especially the technical ones, Jaroslav will be happy to have a conversation with you.

 

The post What to Expect when Deploying WebRTC in Contact Centers? appeared first on BlogGeek.me.

Just few days and the 5th edition of Kamailio World Conference & Exhibition starts. Two days and a half full with technical tutorials, presentations, open discussions and demos. With more participants than the previous edition, the event consolidates the ecosystem around Kamailio and other VoIP related projects and products such as Asterisk, FreeSwitch, Janus, Jitsi, OpenBaton, CGRateS, Homer Sipcapture, SIPVicious, RTPEngine, Zoiper or SEMS.Along the event, Obihai, Telnyx, Digium, NG Voice, Core Network Dynamics and FhG Fokus will be present in the exhibition area showing demos of their services and products.The topics span from VoIP/SIP-based platform scalability and security, to next generation emergency services, usage in broadcasting industry, 5G and IMS/VoLTE, IoT and WebRTC:

• https://www.kamailioworld.com/k05/schedule/

With the great help of our sponsors, FhG Fokus, FhG Forum, Flowroute, Telnyx, Sipwise, Asipto, Sipgate, Simwood, Obihai, Digium, NG Voice, Core Network Dynamics, VoiceTel, Evariste Systems, Pascom and VUC, we are ready for another amazing edition of Kamailio World!See you in Berlin next week!

How do you find good WebRTC outsourcing talent?

At least once a week.

That’s about the current rate in which I bump into a hiring or talent question related to WebRTC.

Recently, I got a few calls with companies that went through the process of working with an outsourcing vendor who developed their app and got stuck.

Sometimes it was due to bad blood going between the two companies. But more often than not it was because the company that approached me wasn’t happy with the delivered results. The application that was developed just didn’t really work as expected. Looking at some of these apps, it was easily apparent to see that the developers were clueless about WebRTC. Things like wrong NAT traversal configurations (or none at all), or the use of mesh media delivery for large multiparty video sessions are the most obvious warning signs here.

If I had to think why this is so, my guess it boils down to three reasons:

1. WebRTC is still rather new. 5 years. So there’s still not enough mileage on it for most developers
2. It isn’t Web and it isn’t VoIP. But it is also Web and VoIP together. Which means many seem to misunderstand it
3. Skilled WebRTC developers are hard to find. Less than 12,000 profiles with that term on LinkedIn’s now 500 million profiles

When you go and ask from an outsourcing vendor to build you a service, the answer you will get is “sure thing”. And then a price and a timeline. That’s their business, and most would often use that project as their jumping board towards another domain of expertise for them. Many of these outsourcing vendors won’t invest in learning new technologies without a customer paying for that investment.

This means that a lot of the market for WebRTC outsourcing is a market of lemons. Which is why it is so important you check and validate your prospective WebRTC outsourcing vendor before signing an agreement with him.

Picked a WebRTC outsourcing vendor? Here are a few quick telltale signs that will help you determine just how knowledgeable he is about WebRTC:

Get the WebRTC Outsourcing Vendor Signals swipefile

Here are 6 questions to ask yourself before you hire a WebRTC outsourcing vendor.

#1 – Do I know my own requirements?

There are two parts to knowing your requirements from the product:

1. Knowing and understanding your business and the interactions you want for it
2. Understanding what’s realistic for you with WebRTC to set your expectations accordingly – this also means understanding the costs of certain features versus how important they are for you

For that, I suggest you use something like my WebRTC requirements template.

#2 – Am I their first WebRTC customer?

This is a biggie.

Try. Not. To be. Their FIRST. Customer. That does. WebRTC.

Don’t be their first customer doing WebRTC.

Make sure you’re not the first one they build a WebRTC product for.

Their first WebRTC project? You shouldn’t be the one they do it for.

Got the point?

One more time if you missed it:

I knew that picture (and font) would come in handy some day.

#3 – Is the team working for me built a WebRTC product before?

This one is somewhat tricky, and I must say – a bit new in my list of top questions to a WebRTC outsourcing vendor.

If you’ve been reading this from the start instead of skimming through, you might have seen the number 12,000. This number is higher than the number of profiles in LinkedIn that have the term WebRTC in them anywhere. It means that with some of these WebRTC outsourcing vendors, the people put in place on your project might not be the ones who know WebRTC – these are already fully booked by other clients – or they might have gone elsewhere (with the demand of WebRTC developers, I wouldn’t be surprised to see them learn the trade in one vendor and move on to the next).

I’ve seen it happen once or twice before.

So make sure that not only does the vendor knows WebRTC well – he is also placing the right people on your project. And understand that there are times when not the whole team must know WebRTC to develop a successful project.

#4 – Can I validate what they build for me?

Developers who don’t know and understand WebRTC won’t be able to deliver a commercial product for you.

If they don’t understand the server side of WebRTC and its implications (check my free mini course on WebRTC server side), then the end result will run great between you and your pal sitting next to you, but when you take it to production it will fail spectacularly.

Things to look for:

1. Not configuring NAT traversal properly (public STUN servers, no TURN servers, no TCP or TLS configurations for TURN)
2. Using mesh instead of mixing or routing the media (see here) – in plain English, not using a media server in scenarios that beg for it to be used
3. Not testing for scale (see here)
4. Not checking the result in varying network conditions

While some of these can be solved just by more testing (and focused testing – one where the tester actually knows what to look for), there are times when the architecture selected for the product is just all wrong. It should have been apparent from the get go that it won’t hold water.

But anyways – make sure you’ve got a plan in place on what and how to test to validate that that thing that was given to you as the finished good is actually the finished good and not finished for good.

#5 – Should I ask for something On Premise or CPaaS based?

This goes back to #1, but slightly different. Probably should have placed it as #2.

Developing your own product from scratch will be more expensive than using a CPaaS vendor. CPaaS vendors are those vendors that take the whole hassle of real time communications, wrap it with their nice API and manage it all for you (and yes, I wrote a report about them).

Whenever I sit down with an entrepreneur that wants a product I start there when it comes to vendor and technology stack selections. Trying to understand his restrictions and requirements. Oftentimes, entrepreneurs are deterred by the seemingly high pricing of CPaaS vendors. Especially at the beginning – when they believe they will get to a million monthly active subscribers within a month. Well… it won’t happen to you. And if it does, a VC or two will probably be happy to foot that bill, understanding you probably found a real boon.

What should you do?

1. Read this one. And then read this one from Chris Kranky
2. Make your  decision on that build vs buy decision (in both you will be building – don’t worry)
3. Revisit your initial requirements
4. Revisit that vendor you plan on working with

#6 – Who is the owner of this project on my end?

Someone needs to be the owner of this project on your end.

Yes. You have a WebRTC outsourcing vendor developing this thing for you, but you need someone to have that vendor behave and deliver.

That someone needs to understand WebRTC well enough to handle the requirements, the discussions with the vendor for all the issues that will arise along the way.

I’d also recommend having that someone on the payroll and not external.

If you don’t have such a someone then you effectively selected you for that job. Congrats!

Do Your Homework

If you plan on starting a project that makes use of WebRTC, and you plan on using a WebRTC outsourcing vendor for it, start by doing your homework.

Make sure you have the answers to the questions above.

And if you need help along the way – with the requirements, the architecture, the vendor to select, the process – you know where to find me.

Picked a WebRTC outsourcing vendor? Here are a few quick telltale signs that will help you determine just how knowledgeable he is about WebRTC:

Get the WebRTC Outsourcing Vendor Signals swipefile

The post 6 Questions to Ask Yourself BEFORE Hiring a WebRTC Outsourcing Vendor appeared first on BlogGeek.me.

My physical machine runs Debian Jessie, and it has several LXC containers (mostly Debian and Ubuntu). Now I needed to test some software under CentOS, and I bumped into the following error when installing Apache HTTP server:

Downloading packages: httpd-2.4.6-45.el7.centos.4.x86_64.rpm                                                                        | 2.7 MB  00:00:00      Running transaction check Running transaction test Transaction test succeeded Running transaction   Installing : httpd-2.4.6-45.el7.centos.4.x86_64                                                                                1/1 Error unpacking rpm package httpd-2.4.6-45.el7.centos.4.x86_64 error: unpacking of archive failed on file /usr/sbin/suexec;590112cd: cpio: cap_set_file   Verifying  : httpd-2.4.6-45.el7.centos.4.x86_64                                                                                1/1 Failed:   httpd.x86_64 0:2.4.6-45.el7.centos.4

The thing is, that by default “/usr/share/lxc/config/centos.common.conf” defines the following capability drops:

lxc.cap.drop = mac_admin mac_override setfcap setpcap lxc.cap.drop = sys_module sys_nice sys_pacct lxc.cap.drop = sys_rawio sys_time

So, setfcap capability is required in order to install Apache. Use the following lines in your “/var/lib/lxc/NAME/config” to drop previously defined drops and set up a new list:

# flush all defined drops and define a new list lxc.cap.drop = lxc.cap.drop = mac_admin mac_override setpcap lxc.cap.drop = sys_module sys_nice sys_pacct lxc.cap.drop = sys_rawio sys_time

then restart the container, and “yum install httpd” should run as expected.

Filed under: Networking Tagged: debian, hosting, linux

Security is… complex. Even with WebRTC.

I’ve always been one to praise the security measures placed in  WebRTC.

While WebRTC is a secure protocol by nature, it seems that browsers take different approaches to who needs to take responsibility of any additional means of security.

The gist of it:

• WebRTC is secure by default
• Whenever a developer’s mistake can be thwarted by tweaking WebRTC – it gets tweaked
• Whenever a security hole is found, it gets fixed and deployed by the browser vendors faster than most other companies in the industry can even perceive the notion of a threat

Seriously – what’s not to like?

Recently though, I started thinking about it. How do browser vendors think about security? How much do they take it upon themselves to be the guardians of their users? His trusted guide in the big bad world that is the Internet?

Which brings me to the big one –

Are browser vendors responsible to the actions of their users when it comes to WebRTC?

It seems that they have different approaches and concepts to this one.

Google Chrome

Moto: Users are stupid and should be protected

That’s how I’d put their mindset to words.

getUserMedia

Chrome has long been one to clamp down on where and when can WebRTC be used.

They started off with voice and video working on HTTP and HTTPS, while HTTP access granting to the camera and microphone were forgotten, and required a user’s approval each and every time.

They shifted towards HTTPS only. You can’t access the microphone or the camera in an HTTP page.

Persistence

The decision a user made is persistent. If you granted a domain access to your microphone or camera – Chrome remembers it – for eternity. Your only way of revoking that is by clicking the camera icon on the address bar (if you can even notice it):

Oh, and for persistency – Chrome offers you two choices:

1. Ask when there’s a need (and Chrome will remember the answer for that domain for you)
2. Never ever share your device

No middle-ground here.

Screen sharing

You can share your screen with Chrome.

But it will ask the user each time for his permission.

And to enable screen sharing, you will first need to create a Chrome Extension for your web app and have the user install it. Not a biggie, but a hurdle.

Now, to publish a Chrome Extension on the Chrome Web Store, you’ll need to pay a small $5 fee.

Why? Fraud – obviously:

You see, screen sharing is considered by Google (and most other browsers) as more of a security threat than camera and microphone access.

By forcing the Chrome Extension, Google raises the bar against abuse, and can theoretically remove any abusive accounts and extensions with better tracability to their source.

The only real downside of it? I have over 10 icons on my toolbar now in Chrome, and most of them are for screen sharing on different services. Once a move I remove a few of them to declutter my browser. Yuck.

Mozilla Firefox

Moto: Users are intelligent

Maybe. But not all of humanity. Or even the billion or two that use browsers.

getUserMedia

In Firefox, getUserMedia will work in HTTP.

Not sure if persistence can be configured for Firefox for HTTP websites. I guess it is akin to herd immunity in vaccination. Since Chrome is THE browser, developers make sure their WebRTC service works on Chrome (lets call it Chrome first?) so their service starts by running only on HTTPS anyway.

Persistence

Anyways, Philipp Hancke wrote a great post about getUserMedia and timing with browsers. Here’s how timing looks for appear.in from the moment getUserMedia is called and until it is completed:

Firefox tend to take longer to complete its getUserMedia calls. Philipp attributes it to this little UI design in Firefox:

In Firefox, if you want to decision (allow/disallow) to be persisted, you need to opt in for it. And for appear.in, most people don’t opt in.

This is great, especially for the Don’t Allow option (it is quite a hassle to remove that restriction from Chrome once you decided not to allow such access in a session).

Screen sharing

For screen sharing, Firefox used to have a whitelist of domains you had to register on to get screen sharing to work.

From Firefox 52, this restriction has been removed. Mozilla wrote a post about it, explaining their millions of users around the world about the dangers.

I am not sure about you, but I’ve learned early on as a developer catering to developers that other developers are stupid (if you are a developer, then I am sorry, but bear with me – and read this one while you’re at it). So when I wrote code for developers, I made sure that if they screw things up, we crash spectacularly. The reasoning was, the sooner we crash the faster our customers (who are developers) will fix their bugs – and do that during development – so they won’t get into deadlocks or weird crashes in production that are way harder to find. These were the good old days of C programming.

Now… if developers are stupid, then what would mere users do about their understanding of security and threats?

In Firefox, they need to read and understand that yellowish warning when all they want to do is share their screen now – after all – people are waiting for them to do so in the session already.

With such a warning… I am not sure I am going to be in a trusting mood no matter the site.

While I mostly prefer Firefox approach for getUserMedia permissions, I think Chrome does a better job at it with the extensions mechanism.

Microsoft Edge

Microsoft Edge has started to support WebRTC (finally).

While I a, in the process of installing my Creators update (where I am promised proper support for WebRTC), this will take more time than I have to get some nice screenshots of what Edge is doing.

So I asked Philipp Hancke (like I do about these things).

Here’s what I got:

• Edge enable persistence for getUserMedia
• It has a model similar to Firefox – you need to opt-in for persistency
• It doesn’t support screen sharing yet

Download the WebRTC Device Cheat Sheet to learn more on how to get WebRTC to as many devices and environments as possible.

Are Browser Vendors Responsible for Our WebRTC Actions?

Yes they are.

In the same approach that browser vendors are taking in HTTPS everywhere, removing Flash from the web, protecting against known phishing sites, etc; they need to also protect users from the abuse of WebRTC.

The first step is by not allowing developers to do stupid (by forcing encryption and DTLS-SRTP for example). The second one and just as important is by not allowing users to do stupid.

 

The post Should Browser Vendors be Responsible for their User’s WebRTC Actions? appeared first on BlogGeek.me.

With a bit of delay, given that had to accommodate and accept some requests from the last year queue, I am glad to announces that we can offer two more seats at Kamailio World Conference, May 8-10, 2017, in Berlin.Therefore we are continuing the program from last years, based on the roots and the tight relation of Kamailio project with the academic environment, the eligible people being students enrolled in universities or research institutes (both bachelor and PhD programs qualify) as well as people from underrepresented groups.If you think you are eligible and want to participate, email to <registration [at] kamailio.org> . Participation to all the content of the event (workshops, conference and social event) is free, but you will have to take care of expenses for traveling and accommodation. Write a short description about your interest in real time communications and, when it is the case what is the university or the research institute you are affiliate to.Also, if you are not a student, but you are in touch with some or have access to students forums/mailing lists, it will be very appreciated if you forward these details.Time is short, we already received few requests based on my remark in a message sent out few weeks ago, therefore in a matter of days we will do the selection, so you have to hurry up a bit.More information about Kamailio World is available on the web site:

• https://www.kamailioworld.com

Many thanks to the event sponsors that allowed to continue this program, respectively: FhG Fokus, Asipto, Flowroute, Telnyx, Sipwise, Sipgate, Simwood, Obihai, NG-Voice, Evariste Systems, Digium, VoiceTel, Pascom and Core Network Dynamics.Expect a full house event! Looking forward to meeting many of you in Berlin!

We would like to announce that Guillaume Bour (https://github.com/gbour) has now developer privileges on Kamailio’s gihub project. He has contributed recently a new module named keepalive – more about it at:

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

There is also a pull request from him waiting to be merge to drouting module, adding capability of detecting active/inactive gateways used by the module.Thanks for the contributions so far and looking forward to collaborate in the future!

A new IRC devel meeting to is planned in order to discuss the current major issues and the logistics around the project as well as the plans for next Kamailio releases.The date is Thursday, April 20, 2017, at 14:00UTC (16:00 in Berlin, Madrid, Paris; 15:00 in London; 10:00 in New York, …).A wiki page has been created for it, with more details about how to join the discussions and a draft agenda:

• https://www.kamailio.org/wiki/devel/irc-meetings/2017a

Everyone is welcome to participate, you just need an IRC client or a modern web browser.You are encouraged to add there the topics that you want to be discussed and your intention to join the session in order to plan the schedule properly and estimate more accurately the duration.Thanks for flying Kamailio!

And I have a couple of bonuses waiting for you in this WebRTC course launch.

I’ve been thinking lately on how to make this course available throughout the year, but still “launch” it as a live program once or twice every year. The idea here is to get as many people as possible into the course and improve our current market state (which is rather abysmal):

I always say that WebRTC sits between Web and VoIP, but I guess this says it best.

You can find a million people whose profile contain either “VoIP” or “HTML5”. If you go into specifics, you’ll have hundred of thousands of people with either “SIP” or “Node.js”. But “WebRTC”? Only 11,874 righteous people. We’re a pretty small industry. And those with enough understanding and knowledge of WebRTC? Probably less than that.

What are people challenged with?

The request that comes up almost every time someone contacts me through the blog? It is about finding an experienced WebRTC developer. Here are a few “sound bites” from these emails I am getting:

if we were to hire someone to build our own platform – what qualifications in a programmer would I need to look for?!!

 

We are needing to develop video chat and having a difficult time finding a qualified developer to create this

 

I am seeking a WebRTC engineer to do a peer review on a WebRTC app I had developed in oversees (west Russia.)

 

A couple of thoughts about this

1. If you are a developer and you know WebRTC well, then your talents are in high demand – and if you aren’t conversant in WebRTC, this can be an opportunity for you to learn and grow
2. If you are an employer and you need someone to build a real time comms product, you’re going to be hard pressed to find good talent. Your three best choices are:
   1. Outsource the whole project to a company who is skilled in WebRTC
   2. Hire a freelancer to help your team with the WebRTC parts
   3. Grow your in-house team to make them skilled with WebRTC
3. If you are an outsourcing vendor and you have WebRTC talent, then you’ve got a different set of challenges:
   1. The more projects you take, the more WebRTC talent you need, which means you are back to the hiring challenge as anyone else
   2. Your best WebRTC talent is always on high demand outside, getting job proposals and needing to think how happy are they (so you have a retention issue in your hands, which gets worse due to the high demands of the skillset you are nurturing)

And since the market is so slim on resources (around 12,000 people know WebRTC out of a million who know VoIP – when all VoIP projects are adding WebRTC these days), demand and supply don’t match.

My WebRTC course and its bonuses

Tomorrow, my Advanced WebRTC Architecture course officially launches. If you haven’t enrolled already, then you should seriously consider doing so.

The previous round had almost 100 students going through it with some very positive feedback.

There are going to be a few bonus materials that I will be giving for anyone who enrolls today (or already enrolled):

#1 – 2 live lessons

There are going to be 2 special live lessons taking place. They will be recorded for those who can’t join live. But the lessons as well as the recordings will only be available as part of the course bonuses.

LIVE Lesson 1: Philipp Hancke – Video Quality in WebRTC: The audio and video quality WebRTC provides is amazing. Well, most of the time at least. Sometimes, the video gets pixelated and audio starts dropping out even. What is going on here and why is bandwidth estimation still a problem?

LIVE Lesson 2: Bradley T. Hughes – How to deploy TURN on AWS? TURN servers are boring. They do nothing but relay data. However, they are necessary in WebRTC. Here’s how appear.in’s global TURN infrastructure works – and how you should think of when deploying your own.

So…

2 live lessons.

With top industry experts.

Recorded and available only for you.

#2 – The Perfect WebRTC Developer Profile ebook

Recently I’ve been asked multiple times about CVs and profiles and stuff. It goes both ways:

1. Recruiters want to know what experience to look for in order to find experienced WebRTC developers
2. Developers want to know what to learn and put in their CV to be attractive

I had my own thoughts about it, but decided to take a different route on this one. I went and asked top developers and “recruiters” who work with WebRTC for quite some time now. I asked them about the ideal WebRTC developer and what they’d look for in a CV. Collected the answers and created an ebook out of it:

Who’s in there? Amir Zmora, Arin Sime, Chad Hart, Emil Ivov, Gustavo García, Iñaki Baz Castillo and Philipp Hancke.

You’ll get to see what they think about WebRTC developers and what it means to be a WebRTC professional.

#3 – WebRTC Course FAQ

There are a lot of popular questions out there about WebRTC. You can find them lurking on webrtc-discuss forum, stackoverflow, Quora and elsewhere. But what are the answers? And how should you go about finding them?

What I did in the past few weeks was collect questions and map them to the course lessons. To these questions I provided short and clear answers for you, packaging it all in a neat document.

Now, you can use these questions to tackle specific issues you bump into – or to check how much you understood of the lessons of the course. Hell – if you need to recruit someone – you might as well use it as some good questions to ask to gauge experience.

What if you are not sure?

Besides looking at the testimonials from previous students, I can suggest checking out two things:

1. My free WebRTC server side mini-course. You can expect this kind of content in the course itself, just on a deeper level, on a lot more WebRTC related topics AND with the option of asking questions on the online course forum or during the live Office Hours
2. Join me for the WebRTC Course AMA on Wednesday this week. I will be answering any questions related to WebRTC or the course, so you can make your decision about enrolling to the course (or just get some free advice for your current project)

What if you wait and don’t enroll today?

Bonuses will go away in 48 hours.

After that, the only price plan available for the course will be the Plus price plan and it will only include the Office Hours for the initial duration of this course.

My suggestion?

Enroll now to the Advanced WebRTC Course

The post How to find (or create) WebRTC Developers? appeared first on BlogGeek.me.

NanoPi NEO2 by FriendlyElec is a new sub-$20  Linux microcomputer, built on Allwinner H5 SoC, providing a Gigabit Ethernet and USB 2.0 interface. Also additional interfaces are possible via expansion headers (needs some soldering work). The board is equipped with 512MB DDR3 RAM.

It is highly recommended to buy the heatsink alongside with the board. The CPU is heating up quite significantly, and it needs cooling. With “stress -c 4” CPU load test, “armbianmonitor -m” shows the core temperature rising up to 75C. The board sustains long-term load under such conditions. But with a fan, the core temperature drops below 40C, and the power consumption drops significantly too.

The plastic 3D-printed enclosure is of little use. First, it’s quite easy to break when you insert the board. Also it does not fixate the heatsink properly.

So, I ended up in using the original cardboard packaging as a base for the board, just to avoid extra touching of electronic circuits, and to fixate the USB power cable:

Armbian nightly image booted without problems. Up to now, I noticed the following minor problems with it:

1. it does not come up after reboot;
2. “cpufreq-info” complains about unknown driver.

Network traffic tests with tcpkali (debs, deb build scripts) demonstrated that the CPU is able to saturate the Gigabit Ethernet port with TCP traffic, reaching above 900Mbps throughput.

All in all, this board looks much more reliable than Orange Pi Zero: it can work for long hours with an  USB Wifi dongle, whereas OPI0 was hanging up after few minutes of work (using the same USB power cable and power source and the dongle).

 

Filed under: Networking Tagged: arm, iot, linux, networking

Mikko Lehto has joined the developers group on Kamailio gihub project. He has made a lot of good pull requests over the time, specially to documentation and unit tests that could have been just committed directly without delay on waiting for a review, saving also time from the other developers.Taking time to express my thanks for all his contributions so far and reviving those unit tests, and I am looking forward to future collaboration!Thank you for flying Kamailio!

All the mailing lists related to the Kamailio project are using now lists.kamailio.org as primary domain, including sr-dev and sr-users.From now on, the emails to these mailing lists should be addressed to:

• [email protected]
• [email protected]

The previously used domain (lists.sip-router.org) should still work, being redirected to lists.kamailio.org, so existing mailing list discussions can go on as usual.If anyone is encountering any problems or notices some information on kamailio.org that needs to be updated, do not hesitate to contact us.In addition, the mailing lists URLs should use now HTTPS, browsing the archive with plain HTTP being redirected to HTTPS URLs.

Have questions about my course? Here’s a WebRTC Course AMA for you.

Later this week, I will be opening my Advanced WebRTC Architecture course for enrollment again.

Last year, I decided to launch a course to teach WebRTC. Something different than just going through the WebRTC APIs or explaining the network specification. The end result? A 100 people enrolled and when through the course (!) And more than that – people seemed to be genuinely satisfied with it (!!)

It was fun, so it is time to do it again.

While I am changing and adding stuff to the course, the baseline material is going to stay the same – most of it is “timeless” anyway.

I am adding to this round a couple of things, and this one I want to mention two of them:

#1 – Corporate Plans

The course now has a corporate plan, for larger teams who need to use WebRTC. I’ve got a couple of companies already enrolled to it, which is great.

Corporate plans include a private Slack forum for Q&As alongside the course’ forum. They also include a corporate badge that you can use on your own site, along with their logo on my own site as Corporate Partners.

If you want to learn more about the corporate plans, check out the course syllabus (PDF).

#2 – Course AMA

Philipp forced my hands on this one…

Really looking forward to @tsahil's #WebRTC architecture course: https://t.co/srDBNUuN46
We have a bet running if he can teach me things!

— Philipp Hancke (@HCornflower) April 7, 2017

Only thing left to do is…

But seriously.

I am trying to make this the best place for people to get their WebRTC education.

For those who aren’t sure yet, I’ll be hosting a WebRTC Course AMA, where you can Ask Me Anything. About the course. About WebRTC. About me. About the weather (though I know nothing interesting about the weather).

The WebRTC Course AMA is free to attend. It will be part webinar, part Q&A, but mostly fun.

Philipp – you are hereby cordially invited to join as well

Register to the WebRTC Course AMA – and even write down your questions on the event’s page right now – no need to wait until the 19th for that!

#3 – A few more launch bonuses

For those who end up enrolling early, I’ll have a few additional launch bonuses, but that’s for later.

–

On a personal note, today is Passover here in Israel.  If I seemed somewhat “off” in the past couple of days (or will seem like that in the coming days), then it probably has to do with me eating too much food and spending some time with my family.

 

The post My Advanced WebRTC Architecture Course is back with an AMA appeared first on BlogGeek.me.

Today, Apr 11, 2017,  maintenance work is scheduled to be done on the mailing lists server, including the shifting from lists.sip-router.org to lists.kamailio.org as primary domain.It is expected to not have any relevant downtime, but one never knows. Maybe the archive won’t be available for short time due to changes needs to be done to the web server after mailing lists server is upgraded.

Getting closer and closer to the Kamailio World Conference 2017 – the 5th edition is just four weeks away!The schedule has been published, with some adjustments still expected to happen. The event starts like the past edition with a half a day of technical workshops, followed by two full conference days.Thanks a consistent group of speakers the agenda is filled with topics that cover many of the interesting aspects of real time communications, from security and scalability to WebRTC and VoLTE, touching Kamailio and other open source projects like Asterisk or FreeSwitch. We will enjoy again the two sessions that never missed a Kamailio World edition: VUC Visions (open discussions panel) and Dangerous Demos (demo your crazy RTC idea in less than 5 minute and win one of the prizes in the game).More details can be found on the website of the event:

• https://www.kamailioworld.com

We expect to fill again the capacity of the conference room, if you haven’t registered yet and plan to attend, do it as soon as possible to secure your seat!Many thanks to our sponsors that made possible this event: FhG Fokus, Asipto, Flowroute, Telnyx, Sipwise, Sipgate, Obihai, Simwood, Evariste Systems, NG Voice, Digium, VoiceTel, Core Network Dynamics, Pascom.Thank you for flying Kamailio and looking forward to meeting many of you at Kamailio World 2017!