New UI, assets and better WebRTC analytics

Earlier this week we’ve started rolling out our new version of testRTC to our customers. This was one of these releases that we’ve worked on for a long time, starting two or three releases back when a decision was made that enough technical debt has been accumulating and a refresh was needed. It started as a nicely sized Angular to React rewrite and “redesign” which ended up being a lot more than that.

The results? Something that I am really proud of.

The switch to React included a switch to Highcharts as well, so we can offer better graphs moving forward. This isn’t why I wanted to write about this release though.

If you’ve been using testRTC already, you will be quite comfortable with this new release. It will feel like an improvement, while keeping everything you wanted and were used to using in the same place.

There are four things we’ve added that you should really care about:

#1 – Assets

This is something we are asked for quite some time now. We have clients who are running multiple tests and multiple monitors.

In some cases, the different scripts have only slight variations in them. In others, they share common generic tasks, such as login procedures.

The problem was that we were allowing customers to create a single file script only, and run it as a fully contained “program”. This kept our solution simple and elegant, but not flexible enough for growth.

This is why we are introducing Assets into the mix.

You can now create asset files which are simple scripts. Once created, you can include them into any of your running test scripts. You do that by simply adding an .include(‘<asset-name>’) command into your test script.

#2 – Advanced WebRTC Analytics

We’ve kinda revamped the whole Advanced WebRTC Analytics screen.

Up until now, it was a braindump of all getstats statistics without much thought. It gave power to the people, but it took its toll.

This time around, we’ve sat down to decide what information we have available, looked at what others are doing, and ended up with our own interpretation of what’s needed:

The Advanced WebRTC Analytics section now includes the following capabilities:

• Splits information into peer connection for easy view
• Shows getUserMedia constraints
• Show the PeerConnection configuration, so it is now super easy to see what STUN and TURN servers were configured
• Show cipher information for the security conscious
• Show ICE state machine progress, correlating it with the events log
• Show ICE negotiation table, to pinpoint on failure reasons (and understand what candidate pair got selected)
• Show WebRTC API events log, with the detailed calls and callbacks
• Show the actual graphs, just nicer, with Highcharts

I’ve been using these new capabilities just last week to explain to a new lead why his calls don’t connect with our probe’s firewall configuration.

#3 – Media scores everywhere

We’ve added media scores to our test results in the last release, but we placed them only on the test results page itself.

Now we’re taking the next step, putting the scores in monitor lists and test run lists. This means they are more accessible to you and can be seen everywhere.

What can you do with them?

1. Quickly understand if your service degrades when you scale
   1. Run the smallest test possible. See the media score you get
   2. Start scaling the test up. Expect the media score to not drop. If it does, check why
2. Make sure monitors are stable
   1. Run a monitor
   2. Check if the media score changes over it
   3. If it changes too much, you have an infrastructure problem

#4 – Client performance data

Another thing we’ve had for quite some time, but now decided to move front and center

There’s now a new tab in the test results of a single probe called “Performance”:

When opened, if you have the #perf directive in your run options, it will show you the probe’s machine performance – the CPU, memory and network use of the probe and browser.

This will give you some understanding of what user machines are going to be “feeling”, especially if you are aiming for a UI-heavy implementation.

We see customers using this for performance and stress testing.

Other

Other improvements that made it into this release?

• Filtering webhooks to run only on failed test runs
• Automating dynamic allocation of probes when no static ones are available
• Export test run history
• Ability to execute and collect traceroute on DNS lookups in the browser
• Added support to run longer tests
• Modified fields in most tables to make them more effective to users

Check it out 🙂

Testing Firefox has just become easier (and other additions in testRTC)

We’ve pushed a new release for our testRTC service last month. This one has a lot of small polishes along with one large addition – support for Firefox.

I’d like to list some of the things you’ll be able to find in this new release.

Firefox

When we set out to build testRTC, we knew we will need to support multiple browsers. We started off with Chrome (just like most companies building applications with WebRTC), and somehow drilled down into more features, beefing up our execution, automation and analysis capabilities.

We tried adding Firefox about two years ago (and failed). This time, we’re taking it in “baby steps”. This first release of Firefox brings with it solid audio support with rudimentary video support. We aren’t pushing our own video content but rather generating it ad-hoc. This results less effective bitrates that we can reach.

The challenge with Firefox lies in the fact that it has no fake media support the same way Chrome does – there is no simple way to have it take up media files directly instead of the camera. We could theoretically create virtual camera drivers and work our way from there, but that’s exactly where we decided to stop. We wanted to ship something usable before making this a bigger adventure (which was our mistake in the past).

Where will you find Firefox? In the profile planning section under the test editor:

When you run the tests, you might notice that we alternate the colors of the video instead of pushing real video into it. Here’s how it looks like running Jitsi between Firefox and Chrome:

That’s a screenshot we’ve taken inside the test. That cyan color is what we push as the video source from Firefox. This will be improved over time.

On the audio side you can see the metrics properly:

If you need Firefox, then you can now start using testRTC to automate your WebRTC testing on Firefox.

How we count minutes

Up until now, our per minute pricing for tests was built around the notion of a minimum length per test of 10 minutes. If you wanted a test with 4 probes (that’s 4 browsers) concurrently, we calculated it as 4*10=40 minutes even if the test duration was only 3 minutes.

That has now changed. We are calculating the length of tests without any specific minimum. The only things we are doing is:

1. Length is rounded up towards the nearest minute. If you had a test that is 2:30 minutes long, we count it as 3 minutes
2. We add to the test length our overhead of initiation for the test and teardown. Teardown includes uploading results to our servers and analyzing them. It doesn’t add much for smaller tests, but it can add a few minutes on the larger tests

End result? You can run more tests with the minutes allotted to your account.

This change is automatic across all our existing customers – there’s nothing you need to do to get it.

Monitoring tweaks

We’ve added two new capabilities to monitoring, due to requests of our customers.

#1 – Automated run counter

At times, you’ll want to alternate information you use in a test based on when it gets running.

One example is using multiple users to login to a service. If you run a high frequency monitor, which executes a test every 2-5 minutes, using the same user won’t be the right thing to do:

• You might end up not leaving the first session when running the next monitor a couple of minutes later
• Your service might leave session information for longer (webinars tend to do that, waiting for the instructors to join the same session for ten or more minutes after he leaves)
• If a monitor fails, it might cause a transient state for that user until some internal timeout

For these, we tend to suggest clients to use multiple users and alternate between them as they run the monitors.

Another example is when you want in each round of execution to touch a different part of your infrastructure – alternating across your data centers, machines, etc.

Up until today, we’ve used to do this using Firebase as an external database source that knows which user was last used – we even have that in our knowledge base.

While it works well, our purpose is to make the scripts you write shorter and easier to maintain, so we added a new (and simple) environment variable to our tests called RTC_RUN_COUNT. The only thing it does is return the value of an iterator indicating how many times the test has been executed – either as a test or as a monitor.

It is now easy to use by calculating the modulu value of RTC_RUN_COUNT with the number of users you created.

You can learn more about RTC_RUN_COUNT and our other environment variables in our knowledge base.

#2 – Additional information

We had a customer recently who wanted to know within every run of a monitor specific parameters of that run – in his case, it was the part of his infrastructure that gets used during the execution.

He could have used rtcInfo(), but then he’ll need to dig into the logs to find that information, which would take him too long. He needed that when the monitors are running in order to quickly pinpoint the source of failures on his end.

We listened, and added a new script command – rtcSetAdditionalInfo(). Whatever you place in that command during runtime gets stored and “bubbled up” – to the top of test run results pages as well as to the test results webhook. This means that if you connect the monitor to your own monitoring dashboards for the service, you can insert that specific information there, making it easily accessible to your DevOps teams.

Onwards

We will be looking for bugs (and fixing them) around our Firefox implementation, and we’re already hard at work on a totally new product and on some great new analysis features for our test results views.

If you are looking for a solid, managed testing and monitoring solution for your WebRTC application, then try us out.

Monitoring WebRTC apps just got a lot more powerful

As we head into 2019, I noticed that we haven’t published much around here. We doubled down on helping our customers (and doing some case studies with them) and on polishing our service.

In the recent round of updates, we added 3 very powerful capabilities to testRTC that can be used in both monitoring and testing, but make a lot of sense for our monitoring customers. How do I know that? Because the requests for these features came from our customers.

Here’s what got added in this round:

1. HAR files support

HAR stands for HTTP Archive. It is a file format that browsers and certain viewer apps support. When your web application gets loaded by a browser, all network activity gets logged by the browser and can be collected by a HAR file that can later be retrieved and viewed.

Our focus has always been WebRTC, so collecting network traffic information that isn’t directly WebRTC wasn’t on our minds. This changed once customers approached us asking for assistance with sporadic failures that were hard to reproduce and hard to debug.

In one case, a customer knew there’s a 502 failure due to the failure screenshot we generate, but it wasn’t that easy to know which of his servers and services was the one causing it. Since the failure is sporadic and isn’t consistent, he couldn’t get to the bottom of it. By using the HAR files we can collect in his monitor, the moment this happens again, he will have all the network traces for that 502, making it easier to catch.

Here’s how to enable it on your tests/monitors:

Go to the test editor, and add to the run options the term #har-file

Once there and the test/monitor runs next, it will create a new file that can be found under the Logs tab of the test results for each probe:

We don’t handle visualization for HAR files for the moment, but you can download the file and place it on a visual tool.

I use netlog-viewer.

Here’s what I got for appr.tc:

2. Retry mechanism

There are times when tests just fail with no good reason. This is doubly true for automating web UI, where minor time differences may cause problems or when user behavior is just different than an automated machine. A good example is a person who couldn’t login – usually, he will simply retry.

When running a monitor, you don’t want these nagging failures to bog you down. What you are most interested in isn’t bug squashing (at least not everyone) it is uptime and quality of service. Towards that goal, we’ve added another run option – #try

If you add this run option to your monitor, with a number next to it, that monitor will retry the test a few more times before reporting a failure. #try:3 for example, will retry twice the same script before reporting a failure.

What you’ll get in your monitor might be something similar to this:

The test reports a success, and the reason indicates a few times where it got retried.

3. Scoring of monitor runs

We’ve started to add a scoring system to our tests. This feature is still open only to select customers (want to join in on the fun? Contact us)

This scoring system places a test based on its media metrics collected on a scale of 0-10. We decided not to go for the traditional MOS scoring of 1-5 because of various reasons:

1. MOS scoring is usually done for voice, and we want to score video
2. We score the whole tests and not only a single channel
3. MOS is rather subjective, and while we are too, we didn’t want to get into the conversation of “is 3.2 a good result or a bad result?”

The idea behind our scores is not to look at the value as good or bad (we can’t tell either) but rather look at the difference between the value across probes or across runs.

Two examples of where it is useful:

1. You want to run a large stress test. Baseline it with 1-2 probes. See the score value. Now run with 100 or 1000 probes. Check the score value. Did it drop?
2. You are running a monitor. Did today’s runs fair better than yesterday’s runs? Worse? The same?

What we did in this release was add the score value to the webhook. This means you can now run your monitors and collect the media quality scores we create and then trendline them in your own monitoring service – splunk, elastic search, datadog, whatever.

Here’s how the webhook looks like now:

The rank field in the webhook indicates the media score of this session. In this case, it is an AppRTC test that was forced to run on simulated 3G and poor 4G networks for the users.

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As with any release, a lot more got squeezed into the release. These are just the ones I wanted to share here this time.

If you are interested in a monitoring service that provides predictable synthetics WebRTC clients to run against your service, checking for uptime and quality – check us out.

Just Landed: Automated WebRTC Screen Sharing Testing in testRTC

Well… this week we had a bit of a rough start, but we’re here. We just updated our production version of testRTC with some really cool capabilities. The time was selected to fit with the vacation schedule of everyone in this hectic summer and also because of some nagging Node.js security patch.

As always, our new release comes with too many features to enumerate, but I do want to highlight something we’ve added recently because of a couple of customers that really really really wanted it.

Screen sharing.

Yap. You can now use testRTC to validate the screen sharing feature of your WebRTC application. And like everything else with testRTC, you can do it at scale.

This time, we’ve decided to take appear.in for a spin (without even hinting anything to Philipp Hancke, so we’ll see how this thing goes).

First, a demo. Here’s a screencast of how this works, if you’re into such a thing:

Testing WebRTC Screen Sharing

There are two things to do when you want to test WebRTC screen sharing using testRTC:

1. “Install” your WebRTC Chrome extension
2. Show something interesting

#1 – “Install” your WebRTC Chrome extension

There are a couple of things you’ll need to do in the run options of the test script if you want to use screen sharing.

This is all quite arcane, so just follow the instructions and you’ll be good to go in no time.

Here’s what we’ve placed in the run options for appear.in:

#chrome-cli:auto-select-desktop-capture-source=Entire screen,use-fake-ui-for-media-stream,enable-usermedia-screen-capturing #extension:https://s3-us-west-2.amazonaws.com/testrtc-extensions/appearin.tar.gz

The #chrome-cli thingy stands for parameters that get passed to Chrome during execution. We need these to get screen sharing to work and to make sure Chrome doesn’t pop up any nagging selection windows when the user wants to screen share (these kills any possibility of automation here). Which is why we set the following parameters:

• auto-select-desktop-capture-source=Entire screen – just to make sure the entire screen is automatically selected
• use-fake-ui-for-media-stream – just add it if you want this thing to work
• enable-usermedia-screen-capturing – just add it if you want this thing to work

The #extension bit is a new thing we just added in this release. It will tell testRTC to pre-install any Chrome extensions you wish on the browser prior to running your test script. And since screen sharing in Chrome requires an extension – this will allow you to do just that.

What we pass to #extension is the location of a .tar.gz file that holds the extension’s code.

Need to know how to obtain a .tar.gz file of your Chrome extension? Check out our Chrome extension extraction guide.

var videoURL = "https://www.youtube.com/tv#/watch?v=INLzqh7rZ-U";

client
   .click('.VideoToolbar-item--screenshare.jstest-screenshare-button')
   .pause(300)
   .rtcEvent('Screen Share ' + agentSession, 'global')
   .rtcScreenshot('screen share ')
   .execute("window.open('" + videoURL + "', '_blank')")
   .pause(5000)

   // Switch to the YouTube
   .windowHandles(function (result) {
       var newWindow;
       newWindow = result.value[2];
       this.switchWindow(newWindow);
   })
   .pause(60000);
   .windowHandles(function (result) {
       var newWindow;
       newWindow = result.value[1];
       this.switchWindow(newWindow);
   });

The video bitrate still runs at around 280 kbps, but it oscillates a lot more. BTW – I am using the mesh version of appear.in here with 4 participants, so it is going low on bitrate to accommodate for it.

The number of video packets per second on that incoming blue line goes down from around 40 to around 25. Probably due to the lower number of frames per second.

What else is new in testRTC?

Here’s a partial list of some new things you can do with testRTC

• Manual testing service
• Custom network profiles (more about it here)
• Machine performance collection and visualization
• Min/max bands on high level graphs
• Ignore browser warnings and errors
• Self service API key regeneration
• Show elapsed time on running tests
• More information in test runs on the actual script and run options used
• More information across different tables and data views

Want to check screen sharing at scale?

You can now use testRTC to automate your screen sharing tests. And the best part? If you’re doing broadcast or multiparty, you can now test these scales easily for screen sharing related issues as well.

If you need a hand in setting up screen sharing in our account, then give us a shout and we’ll be there for you.

Check out the enhancements we’ve made to testRTC

It has been a while since we released a version, so it is with great pleasure that I am writing this announcement.

Yes. Our latest release is now out in the wild. We’ve upgraded our service on Sunday, so it is about time we take you for a quick roundup of the changes we’ve made.

#1 – Support for projects and users

This one is long overdue. Up until today, if you signed up for testRTC, you had to share your credentials with whoever was on your team to work with him on the tests. This was impossible to work with, assuming you wanted QA, R&D and DevOps to share the account and work cooperatively with the tests and monitors that got logged inside testRTC.

So we did what we should have – we now support two modes of operation:

1. A user can be linked to multiple projects
   • So if your company is running multiple projects, you can now run them separately, having people focused on their own environment and tests
   • This is great for those who run segregated services for their own customers
   • It also means that now, a user can switch between projects with a single set of credentials in the system
2. A project can belong to multiple users
   • Need someone to work on writing the scripts and executing them? You got it
   • Have a developer working on a bug that got reported with a link to testRTC? Sure thing
   • The IT guy who just received a downtime alarm from the WebRTC monitor we run? That’s another user
   • Each user has his own place in the project, and each is distinguished by his own credentials

If you require multiple projects, or want to add more users to your account just contact our support.

#2 – Longer, bigger tests

While theoretically, testRTC can run any test at any length and size, things aren’t always that easy.

There are usually two limitations to these requirements:

1. The time they take to prepare, execute, run and collect results
2. The time it takes to analyze the results

We worked hard in this release on both elements and got to a point where we’re quite happy with the results.

If you need long tests, we can handle those. One of the main concerns with long tests is what to do if you made a mistake while configuring them? Now you can cancel such tests in the middle if necessary.

If you need to scale tests to a large number of browsers – we can do that too.

We are making sure we bubble up the essentials from the browsers, so you don’t have to work hard and rummage through hundreds of browser logs to find out what went wrong. To that end, the tables that show browser results have been reworked and are now sorted in a way that will show failures first.

#3 – Advanced WebRTC analysis

We’ve noticed in the past few months that some of our customers are rather hard core. They are technology savvy and know their way in WebRTC. For them, the graphs we offer of bitrates, latencies, packet losses, … – are just not enough.

Chrome’s webrtc-internals and getstats() offer a wealth of additional information that we offered up until now only in a JSON file download. Well… now we also visualize it upon request right from the report itself:

These graphs are reachable by clicking the webrtc_internals_dump.txt link under the Logs tab of a test result. Or by clicking the Advanced WebRTC Analytics button located just below the channels list:

I’d like to thank Fippo for the work he did (webrtc-dump-importer) – we adopted it for this feature.

#4 – Simulation of call drops and dynamic network changes

This is something we’ve been asked more than once. We have the capability of modeling the network of our probes, so that the browser runs with a specific configuration of a firewall or via a specific type of simulated network. We’re modifying and tweaking the profiles we have for these from time to time, but now we’ve added a script command so that you can change this configuring in runtime.

What can you do with it? Run two minutes of a test with 2 Mbps, then close virtually everything for 20-30 seconds, then open up  the network again – and see what happens. It is a way to test WebRTC in your application in dynamic network conditions – ones that may require ICE restarts.

In the test above, we dynamically changed the network profile in mid-call to starve WebRTC and see how it affects the test.

How do you use this new capability? Use our new command rtcSetNetworkProfile(). Read all about it in our knowledge base: rtcSetNetworkProfile()

#5 – Additional test expectations

We had the basics covered when it came to expectations. You could check the number and types of channels, validate that there’s some bits going on in there, validate packet loss. And that’s about it.

To this list of capabilities that existed in rtcSetTestExpectations() we’ve now added the ability to add expectations related to jitter, video resolutions, frame rate, and call setup time. We’ve also taken the time to handle expectations on empty channels a lot better.

There’s really nothing new here, besides an enhancement of what rtcSetTestExpectations() can do.

#6 – Additional information in Webhook responses

testRTC can notify your backend whenever a test or a monitor run ends on the status of that run – success or failure. This is done by configuring a webhook that is called at the end of the test run. We’ve had customers use it to collect the results to their own internal monitoring systems such as Splunk and Elastic Search.

What we had on offer in the actual payload that was passed with the webhook was rather thin, and while we’re still trying to keep it simple, we did add the leading error in that response in cases of failure:

#7 – API enabled to all customers

Yes. We had APIs in the past, but somehow, there was friction involved, with customers needing to ask for their API key in order to use the API for their continuous integration plans. It worked well, but the number of customers asking for API keys – both customers and prospects under evaluation – has risen to a point where it was ridiculous to continue doing this manually. Especially when our intent is for customers to use our APIs.

So we took this one step forward. From now on, every account has an API key by default. That API key is accessible from the account’s dashboard when you login, so there’s no need to ask for it any longer.

For those of you who have been using it – note that we’ve also reset your key to a new value.

Your turn

This has been quite a big release for us, and I am sure to miss an enhancement or two (or more).

Now back to you. How would you want to test WebRTC in your product?

Introducing: Our Brand New Dashboard

We’ve been working hard these past two months, ever since we got our previous release out the door. This time, we invested a lot of time and thought on the small items. And one big item as well.

All over the service, you’ll notice some slight changes to the UI. This is an ongoing process to fine-tune the service and make it simpler to use for our customers.

The biggest visible addition to our latest release is the introduction of a new user dashboard.

From now one, when a user logs in, he gets a bird’s eye view of his activities in testRTC:

What can you see on the dashboard?

Usage

This area of the dashboard highlights the usage done in the account.

It allows you to understand what resources are available to you, so if you want to run a stress test, you will be able to use enough browsers.

If you want to do ad-hoc testing with more browsers than are available in your account, you’ll need to holler us and we’re enable more browsers on your account for a period of time.

Stats

This area shows the statistics of your use over a span of time. It is quite useful for managers to understand how many tests were conducted and know how they fared.

• In red, we indicate tests and monitor executions that failed for the period selected
• In green, we indicate tests and monitor executions that succeeded for the period selected
• In blue, we indicate the total number of tests and monitor executions for the period selected

And you can select a different period to look at.

Active Monitors

This area indicate what monitors are up and running at the moment, along with the status of the most recent execution.

If you click on any of the rows, it will get you to the monitor run results, filtered for that specific monitor.

Recent Tests

This area shows the last 5 tests that got executed, along with their results.

As with the active monitors, clicking on the test gets you to the results themselves.

News and Announcements

This area shows some news and announcements we have for our users.

What’s Next?

Consider the dashboard a work in progress. We’re sure there’s much to be improved here. We wanted to get this out the door and into the hands of our users. Ping us if you have any suggestions on how to improve it.

If you need to test or monitor a WebRTC product – don’t be shy – sign up for testRTC.

WebRTC Test Automation and where it fits in your roadmap

I see mixed signals about the popularity and acceptance of test automation. It is doubly so when testing WebRTC.

Time to consider some serious WebRTC test automation.

In favor of automation

A tester automated his job for 6 years – most probably a hoax, but one that rings partially true. The moral of the story is simple – if you invest time in automating rudimentary tasks – you get your ROI back tenfold in the future.

That’s… about it.

We have customers who use us to automate areas of their testing, but not many. At least not as many as I’d expect there to be – WebRTC being new and all – and us looking at best practices and changing our bad ways and habits of the past when stating with green field projects.

Against automation

Why is Manual QA Still So Prevalent? – it seems like SauceLabs, who delve into general purpose browser automation, is also experiencing the same thing. Having companies focus on manual testing instead of moving to automation.

Best explanation I heard from someone? They can get a cheap tester to do the work for them by outsourcing it to a developing country and then it costs them less to do the same – just with humans.

For me, that’s taking Amazon’s Mechanical Turk a step too much. For a repetitive task that you’re going to do in each and every release (yours and of browser vendors), to have different nameless faces (or even named ones) do the same tasks over and over again?

Dog-fooding at testRTC

We’ve been around for almost 2 years now. So it is high time we start automating our own testing as well.

The first place where we will be automating our own testing is in making sure our test related feature set works:

• Our special script commands and variables
• Running common test scenarios that our customers use in WebRTC

Now, we have test scripts that run these tests, so we can automate them individually. Next step would be to run them sequentially with a “click of a button”. Or more accurately, an execution of a shell script. Which is where we’re taking this in our next release.

The rest will stay manual for now. Mostly because in each version we change our UI based on the feedback we receive. One of our top priorities is to make our product stupidly simple – so that our customers can focus on their own product and need to learn as little as possible (or nothing at all) to use testRTC.

Why our customers end up automating?

There are several huge benefits in automating at least parts of your testing. Here are the ones we see every day from the way our customers make use of WebRTC:

• Doing the most basic sanity tests – answering the question “is it broken?” and getting an answer fast with no human intervention. This is usually coupled with continuous integration, where every night the latest build is tested against it
• Scale tests – when a service needs to grow, be it to 10 users in the same session, 40 people across 20 1:1 sessions or 100 viewers of a webinar – it becomes hard to manually test. So they end up writing a simple script in our platform and running it on demand when the time comes to stress test their product
• Network configurations – taking a script and running it in various network conditions – with and without forcing TURN, packet losses, etc. Some also add different data center locations for the browsers and play with the browser versions used. The idea is to get testing to the edge cases where a user’s configuration is what comes back to bite you
• Debugging performance – similar to scale tests, but slightly different. Some require the ability to check the capacity of a given machine in their product. Usually the media server. There’s much to be said about that, but being able to run a large scale test, analyze the performance report testRTC produces, and then rinse and repeat means it is easier to find the bottlenecks in the system and fix them prior to deployment

Starting out with WebRTC, we’ve seen other things getting higher priority by customers. They all talk about scenarios and coverage of their test plans. Most don’t go there due to that initial high investment.

What we do see, and what effectively improves our customer’s product, is taking one scenario. Usually a simple one. Writing it in a way that allows for scaling it up. Once a customer runs it for a few days, he sees areas he needs to improve in his product, and how that simple script can expand to encompass more of his testing needs.

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This is also why we try to be there with our customers every step of the way. From assisting in defining that test, to writing it and following through with analysis if need be.

Are you serious about your WebRTC product? Don’t waste your time and try us out.