Category Archives for "How to"


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 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

#chrome-cli:auto-select-desktop-capture-source=Entire screen,use-fake-ui-for-media-stream,enable-usermedia-screen-capturing #extension:

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.

Now that we’ve got everything enabled, we can focus on the part of running a test that uses screen sharing.

#2 – Show something interesting

Screen sharing requires something interesting on the screen, preferably not an infinite video recursion of the screen being shared in one of the rectangles. Here’s what you want to avoid:

And this is what we really want to see instead:

The above is a screenshot that got captured by testRTC in a test scenario.

You can see here 4 participants where the top right one is screen sharing coming from one of the other participants.

How did we achieve this in the code?

Here are the code snippets we used in the script to get there:

var videoURL = "";

   .rtcEvent('Screen Share ' + agentSession, 'global')
   .rtcScreenshot('screen share ')
   .execute("'" + videoURL + "', '_blank')")

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

We start by selecting the URL that will show some movement on the screen. In our case, an arbitrary YouTube video link.

Once we activate screen sharing in, we call rtcEvent which we’ve seen last time (and is also a new trick in this new release). This will add a vertical line on the resulting graphs so we know when we activated screen sharing (more on this one later).

We call execute to open up a new tab with our YouTube link. I decided to use the URL to get the video to work close to full screen.

Then we switch to the YouTube in the first windowHandles call.

We pause for a minute, and then go back to the tab in the browser.

Let’s analyze the results – shall we?

Reading WebRTC screen sharing stats

Screen sharing is similar to a regular video channel. But it may vary in resolution, frame rate or bitrate.

Here’s how the graphs look like on one of the receiving browsers in this test run. Let’s start with the frame rate this time:

Two things you want to watch for here:

  1. The vertical green line – that’s where we’ve added the rtcEvent call. While it was added to the browser who is sending screen sharing, we can see it on one of the receiving browsers as well. It gets us focused on the things of interest in this test
  2. The incoming blue line. It starts off nicely, oscillating at 25-30 frames per second, but once screen sharing kicks in – it drops to 2-4 frames per second – which is to be expected in most scenarios

The interesting part? made a decision to use the same video channel to send screen sharing. They don’t open an additional video channel or an additional peer connection to send screen sharing, preferring to repurpose an existing one (not all services behave like that).

Now let’s look at the video bitrate and number of packets graphs:

The video bitrate still runs at around 280 kbps, but it oscillates a lot more. BTW – I am using the mesh version of 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.


Cross Platform WebRTC Browser Testing: Chrome, Firefox, Edge & Safari

This is a guest post by Philipp Hancke. He was kind enough to share the work he’s done already in automating his own WebRTC Safari tests.

Now that Apple added WebRTC to its Safari browser, it is time to ask –

How do you test WebRTC four browsers on different operating systems? Using Selenium Grid it is surprisingly easy, learn how.

Webkit recently joined the WebRTC ship which means we now have 5 major browsers to test:

  • Chrome, any operating system
  • Firefox, any operating system
  • Microsoft Edge, Windows 10
  • Safari (Technology Preview), OSX

Until a few days ago it was feasible to run tests all WebRTC supported browsers only on a single operating system (Getting Chrome, Firefox and Edge tested on Windows 10). With Safari, that has changed. Sounds like a big problem, no?

This kind of testing is typically done using Selenium which is used to remotely control the browser. And if it is remotely controlled it can be done with selenium’s grid functionality. Which is surprisingly easy.

Say we want to test Edge and Safari from a Windows machine. First step is to download the selenium standalone jar file. I am still sticking to version 3.3.1. Next, set up a selenium grid hub on one machine by running:

java -jar selenium-server-standalone-3.3.1.jar -role hub

This will start a hub which is where you connect your selenium nodes as well as test clients to. It will also tell you the URL that you need to connect your nodes to.

Next, start a node on your OSX machine by running:

java -jar selenium-server-standalone-3.3.1.jar -role node -hub http://where-your-hub-tells-you-it-is-running -browser “browserName=safari”

This will start a node that will run your Safari instance.

Now, in the code running selenium you need to instruct it to do two things:

Next run your test code. If it works out of the box, great. If it does not you will probably spend a lot of time figuring out the issues. The most common issue here is forgetting to include the respective WebDriver binary in a path where it is found.


The Missing chrome://webrtc-internals Documentation

There’s a wealth of information tucked into the chrome://webrtc-internals tab, but there was up until recently very little documentation about it. So we set out to solve that, and with the assistance of Philipp Hancke wrote a series of articles on what you can find in webrtc-internals and how to make use of it.

The result? The most up to date (and complete) webrtc-internals documentation.

To make sure it doesn’t get lost, here are the links to the various articles:

  1. webrtc-internals and getstats parameters – a detailed view of webrtc-internals and the getstats() parameters it collects
  2. active connections in webrtc-internals – an explanation of how to find the active connection in webrtc-internals – and how to wrap back from there to find the ICE candidates of the active connection
  3. webrtc-internals API trace – a guide on what to expect in the API trace for a successful WebRTC session, along with some typical failure cases



Your best WebRTC debugging buddy? The webrtc-internals API trace

This time, we take you through the webrtc-internals API trace to see what can you learn from it.

To make this article as accurate as possible, I decided to go to my source of truth for the low level stuff related to WebRTC – Philipp Hancke, also known as fippo or hcornflower. This in a way, is a joint article we’ve put together.

Before we do, though, you should probably check out the other articles in this series:

  1. Parameter’s meaning in webrtc-internals
  2. Finding the current active connection in webrtc-internals

Now back to the API trace.

WebRTC is asynchronous

Here’s something you probably already noticed. WebRTC is asynchronous to the extreme. It almost painstakingly makes sure that whatever you are trying to achieve – you won’t be able to without multiple calls in different contexts of your JavaScript app in the browser.

It isn’t because the authors of WebRTC are mean. It is because the nature of communications is asynchronous. It is made worse by the various network topologies that require the use of curses like STUN, TURN and ICE and by the fact that we require the user to authorize things like accessing his camera.

This brings us to the tricky situation of error handling. With WebRTC, it takes place everywhere. Anything you do can fail twice:

  1. When you call the API and it returns
  2. When the callback/promise/event handler/whatever returns back with the result of your API call

This means that in many cases, you are going to be left with a half baked solution that looks at some of the error cases (did you ever see a sample that takes care of edge cases or failure scenarios?).

It also means that often times you’ll need to be able to debug them. And that’s what the API trace in webrtc-internals can help you with.

The webrtc-internals API trace

If you open chrome://webrtc-internals while in an active WebRTC session, you will immediately see the API trace:

WebRTC API trace sample

This is the list of API calls and events done on the peer connection, informing you of the progress and state of the connection.

You can click on any of these APIs to see its parameters.

WebRTC API trace click to expand

Before we look at what kind of analysis we can derive from these traces, let’s look at what some of the connection methods and events do.

    • addStream: if this method was called, the Javascript code has added a MediaStream to the peerconnection. You can see the id of the stream as well as the audio and video tracks. onAddStream shows a remote stream being added, including the audio and video track ids, it is called between the setRemoteDescription call and the setRemoteDescriptionOnSuccess callback
    • createOffer shows any calls to this API including the options such as offerToReceiveAudio, offerToReceiveVideo or iceRestart. createOfferOnSuccess shows the results of the createOffer call, including the type (which should be ‘offer’ obviously) and the SDP resulting from it. createOfferOnFailure could also be called indicating an error but that is quite rare
    • createAnswer and createAnswerOnSuccess and createAnswerOnFailure are similar but with no additional options
    • setLocalDescription shows you the type and SDP used in the setLocalDescription call. If you do any SDP munging between createOffer and setLocaldescription you will see this here. This results in either a setLocalDescriptionOnSuccess or setLocalDescriptionOnFailure callback which shows any errors. The same applies to the setRemoteDescription and its callbacks, setRemoteDescriptionOnSuccess and setRemoteDescriptionOnFailure
    • onRenegotiationNeeded is the old chrome-internal name for the onnegotiationneeded event. If your app uses this you might want to look for it
    • onSignalingStateChange shows the changes in the signaling state as a result of calls to setLocalDescription and setRemoteDescription. See the wonderful diagram in the specification for the gory details. At the end of the day, you will want to be in the stable state most of the time
    • iceGatheringStateChange is the little brother of the ice connection state. It will show you the state of the ice gatherer. It will change to gathering after setLocalDescription if there are ICE candidates to gather
    • onnicecandidate events show all candidates gathered, with information for which m-line and MID. Likewise, the addIceCandidate method shows that information from the other side. Typically you should see both event types. See below for a more detailed discussion of these events
    • oniceconnectionstate is one of the most important event handlers. It tells you whether a peer-to-peer connection succeeded or not. From here, you can start searching for the active candidate as we explained in the previous post

The two basic flows we can see are that of something offering connections and answering. The offerer case will typically consist of these events:

WebRTC API Trace - offer side

  • (addStream if the local side wants to send media)
  • createOffer
  • createOfferOnSuccess
  • setLocalDescription
  • setLocalDescriptionOnSuccess
  • setRemoteDescription
  • (onaddstream if the remote end signalled media streams in the SDP)
  • setRemoteDescriptionOnSuccess

While the answerer case will have:

WebRTC API Trace - answer side

  • setRemoteDescription
  • (onaddstream if the remote end signalled media streams in the SDP)
  • createAnswer
  • createAnswerOnSuccess
  • setLocalDescription
  • setLocalDescriptionOnSuccess

In both cases there should be a number of onicecandidate events and addIceCandidate calls along with signaling and ice connection state changes.

Let us look at two specific cases next.

Example #1 – My WebRTC app works locally but not on a different network!

This is actually one of the most frequent questions on the discuss-webrtc list or on stackoverflow. Most of the time the answer is “you need a TURN server” and “no, you can not use some TURN server credentials that you found somewhere on the internet”.

So it works locally. That means that you are creating an offer, sending it to the remote side, calling setLocalDescription() and are getting an answer that you feed into setRemoteDescription(). It also means that you are getting candidates in the onicecandidate() event, sending them to the remote side and getting candidates from there which you call the addIceCandidate() method with.

And locally you get a oniceconnectionstatechange() event to connected or completed:

WebRTC API trace - ICE state changes

Great! You probably just copied and pasted these pieces of code from somewhere on github.

Now… why does it not work when you’re on a different network? On different networks, you need both a STUN and a TURN server. Check if your app is using a STUN and TURN server and that you’re passing them correctly at the top of webrtc-internals:

NAT configuration in webrtc-internals

As you can see (assuming you have good eyes), there are a number of ice servers used here. In the case of our screenshot, it’s Google’s apprtc sample. There is a stun server, There are also four TURN servers:

  1. turn:
  2. turn:[2A00:1450:4010:C01::7F]:19305?transport=udp
  3. turn:
  4. turn:[2A00:1450:4010:C01::7F]:443?transport=tcp

As you can see, apprtc uses TURN over both UDP and TCP and is running TURN servers for both IPv4 and IPv6.

Now just because you configured a TURN server does not mean there won’t be any errors. The TURN server might not be reachable. Or your credentials might not work (this will happen if you “found” the credentials on a list of “free public servers”). In order to verify that the STUN and TURN servers you use actually work you need to look at the onicecandidate() events.

If you use a STUN or a TURN server, you should see a onicecandidate() event with a candidate that has a ‘typ srflx’.

Similarly, if you use a TURN server, you need to check if you get an onicecandidate() event where the candidate has a ‘typ relay’.

Note that Chrome stops gathering candidates once it establishes a connection. But if your connection gets established you are probably not going to debug this issue.

If you get both of these you’re fine. But you also need to check what candidates your peer sent you with which addIceCandidate() was called.

Example #2 – The network is blocking my connection

Networks that block UDP traffic are quite common. TURN/TCP and TURN/TLS (as well as ICE-TCP even though we mention this mostly to make Emil Ivov happy) provide a way to enable calls even on those networks. This has some effect on the media quality as we discussed previously but let us see how we can detect whether we are on a network that is blocking UDP traffic to begin with.

If you want to follow along, open webrtc-internals and the webrtc candidate gathering demo page and start gathering. By default, it uses one of Google’s STUN servers. To keep things simple, uncheck the “gather IPv6 candidates” and “gather RTCP candidates” boxes before clicking on the “gather candidates” button:

Uncheck ICE gathering candidates

On webrtc-internals you will see a createOffer call with offerToReceiveAudio set to true (this is to create an m-line and gather candidates for it):

WebRTC ICE gathering receive audio

Followed by a createOfferOnSuccess and a setLocalDescription call. After that there will be a couple of onicecandidate events and an icegatheringstatechange to completed, followed by a stop call.

There should be an onicecandidate with a candidate that has a “typ srflx” in it:

ICE candidate types

It shows your public ip. If you don’t get such a candidate but only host candidates, either the STUN server is not working (which in the case of Google’s STUN server is somewhat unlikely) or your network is blocking UDP.

Now block UDP on your network (but mind you, do not block port 53 for DNS). If you don’t know a quick way to block UDP, lets try to simulate that by changing the stun server to something that will not respond, in this case Google’s well-known DNS server running at

Fudging STUN configuration for WebRTC

Click “gather candidates” again. After around 10 seconds you will see a gathering state change to completed in webrtc-internals. But you will not see a server-reflexive candidate:

ICE negotiation with no STUN connectivity

You can try the same thing with a TURN UDP server. Make sure your credentials are valid (again, the “public TURN server list” is not a thing). This will show both a srflx and a relay candidate.

One of the nice tricks you can do here is to change the password to something invalid. Then you will only get a srflx but no relay candidate. Which is a nice and easy way to detect if your credentials are invalid — the candidates page even suggests this.

You can repeat this with TURN/TCP and TURN/TLS servers. You can even add all kinds of TURN servers and then use the priority trick we have shown in the last blog post to figure out from which servers you gathered candidates.

If you don’t get anything but host candidates you might be on a network which blocks both UDP traffic and is successful at blocking TURN/TCP and TURN/TLS. One scenario where that might happen currently is if there is a proxy that requires authentication which is not yet supported by Chrome.

Now let us take a step back. When is this useful? In a real-world scenario you will want to run with all kinds of STUN and TURN servers, otherwise you will get high failure rates. If you need to debug a failure to establish a connection, you should look for the onicecandidate and addIceCandidate events. They will allow you to figure out if the local or remote client was on a network that blocked it from establishing a connection to any peer outside the network.

What’s next?

So this time around, we’ve focused on the API traces:

  • We’ve acquainted with the fact that this is something that webrtc-internals does us a great service just by capturing all of these WebRTC API calls
  • We even went through the typical API calls and flows that are expected to appear in the WebRTC API trace
  • We’ve looked at two examples where the WebRTC API trace can help us debug the problems we’re seeing (there are more)
    • #1 – misconfiguration of NAT traversal servers
    • #2 – network blocking and the forgotten TURN/TCP configuration

We’re not done yet with this series. We still have one or more articles in the pipeline to close the basics of what webrtc-internals got up its sleeves.

If you are interested in keeping up with us, you might want to consider subscribing.

Huge thanks for Fippo in assisting with this series!


How do you find the current active connection in webrtc-internals?

Today, we want to help you find the current active connection in webrtc-internals.

To make this article as accurate as possible, I decided to go to my source of truth for the low level stuff related to WebRTC – Philipp Hancke, also known as fippo or hcornflower. This in a way, is a joint article we’ve put together.

Before we do, though, make sure that you know what the parameters in chrome://webrtc-internals mean.  If you don’t, then you should start with our previous article, which explains the meaning of the parameters in webrtc-internals (and getstats).

First things first:

Why is this of any importance?

While you can always go and check the statistics on your connection and channels, there are times when what you really want to do is understand HOW you got connected.

WebRTC allows connections to take one of 3 main routes:

  1. Direct P2P
  2. By making use of STUN (to find a public IP address)
  3. By making use of TURN (to relay all of your media through it)

WebRTC connection types

Which one did you use for this connection? Did this call end up going via TURN over TCP?

There are two times when this will be useful to us:

  1. When trying to understand what occurred in this call, how the ICE negotiation took place, which candidate ended up being used, did we need to renegotiate or switch between candidates – we need to know how to find the active connection
  2. When we want to know this programmatically – especially if we are collecting these WebRTC stats and trying to ascertain the test results from it or how our network behaves in production with live users

What is an active connection in WebRTC?

WebRTC uses the ICE protocol (described in RFC 5245 if you are really keen on the details) to find a way to connect two peers which may be behind NAT routers. To establish a connection, each peer gathers a number of candidates.

There are host candidates which represent local ip addresses and ports. These have a “typ host” when you expand the onicecandidate or addIceCandidate events in webrtc-internals.


Similarly, there are serverreflexive candidates where the peer asked a STUN server for its public ip address. These will have a “typ srflx” and an additional raddr field that shows the local ip address where this originated.

Last but not least there are relay candidates. Here, the peer asked a TURN server to open a port and relay traffic. These will show up in the onicecandidate and addIceCandidate with a “typ relay”. To make things more difficult, WebRTC clients can use either UDP, TCP or TLS to connect to the TURN server.

Let’s assume that you see a number of onicecandidate and addIceCandidate calls in webrtc-internals. There are conditions where this does not happen but typically WebRTC will use a technique known as “trickle ice”.

Once candidates have been exchanged, the WebRTC engine forms pairs of local and remote candidates and starts sending STUN packets to check if it gets a response. While the details of this are very much hidden from the application we can observe what is going on in webrtc-internals (and the getStats() API).

TURN in wireshark

The screenshot above was taken from Wireshark, a tool that can be used to sniff the local network. It shows a TURN message flowing in a session during the connection setup stage.

How do we find the current active connection in webrtc-internals?

Open one of the WebRTC samples that establishes a local peer-to-peer connection, mute your speaker and start a call. Also open chrome://webrtc-internals. Now let’s see if we can find out together the active connection.

Now that we’re running the “local peer-to-peer connection” sample off the WebRTC samples repository, there’s something to remember – it does not use STUN and TURN servers, so the number of candidates exchanged will be quite small. We’ve selected it on purpose, so we won’t have so much clutter to go over. What you will see is just four candidates in the onicecandidate event of the first connection (because of some rules about rtcp-mux and BUNDLE) and a single candidate in the addIceCandidate event.

An active connection in webrtc-internals

Now which of those candidate pairs is used? Fortunately webrtc-internals gives up a hint by showing the active candidate pair in bold. Typically the id of this will be Conn-audio-1-0 (with BUNDLE, the same candidate pair and connection is used also for the video). Let’s expand this and see why it is marked as bold:

WebRTC active connection statistics

The most important thing here is the googActiveConnection attribute which is true for the currently active connection and false for all others. We see quite a number of attributes here and while some of them have been discussed in the previous post let us repeat what we see here:

  1. bytesReceived, bytesSent and packetsSent specify the number of bytes and packets sent over this connection. packetsReceived is still MIA but since it is not part of the specification that probably will not change anymore
  2. googReadable specifies whether STUN packets were received on this pair, googWritable whether responses were sent. These google properties have been replaced by the requestsReceived, responsesSent, requestsReceived and responsesSent standard counters
  3. googLocalAddress and googRemoteAddress show the local IP and port. This is quite useful if you need to correlate the information in webrtc-internals with a Wireshark dump. googLocalCandidateType and googRemoteCandidateType give you more information about whether this was a local, serverreflexive or relayed candidate
  4. localCandidateId and remoteCandidateId give you the names of the local and remote candidate statistics. These allow you to get even more details

Lets search for the local and remote candidates specified in the localCandidateId and remoteCandidateId fields. Again this is pretty easy since there is only a single candidate pair.

WebRTC connection candidates

Despite having a different type (localcandidate and remotecandidate; the specification changed this slightly to local-candidate and remote-candidate; we imagine there was a lengthy discussion about the merits of the hyphen), the information in both candidates is the same:

  • the ipAddress is where packets are sent from or sent to
  • the networkType allows you to figure out the type of the network interface
  • the portNumber gives you the port that packets are sent to or from
  • the transport specifies the transport protocol of the candidate as shown in the candidate line. Typically this will be udp unless ICE-TCP is used
  • the candidateType shows whether this is a host, srflx, prflx or relay candidate. See the specification for further details
  • the priority is the priority of the candidate as defined in RFC 5245

The most interesting field here is the priority. When the implementation following the definition of the priority from RFC 5245, the highest 8 bit specify the type preference (= you’ll need to shift the priority attribute right >> 24 bits). This is of particular interest since for local candidates with a candidateType relay we can figure out the transport used between the client and the TURN server. Chrome uses a type preference of 2 for relay candidates allocated over TURN/UDP, 1 for TURN/TCP and 0 for TURN/TLS. Calculating priorities this way means that relayed connections will have a lower priority than any non-relayed candidates and that TURN servers will only be used as a fallback. As we have described previously, using TURN/TCP (or TURN/TLS) changes the behaviour of the WebRTC quite drastically. If you find this hard to understand do not worry, the specification is currently updated to address this!

What happens during an ICE restart?

So far we have been using a very simple example. Let’s make things more complicated with one of Philipp’s favorite topics, ICE restarts. Navigate away from the previous sample and go to this sample. Make a call. You will see an active candidate pair on chrome://webrtc-internals. Click on the “Restart ICE” button. Note how the data on the candidate pair Conn-audio-1-0 has changed, the ip addresses have changed and the STUN counters have been reset. And the data we have seen previously is now (usually) on a report with the identifier Conn-audio-1-1. The candidate pair is not used any longer but will stick around in getStats (and consequently webrtc-internals).

This is somewhat unexpected (and lead to filing an issue). It seems like the underlying library is renaming the reports in a way such that the active candidate always has the id Conn-audio-1-0. It has behaved like this for ages but this has some disadvantages. Among other things, it is not safe to assume that the number of bytesSent will always increase as it could reset, similar to a bug resetting the bytesSent counters on the reports with a type of ssrc that we have seen in the last post.

What’s next?

What have we learned so far?

  1. The meaning of parameters in webrtc-internals and getstats
  2. Active connections in WebRTC
    1. We saw why knowing and understanding the active connection is important (mainly for debugging purposes)
    2. We then went to look at how we find an active connection (it is in bold, and it’s googActiveConnection attribute is set to true
    3. We also reviewed the interesting attributes we can find on an active connection.
    4. We then linked back the active connection to its candidate pair using the candidates id’s, and found out the candidate type from it
    5. We even learned how to find over what transport protocol a candidate is allocated on a TURN server

We’ve only just started exploring webrtc-internals, and our plan is to continue working on them to create a series of articles that will act as the knowledge base around webrtc-internals.

If you are interested in keeping up with us, you might want to consider subscribing.

Huge thanks for Fippo in assisting with this series!