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  1. WebRTC API
  2. WebRTC Connectivity
  3. WebRTC
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April 4, 2018

Exploring RTCRtpTransceiver.

Firefox now implements the RTCRtpTransceiver API, a.k.a. stage 3 in my blog post “The evolution of WebRTC 1.0.” from last year. Transceivers more accurately reflect the SDP-rooted network behaviors of an RTCPeerConnection. E.g. addTransceiver() (or addTrack) now creates a receiver at the same time, which correlates better with the bi-directional m-line that begins life in the SDP offer, rather than being something that arrives later once negotiation completes. This lets media arrive early on renegotiation.

The RTCDataChannel interface represents a network channel which can be used for bidirectional peer-to-peer transfers of arbitrary data. Every data channel is associated with an RTCPeerConnection, and each peer connection can have up to a theoretical maximum of 65,534 data channels (the actual limit may vary from browser to browser). Apr 04, 2018 Firefox now implements the RTCRtpTransceiver API, a.k.a. Stage 3 in my blog post “The evolution of WebRTC 1.0.” from last year. Transceivers more accurately reflect the SDP-rooted network behaviors of an RTCPeerConnection. AddTransceiver (or addTrack) now creates a receiver at the same time, which correlates better with the bi-directional m-line that begins life in.

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  1. Client-side WebRTC code samples. To test your webcam, microphone and speakers we need permission to use them, approve by selecting “Allow”.
  2. This article introduces the protocols on top of which the WebRTC API is built. ICE Interactive Connectivity Establishment (ICE) is a framework to allow your web browser to connect with peers.
  3. The ‘RTC’ in WebRTC is an abbreviation for Real-Time-Communication, and is used for voice calls, video chats, and p2p file sharing. WebRTC is not a bug as it was originally developed to facilitate the above mentioned types of connections between browsers independently without depending on any plug-ins.

But most WebRTC sites today are still written the old way, and there are subtle differences to trip over. Web developers are stuck in a bit of limbo, as other browsers haven’t caught up yet. In light of that, we should probably dive right into two immediate breaking spec changes you might have run into in Firefox 59:

  • Remote tracks are now muted and temporarily removed from their stream(s), rather than ended, in response to direction changes (e.g. from the poorly named removeTrack).
  • track.ids at the two end-points rarely correlate anymore.

Importantly, this affects you whether you use transceivers or not. addTrack (even addStream) is just a tweaked version of addTransceiver these days. This blog will look at how to handle this—tying up lose ends from last year’s blog post—and demonstrate how Firefox works.

The short answer to “Why no ended?”, is that incoming tracks now correspond to sender objects on the other side, which may resume in response to transceiver.direction changes.

The short answer to “Why no matching track IDs?”, is that incoming (receiver) tracks now usually exist ahead of connecting, their immutable ids unable to match up with the other side (and were never guaranteed to be unique anyway).

As you might notice, this boils down to changes to the lifetime of remote tracks. Gone is misleading symmetry between local and remote tracks, or the streams they belong to for that matter. That symmetry looked pretty, but got in the way of fully controlling tracks remotely.

Remote-controlled tracks.

A more useful analogy is that of transceiver.sender as a remote-control of the transceiver.receiver.track held by the other peer. Their lifetimes match that of the transceiver itself. Here’s how this remote-control works:

When we do this……the other side sees this
pc.addTransceiver(trackA, {streams})*(stream.onaddtrack only if stream already exists there)
pc.ontrack with new transceiver and streams
…then once media flowstransceiver.receiver.track.onunmute
tranceiver.direction = 'recvonly'*transceiver.receiver.track.onmute
stream.onremovetrack
tranceiver.direction = 'sendrecv'*stream.onaddtrack
pc.ontrack with existing transceiver and streams
…then once media flowstransceiver.receiver.track.onunmute
transceiver.sender.track.enabled = falseMedia seamlessly goes black/silent
transceiver.sender.track.enabled = trueMedia seamlessly resumes
transceiver.sender.replaceTrack(trackB)Media seamlessly changes
transceiver.sender.replaceTrack(null)Media seamlessly halts
Network (RTP SSRC) timeouttransceiver.receiver.track.onmute
tranceiver.stop()*transceiver.receiver.track.onended

* = after renegotiation through pc.onnegotiationneeded completes.
(Note that many of these transitions are state-based and only fire events if the state ends up changing.)

You can try out this remote-control below in Firefox (59):
In the “Result” tab, grant permission, then click the buttons in sequence from top to bottom, to see the video update.

Then try the sequence again from the top. The sequence is repeatable, because we use the newest transceiver returned from addTransceiver each go-around (they accumulate).

Note that we get stream.onaddtrack from addTransceiver only the second time around, since we had no opportunity to add listeners to the remote stream the first time.

Interestingly, the video element, here representing the remote side, will play along with this, showing the latest video, provided we always either:

  1. stop() the previous transceiver, or
  2. set its direction to 'recvonly'.

Stopping works because video elements ignore ended tracks. Changing direction works because the temporarily-muted tracks get removed from the stream the video element is playing. Try clicking the buttons in different order to prove this out.

Now this may look like a lot of ways to accomplish the same thing. The differences may not be appreciable in a small demo, but each control has trade-offs:

  • stop() terminates both directions (in this example we were only sending one way). Also, stopped transceivers stick around in pc.getTransceivers(), at least locally, and litter the remote stream with ended tracks (however, the Offer/Answer m-line may get repurposed in subsequent negotiations apparently).
  • direction-changes reuse the same transceiver and track without waste, but still require re-negotiation.
  • replaceTrack(null) is instant, requiring no negotiation at all, but stops sending without informing the other party. This may be indistinguishable from a network issue if the other side is looking at stats.
  • track.enabled = false never completely halts network traffic, instead sending one black frame per second, or silence for audio. This is the only control that lets browsers know the camera/microphone is no longer in use.

For the above reasons, the spec encourages implementing “hold” functionality using both direction and replaceTrack(null) in combination.

Don’t forget the camera!

In addition to the spec’s recommended “hold” solution, consider setting track.enabled = false at the same time. If you do, Firefox 60 will turn off the user’s camera and hardware indicator light, for less paranoid face-muting. This is a spec-supported feature Chrome does not have yet, and is the subject of my next blog.

Correlate tracks by transceiver.mid or order

Last year’s blog explained how using track ids out-of-band to correlate remote tracks would no longer work. It recommended using transceiver.mid instead for this, but, sans implementation, left out a working example.

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Here’s an example that correlates tracks regardless of arrival order to always put the camera track on the left:
In the “Result” tab, check the boxes in any order; the camera always appears on the left, the other one on the right.

The trick here in ontrack is using camTransceiver.mid to pick between the left or right video element. This is the mid from the other side. In the real world, we’d send this ID over a) transceivers—in transceiver order—before creating new ones. This is a bit magic.

Webrtc Mozilla

On the other hand, calling addTransceiver() 3 times is straightforward, but would give us 6 m-lines total.

To make due with only 3 m-lines, the answerer must effectively modify the 3 transceivers created by setRemoteDescription from the offer, instead of adding 3 of its own. Think of it as the offerer setting up the transceivers, and the answerer plugging into them.

Make sure you’ve stopped the previous example before running this one.
In the “Result” tab, you’ll see 6 remote tracks, 3 each way, over 3 m-lines total. Mute audio in both video elements.

We use addTransceiver() on one end, and addTrack() on the other to re-use m-lines, relying on their order.

How to answer ONLY with transceivers.

addTrack() looks for unused transceivers to usurp, based on order and kind. This reliance on order may not always be practical. E.g. when using out-of-band mid, it’s more natural to want to modify the transceiver directly. Here the specification comes up a bit short unfortunately.

Let’s see how we’d answer the 3 transceivers without relying on their order, and then discuss what works and what doesn’t (again, make sure you’ve stopped the previous example first):

WebRTC API

Rather than resort to addTrack(), we explicitly modify each transceiver on the answering end:

  1. We change its transceiver.direction from 'recvonly' to 'sendrecv', and
  2. we add our track using tranceiver.sender.replaceTrack().

Unfortunately, this API offers no way to associate streams with these tracks, so our tracks end up being stream-less. Our ontrack code becomes more complicated as a result, since the camera and microphone tracks no longer come grouped into a stream. But at least it works.

Extending the API to provide a way to add stream associations in this situation, seems reasonable. I’ve filed an issue on the spec about this. Update: This has been fixed with the sender.setStreams() API.

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In closing

We’ve found people generally don’t care how media is organized over the wire, until they do. The tipping point is usually some combination of needing to do something more complicated, trying to correlate media to some underlying network metric, or explain some anomaly not gleaned from simpler API abstractions. Hopefully this API gives some insight into how WebRTC actually works, giving you options should you need it.

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Real-time communication for the web

WebRTC Connectivity

With WebRTC, you can add real-time communication capabilities to your application that works on top of an open standard. It supports video, voice, and generic data to be sent between peers, allowing developers to build powerful voice- and video-communication solutions. The technology is available on all modern browsers as well as on native clients for all major platforms. The technologies behind WebRTC are implemented as an open web standard and available as regular JavaScript APIs in all major browsers. For native clients, like Android and iOS applications, a library is available that provides the same functionality. The WebRTC project is open-source and supported by Apple, Google, Microsoft and Mozilla, amongst others. This page is maintained by the Google WebRTC team.

What can WebRTC do?

WebRTC

There are many different use-cases for WebRTC, from basic web apps that uses the camera or microphone, to more advanced video-calling applications and screen sharing. We have gathered a number of code samples to better illustrate how the technology works and what you can use it for.

Application flow

Webrtc For Mozilla

A WebRTC application will usually go through a common application flow. Accessing the media devices, opening peer connections, discovering peers, and start streaming. We recommend that new developers read through our introduction to WebRTC before they start developing.

Next steps

Start with our codelab to become familiar with the WebRTC APIs for the web (JavaScript).




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