Technology

Dealing with multi-party video infrastructure can be pretty daunting. The good news is the technology, products, and standards to enable economical multiparty video in WebRTC has matured quite a bit in the past few years. One of the key underlying technologies enabling some of this change is called simulcast. Simulcast has been an occasional sub-topic here at webrtcHacks in the past and it is time we gave it more dedicated attention.

To do that we asked Oscar Divorra Escoda, Tokbox’s Senior Media Scientist and Media Cloud Engineering Lead to walk us through it. Tokbox was one of the first to market with a SFU and Oscar shares some of his learnings below. ...

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Earlier this month Fippo published a post analyzing Slack’s new WebRTC implementation. He did not have direct access or a team account to do a thorough deep dive – not to mention he is supposed to be taking some off this month. That left many with some open questions? Is there more to the TURN network? How does multi-party calling work? How exactly is Slack using the Janus gateway? Fortunately WebRTC has an awesomely active and capable community that quickly picked up the slack (pun intended). ...

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slack webrtc2

Dear Slack,

There has been quite some buzz this week about you and WebRTC.

WebRTC… kind of. Because actually you only do stuff in Chrome and your native apps:

I’ve been there. Launching stuff only for Chrome. That was is late 2012. In 2016, you need to have a very good excuse to launch something with WebRTC and not support Firefox like this:
 

Maybe you had your reasons. As usual, I tried to get a dump from chrome://webrtc-internals to see what is going on. Thanks to Dag-Inge Aas for providing one. The most interesting bit is the call to setRemoteDescription:

I would like to note that you reply to Chrome’s offer of UDP/TLS/RTP/SAVPF with a profile of RTP/SAVPF. While that is still tolerated by browsers, it is improper.
Your a=msid-semantic line looks very interesting. “WMS janus”. Sounds familiar, this is meetecho’s janus gateway (see Lorenzo’s post on gateways here). Which by the way works fine with Firefox from what I hear. ...

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I Think I'm Being Watched

There has been more noise about WebRTC making it possible to track users. We have covered some of the nefarious uses of WebRTC and look out for it before. After reading a blog post on this topic covering some allegedly new unaddressed issues a week ago I decided to ignore it after some discussion on the mozilla IRC channel. But this has some up on a the twitter-sphere again and Tsahi said ‘ouch’, here are my thoughts.

Claims

The blog post (available here) makes a number of claims about how certain Chrome behavior makes fingerprinting easier: ...

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We have been waiting a long time for Microsoft to add WebRTC to its browser portfolio. That day finally came last month when Microsoft announced its new Windows 10 Edge browser had ORTC. This certainly does not immediately address the Internet Explorer population and ORTC is still new to many (which is why we cover it often). On the positive side, interoperability between Edge, Chrome, and Firefox on the audio side was proven within days by multiple parties. Much of ORTC is finding its way into the WebRTC 1.0 specification and browser implementations. ...

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So I talked about Skype and Viber at KrankyGeek two weeks ago. Watch the video on youtube or take a look at the slides. No “reports” or packet dumps to publish this time, mostly because it is very hard to draw conclusions from the results.

The VoIP services we have looked at so far which use the RTP protocol for transferring media. RTP uses a packet header which is not encrypted and contains a number of attributes such as the payload type (identifying the codec used), a synchronization source (which identifies the source of the stream), a sequence number and a timestamp. This allows routers to identify RTP packets and prioritize them. This also allows someone monitoring all network traffic (“Pervasive Monitoring“) to easily identify VoIP traffic. Or someone wiretapping your internet connection. ...

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One of WebRTC’s benefits is that the source to it is all open source. Building WebRTC from source provides you the ultimate flexibility to do what you want with the code, but it is also crazy difficult for all but the small few VoIP stack developers who have been dedicated to doing this for years. What benefit does the open source code provide if you can’t figure out how to build from it?

As WebRTC matures into mobile, native desktop apps, and now into embedded devices as part of the Internet of Things, working with the lower-level source code is becoming increasingly common. ...

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This is the next decode and analysis in Philipp Hancke’s Blackbox Exploration series conducted by &yet in collaboration with Google. Please see our previous posts covering WhatsApp, Facebook Messenger and FaceTime for more details on these services and this series. {“editor”: “chad hart“}

Wire is an attempt to reimagine communications for the mobile age. It is a messaging app available for Android, iOS, Mac, and now web that supports audio calls, group messaging and picture sharing. One of it’s often quoted features is the elegant design. As usual, this report will focus on the low level VoIP aspects, and leave the design aspects up for the users to judge. ...

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Atlassian’s HipChat acquired BlueJimp, the company behind the Jitsi open source project. Other than for positive motivation, why should WebRTC developers care? Well, Jitsi had its Jitsi Video Bridge (JVB) which was one of the few open source Selective Forwarding Units (SFU) projects out there. Jitsi’s founder and past webrtcHacks guest author, Emil Ivov, was a major advocate for this architecture in both the standards bodies and in the public. As we have covered in the past, SFU’s are an effective way to add multiparty video to WebRTC. Beyond this one component, Jitsi was also a popular open source project for its VoIP client, XMPP components, and much more. ...

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WebRTC is supposed to be secure. A lot more than previous VoIP standards. It isn’t because it uses any special new mechanism, but rather because it takes it seriously and mandates it for all sessions.

Alan Johnston decided to take WebRTC for a MitM spin – checking how easy is it to devise a man-in-the-middle attack on a naive implementation. This should be a reminder to all of us that while WebRTC may take care of security, we should secure our signaling path and the application as well. ...

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