Blackbox Exploration

All posts tagged Blackbox Exploration

Slack is an über popular and fast growing communications tool that has a ton of integrations with various WebRTC services. Slack acquired a WebRTC company a year ago and launched its own audio conferencing service earlier this year which we analyzed here and here. Earlier this week they launched video. Does this work the same? Are there any tricks we can learn from their implementation? Long time WebRTC expert and webrtcHacks guest author Gustavo Garica takes a deeper dive into Slack’s new video conferencing feature below to see what’s going on under the hood.

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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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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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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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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 and Facebook Messenger for more details on these services and this series. {“editor”: “chad hart“}

FaceTime is Apple’s answer to video chat, coming preinstalled on all modern iPhones and iPads. It allows audio and video calls over WiFi and, since 2011, 3G too. Since Apple does not talk much about WebRTC (or anything else), maybe we can find out if they are using WebRTC behind the scenes?

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Two weeks ago Philipp Hancke,  lead WebRTC developer of Talky and part of the &yet‘s WebRTC consulting team, started a series of posts about detailed examinations he is doing on several major VoIP deployments to see if and how they may be using WebRTC. Please see that post on WhatsApp for some background on the series and below for another great analysis – this time on Facebook Messenger. {“editor”: “chad hart“}

Last week, Facebook announced support for video chats in their Messenger app. Given that Messenger claims to account for 10% of global mobile VoIP traffic, this made in a very interesting target for further investigation. As part of the series of deconstructions, the full analysis (another fifteen pages, using the full range of analysis techniques demonstrated earlier) is available for download here, including the wireshark dumps.

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One of our first posts was a Wireshark analysis of Amazon’s Mayday service to see if it was actually using WebRTC. In the very early days of WebRTC, verifying a major deployment like this was an important milestone for the WebRTC community. More recently, Philipp Hancke – aka Fippo – did several great posts analyzing Google Hangouts and Mozilla’s Hello service in Firefox. These analyses validate that WebRTC can be successfully deployed by major companies at scale. They also provide valuable insight for developers and architects on how to build a WebRTC service.

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There have been many major WebRTC launches in the past months including Facebook and KimDotCom. Before those, Mozilla started bundling a new WebRTC calling service right into Firefox. Of course we wanted to check out to see how it worked.

To help do this we called on the big guns – webrtcHacks guest columnist Philipp Hancke. Philipp is one of the smartest guys in WebRTC outside of Google. In addition to his paid work for &yet he is the leading non-googler to contribute to the webrtc demos and samples and is also a major contributor to the Jitsi Meet and strophe.jingle projects. Google even asks him to proof-read their WebRTC release notes.

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Update: Philipp continues to reverse engineer Hangouts using chrome://webrtc-internals. Please see the bottom section for new analysis he just put together in the past couple of days based on Chrome 38.

As initiators and major drivers of WebRTC, Google was often given a hard time for not supporting WebRTC in its core collaboration product. This recently changed when WebRTC support for Hangouts was added with Chrome 36.

So obviously we wanted to check out how this worked. We also were curious to see how a non-googler could make some practical use of chrome://webrtc-internals. Soon thereafter I came across a message from Philipp Hancke (aka Hornsby Cornflower) saying he had already starting looking at the new WebRTC hangouts with webrtc-internals. Fortunately I was able to convince him to share his findings and thorough analysis.

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