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insect

Editor Note: Fippo uses a lot of advanced WebRTC terms below – if you are a regular reader of this blog then don’t let that scare  you. Wireshark is a great tool for diagnosing media issues and inspecting signaling packets even if you’re not building a media server. {“editor”, “chad hart“}

Stuff breaks all the time and then you need to debug it. My favorite tool for this remains Wireshark as we have seen previously. Its fairly useful for debugging all the ICE and DTLS stuff but recently I’ve had to debug the media traffic itself.

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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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Speak No Evil

A few days back my old friend Chris Koehncke, better known as “Kranky” asked me how hard it would be to implement a wild idea he had to monitor what percentage of the time you spent talking instead of listening on a call when using WebRTC. When I said “one day” that made him wonder whether he could offshore it to save money. Well… good luck!

A week later Kranky showed me some code. Wait, he is writing code? It was not bad – it was using the WebAudio API so going in the right direction. It was enough to prod me to finish writing the app for him.

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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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Chrome, Firefox, and Edge are all on the same party line. Image from Pillow Talk (1959)

For the first time, Chrome, Firefox and Edge can “talk” to each other via WebRTC and ORTC. Check the demo on Microsoft’s modern.ie testdrive.

tl;dr: don’t worry, audio works. codec interop issue…

Feature Interoperability Notes
ICE yes Edge requires end-of-candidate signaling
DTLS yes
audio yes using G.722, Opus or G.711 codecs
video no standard H.264 is not supported in Edge yet
DataChannels no Edge does not support dataChannels

As a reader of this blog, you probably know what WebRTC is but let me quote this:

WebRTC is a new set of technologies that brings clear crisp voice, sharp high-definition (HD) video and low-delay communication to the web browser.

In order to succeed, a web-based communications platform needs to work across browsers. Thanks to the work and participation of the W3C and IETF communities in developing the platform, Chrome and Firefox can now communicate by using standard technologies such as the Opus and VP8 codecs for audio and video, DTLS-SRTP for encryption, and ICE for networking.

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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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ORTC support in Edge has been announced today. A while back, we saw this on twitter:

“This release [build 10525] lays the groundwork for ORTC” was quite an understatement. It was considered experimental and while the implementation still differs from the specification (which is still work in progress) slightly, it already worked and as a developer you can get familiar with how ORTC works and how it is different from the RTCPeerConnection API.
If you want to test this, please use builds newer than 10547. Join the Windows Insider Program to get them and make sure you’re on the fast ring.

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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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ny times stylized

So the New York times uses WebRTC to gather your local ip addresses… Tsahi describes the non-technical parts of the issue in his blog. Let’s look at the technical details… it turns out that the Javascript code used is very clunky and inefficient.

First thing to do is to check chrome://webrtc-internals (my favorite tool since the hangouts analysis). And indeed, nytimes.com is using the RTCPeerConnection API. We can see a peerconnection created with the RtpDataChannels argument set to true and using stun:ph.tagsrvcs.com as a STUN server.
Also, we see that a data channel is created, followed by calls to createOffer and setLocalDescription. That pattern is pretty common to gather IP addresses.

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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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