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SDP has been a frequent topic, both here on webrtcHacks as well as in the discussion about the standard itself. Modifying the SDP in arcane ways is referred to as SDP munging. This post gives an introduction into what SDP munging is, why its done and why it should not be done. This is not a guide to SDP munging.

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Series preface: We generally lean toward long posts here at webrtcHacks, but not all interesting topics warrant a lot of new text. Sometimes briefer is better. So to better address the many topics that fit into this category, we are starting a new Minimum Duration series. Here is our first post under this set covering Perfect Negotiation.

What is Perfect Negotiation and why do we need it?

Long ago the WebRTC specification designers settled on leaving the signaling communication mechanism between two WebRTC peers up to the application. This means your code needs to handle passing Session Description Protocol (SDP) back and forth and giving that to the peerConnection API. Today WebRTC implementations also almost universally use Trickle-ICE, a form of Interactive Connectivity Establishment (ICE), which passes potential network paths between those peers asynchronously so a connection can be established as soon as possible. The asynchronous but time sensitive nature of all this means it is possible for glare conditions to occur – situations where both sides are making updates at the same time causing their state machines to get out of sync. Differences in how developers implement their code and browsers variances make this worse. ...

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WebRTC has a new browser – kind of. Yesterday Microsoft’s  “new” Edge browser based on Chromium – commonly referred to Edgium – went GA. This certainly will make life easier for WebRTC developers since the previous Edge had many differences from other implementations. The big question is how different is Edgium from Chrome for WebRTC usage?

The short answer is there is no real difference, but you can read below for background details on the tests I ran. If you’re new around WebRTC the rundown may give you some ideas for testing your own product. ...

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Rube Goldberg’s Professor Butts and the Self-Operating Napkin (1931)

Editor’s Note: This post was originally published on October 23, 2018. Zoom recently started using WebRTC’s DataChannels so we have added some new details at the end in the DataChannels section.

Zoom has a web client that allows a participant to join meetings without downloading their app. Chris Koehncke was excited to see how this worked (watch him at the upcoming KrankyGeek event!) so we gave it a try. It worked, removing the download barrier. The quality was acceptable and we had a good chat for half an hour. ...

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As you may have heard, Whatsapp discovered a security issue in their client which was actively exploited in the wild. The exploit did not require the target to pick up the call which is really scary.
Since there are not many facts to go on, lets do some tea reading…

The security advisory issued by Facebook says

A buffer overflow vulnerability in WhatsApp VOIP stack allowed remote code execution via specially crafted series of SRTCP packets sent to a target phone number.

This is not much detail, investigations are probably still ongoing. I would very much like to hear a post-mortem how WhatsApp detected the abuse. ...

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Ammonite

When running WebRTC at scale, you end up hitting issues and frequent regressions. Being able to quickly identify what exactly broke is key to either preventing a regression from landing in Chrome Stable or adapting your own code to avoid the problem. Chrome’s bisect-builds.py tool makes this process much easier than you would suspect. Arne from appear.in gives you an example of how he used this to workaround an issue that came up recently.
{“editor”, “Philipp Hancke“}

In this post I am going to provide a blow-by-blow account of how a change to Chrome triggered a bug in appear.in and how we went about determining exactly what that change was. ...

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A while ago we looked at how Zoom was avoiding WebRTC by using WebAssembly to ship their own audio and video codecs instead of using the ones built into the browser’s WebRTC.  I found an interesting branch in Google’s main (and sadly mostly abandoned) WebRTC sample application apprtc this past January. The branch is named wartc… a name which is going to stick as warts!

The repo contains a number of experiments related to compiling the webrtc.org library as WebAssembly and evaluating the performance. From the rapid timeline, this looks to have been a hackathon project. ...

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Thanks to work initiated by Google Project Zero, fuzzing has become a popular topic within WebRTC since late last year.  It was clear WebRTC was lacking in this area. However, the community has shown its strength by giving this topic an immense amount of focus and resolving many issues.  In a previous post, we showed how to break the Janus Server RTCP parser. The Meetecho team behind Janus did not take that lightly. They got to the bottom of what turned out to be quite a big project. In this post Alessandro Toppi of Meetecho will walk us through how they fixed this problem and built an automated process to help make sure it doesn’t happen again. ...

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