Introduction to capture handle - a new Chrome Origin Trial that lets a WebRTC screen sharing application communicate with the tab it is capturing. Examples use case discussed include detecting self-capture, improving the use of collaboration apps that are screen shared, and optimizing stream parameters of the captured content.
Chrome recently added the option of adding redundancy to audio streams using the RED format as defined in RFC 2198, and Fippo wrote about the process and implementation in a previous article. You should catch-up on that post, but to summarize quickly RED works by adding redundant payloads with different timestamps in the same packet. If you lose a packet in a lossy network then chances are another successfully received packet will have the missing data resulting in better audio quality.
That was in a simplified one-to-one scenario, but audio quality issues often have the most impact on larger multi-party calls. As a follow-up to Fippo’s post, Jitsi Architect and Improving Scale and Media Quality with Cascading SFUs author Boris Grozev walks us through his design and tests for adding audio redundancy to a more complex environment with many peers routing media through a Selective Forwarding Unit (SFU). ...
Back in April 2020 a Citizenlab reported on Zoom’s rather weak encryption and stated that Zoom uses the SILK codec for audio. Sadly, the article did not contain the raw data to validate that and let me look at it further. Thankfully Natalie Silvanovich from Googles Project Zero helped me out using the Frida tracing tool and provided a short dump of some raw SILK frames. Analysis of this inspired me to take a look at how WebRTC handles audio. In terms of perception, audio quality is much more critical for the perceived quality of a call as we tend to notice even small glitches. Mere ten seconds of this audio analysis were enough to set me off on quite an adventure investigating possible improvements to the audio quality provided by WebRTC. ...
Time for another opinionated post. This time on… end-to-end encryption (e2ee). Zoom apparently claims it supports e2ee while it can not satisfy that promise. Is WebRTC any better?
Let’s get to the bottom of things fast: Boo Zoom!
I reviewed how Zoom’s implements their web client last year.
I’m not really surprised of their general lack of e2ee given that their web client did not provide any encryption on top of TLS or WebRTC’s DataChannel. For reasons we will discuss below, this means they weren’t doing any obvious e2ee there. ...
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.
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. ...
We have have had many posts on Session Description Protocol (SDP) here at werbrtcHacks. Why? Because it is often the most confusing yet critical aspects of WebRTC. It has also been among the most controversial. Earlier in WebRTC debates over SDP lead the to the development of the parallel ORTC standard which is now largely merging back into the core specifications. However, the reality is non-SDP based WebRTC is still a small minority of deployments and many have doubts this will change any time soon despite its formal acceptance. ...
If you are new to WebRTC then you have missed out on years of drama in the standards bodies over various issues like SDP and codecs. These standards dictate what vendors must implement so they ultimately dictate the industry roadmap. To get a deep perspective and appreciation of the issues, we like to ask Dan Burnett, W3C editor to comment on where we are at with the standardization process. I caught up with Dan at this year’s IIT Real Time Communications Conference and had the more detailed Q&A with him shortly thereafter. ...
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