Comments on: WebRTC test: How to go about testing WebRTC? https://bloggeek.me/test-your-webrtc-service/ The leading authority on WebRTC Sat, 26 Feb 2022 14:53:56 +0000 hourly 1 By: Tsahi Levent-Levi https://bloggeek.me/test-your-webrtc-service/#comment-123532 Thu, 24 Sep 2020 06:30:18 +0000 https://bloggeek.me/?p=9789#comment-123532 In reply to Amit Verma.

Amit,

testRTC (I am co-founder and CEO there) offers a commercial service for that.

KITE is an open source tool that does similar things.

]]> By: Amit Verma https://bloggeek.me/test-your-webrtc-service/#comment-123531 Thu, 24 Sep 2020 06:23:52 +0000 https://bloggeek.me/?p=9789#comment-123531 Hi

Is there any tool which could test my WebRTC app feature which is for Video Voice calling and check its Quality as well.

Please let me know if you can help.

]]> By: Tsahi Levent-Levi https://bloggeek.me/test-your-webrtc-service/#comment-120812 Mon, 09 Mar 2020 16:37:09 +0000 https://bloggeek.me/?p=9789#comment-120812 In reply to Steve Fox.

Thanks for sharing this Steve.

I had to cut the length of this article at some point (it was getting longer as it is).

Interoperability is a never ending testing effort for browser vendors. My assumption is that their focus is on interoperablity across the browsers as they are trying to hit the WebRTC 1.0 milestone at the W3C, coupled with the effort of introduction of things like AV1 (at some point) and making more use of VP9 in actual services.

]]> By: Steve Fox https://bloggeek.me/test-your-webrtc-service/#comment-120811 Mon, 09 Mar 2020 15:30:14 +0000 https://bloggeek.me/?p=9789#comment-120811 The article could be more detailed; now it’s just a brief overview of the subject.
For example, just the interoperability section deserves the article on it’s own,
demonstrating issues and challenges of a) pushing content generated by various encoding hardware to WebRTC, and b) pulling the streams from WebRTC into other multimedia domains – into ffmpeg, DirectShow, decoding STBs, transcoding clowds etc… Kind of the stuff that Google developers never cared about – they only really care about browser-to-browser WebRTC. Huge demand for interoperability exists; look, for example, at this Stackoverflow thread: https://stackoverflow.com/questions/59865405/use-webrtc-getusermedia-stream-as-input-for-ffmpeg/60143788#60143788
There is not a lot of supply meeting that demand. And yes, lots of interoperability issues do exist. Let me give you one typical example: Intel QuickSync H264 encoder produces a stream with 3-bytes and 4-bytes startup codes in front of H264 NALUs. But H264 decoder used for WebRTC in Chrome (mojovideodecoder), does not accept 3-bytes startup codes, despite that standard allows 3-bytes startup codes. So that video stream will not play in Chrome, when sent via WebRTC.
We send numerous bug reports to Chrome developers, and they are usually very responsive and try to fix these bugs, but the real issue is – WebRTC interoperability is never even on a radar of Google’s developers, they don’t think about it.

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