- From: Cullen Jennings <fluffy@cisco.com>
- Date: Tue, 30 Aug 2011 12:33:37 -0600
- To: public-webrtc@w3.org
When the browser sends and receives audio and video, all kinds of statistics and reporting information is possible. We get information from IP, ICMP, RTP, RTCP, codecs and more. I feel that applications really need access to some of this information and would like to start a thread on what JS applications need. What I really really really don't want to do is enumerate everything that is possible to generate then expose all that via an API. I'd rather use our experience with existing voice and video to figure out what turns out to be useful and report that. Heres is a few things I think might be useful as a starting point Packet loss rate in each direction Round trip time and jitter Current total bitrate each direction These three are the most basic bits of information need to detect and understand why an application may be experiencing poor quality media. Thoughts? What else do we need? If you say all of RTCP-XR I am going to release the flying monkeys. I'm also interested in a simple API proposal for this that has the right sort of extensibility properties as more things get added over time.
Received on Tuesday, 30 August 2011 18:34:17 UTC