[webrtc-pc] Is it possible to manipulate playout-delay extention at SDP of RTCSessionDescription directly affect playout (playback) delay at remote peer connection? (#2213)

guest271314 has just created a new issue for https://github.com/w3c/webrtc-pc:

== Is it possible to manipulate playout-delay extention at SDP of RTCSessionDescription directly affect playout (playback) delay at remote peer connection? ==
Is it possible to manipulate playout-delay extention header at SDP of local RTCSessionDescription directly affect playout (playback) delay at remote peer connection? 

That is, to compose an appropriate character, value pair for the purpose of delaying media being playout (playback) at remote peer, for example, for 10 seconds (given a single or the predominant timing mechanism being used by WebRTC, when local stream is at 10 of playback, remote stream is at 0 of playback)?

Or is the purpose of playout-delay header extension for statistics, i.e., `getStats()`?

-- 

Relevant documentation, Pull Requests, comments

playout-delay https://webrtc.org/experiments/rtp-hdrext/playout-delay/

> The header is attached to the RTP packet by the RTP sender when it needs to change the min and max smoothing delay at the receiver. Once the sender is informed that at least one RTP packet which has the min and max details is delivered, it MAY stop providing details on all further RTP packets until another change warrants communicating the details to the receiver again. This is done as follows:
> 
> RTCP feedback to RTP sender includes the highest sequence number that was seen on the RTP receiver. The RTP sender can track the sequence number on the packet that first had the playout delay extension and then stop sending the extension once the received sequence number is greater than the sequence number on the first packet containing the current values playout delay in this extension.

Chromium source rtp_header_extensions.cc https://chromium.googlesource.com/external/webrtc/+/e2a017725570ead5946a4ca8235af27470ca0df9/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc#177


```
//   0                   1                   2                   3
//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
//  |  ID   | len=2 |   MIN delay           |   MAX delay           |
//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
```

jitterBufferDelay vs playoutDelay #223 https://github.com/w3c/webrtc-stats/issues/223


> SGTM I think playoutDelay is misleading, that could be a different stat if need be. - https://github.com/w3c/webrtc-stats/issues/223#issuecomment-341532688 

Playout delay #216 https://github.com/webrtc/webrtc-org/pull/216



Please view or discuss this issue at https://github.com/w3c/webrtc-pc/issues/2213 using your GitHub account

Received on Monday, 17 June 2019 02:23:09 UTC