Skip to content
New issue

Have a question about this project? Sign up for a free GitHub account to open an issue and contact its maintainers and the community.

Sign up for GitHub

By clicking “Sign up for GitHub”, you agree to our terms of service and privacy statement. We’ll occasionally send you account related emails.

Already on GitHub? Sign in to your account

Jump to bottom

RTCInboundRtpStreamStats.estimatedEndToEndDelay #538

Closed
wants to merge 1 commit into from
Closed

RTCInboundRtpStreamStats.estimatedEndToEndDelay #538

Changes from all commits
Commits
Show all changes
1 commit
Select commit
File filter

Filter by extension

Filter by extension
Conversations
Failed to load comments.
Loading
Jump to
Jump to file
Failed to load files.
Loading
Diff view
Diff view
75 changes: 74 additions & 1 deletion webrtc-stats.html
View file
Open in desktop
Original file line number Diff line number Diff line change
Expand Up @@ -1200,6 +1200,7 @@ <h3>
unsigned long pliCount;
unsigned long sliCount;
DOMHighResTimeStamp estimatedPlayoutTimestamp;
double estimatedEndToEndDelay;
double jitterBufferDelay;
unsigned long long jitterBufferEmittedCount;
unsigned long long totalSamplesReceived;
Expand Down Expand Up @@ -1549,14 +1550,86 @@ <h2>
timestamp (from an RTCP SR packet mapping RTP timestamps to NTP timestamps),
extrapolated with the time elapsed since it was ready to be played out. This is
the "current time" of the track in NTP clock time of the sender and can be present
even if there is no audio currently playing.
even if there is no audio or video currently playing.
</p>
<p>
This can be useful for estimating how much audio and video is out of sync for two
tracks from the same source, <code>audioTrackStats.estimatedPlayoutTimestamp -
videoTrackStats.estimatedPlayoutTimestamp</code>.
</p>
</dd>
<dt>
<dfn><code>estimatedEndToEndDelay</code></dfn> of type <span class=
"idlMemberType">double</span>
</dt>
<dd>
<p>
This is the estimated delay, in seconds, between the time of
capture on the sending side and the time of playout on the
receiving side. It is obtained by converting
<code>estimatedPlayoutTimestamp</code> (the capture time of
media currently being played out, in sender NTP clock time) to
receiver NTP clock time, using an estimation of the delta
between the sender NTP clock and the receiver NTP clock. The
difference between the current time, in receiver NTP clock,
and the estimated capture time, in receiver NTP clock, is the
<code>estimatedEndToEndDelay</code>. This value is present
whenever <code>estimatedPlayoutTimestamp</code> is present,
meaning it can be present even if there is currently no audio
or video playing.
</p>
<p>
Upon receiving an RTCP SR packet, run the following steps:
</p>
<ol>
<li>
Let <var>reportSentInSenderNtp</var> be the NTP timestamp
that the report was sent, according to the sender's
wallclock, as defined in section 6.4.1. of [[!RFC3550]].
</li>
<li>
Let <var>roundTripTime</var> be the estimated round-trip
time, as defined in section 6.4.1. of [[!RFC3550]].
</li>
<li>
Let <var>reportReceivedInReceiverNtp</var> be the NTP
timestamp that the report was received, according to the
receiving endpoint's wallclock.
</li>
<li>
Let <var>estimatedNtpDelta</var> be
<var>reportReceivedInReceiverNtp</var> -
<var>reportSentInSenderNtp</var> -
(<var>roundTripTime</var> / 2).
</li>
</ol>
<p>
Then end-to-end delay is computed as follows:
</p>
<ol>
<li>
Let <var>nowInReceiverNtp</var> be the current NTP time,
according to the receiving endpoint's wallclock.
</li>
<li>
Let <var>estimatedPlayoutTimestamp</var> be calculated
according to the definition of
<code>estimatedPlayoutTimestamp</code>.
</li>
<li>
Let <code>estimatedEndToEndDelay</code> be
<var>nowInReceiverNtp</var> -
(<var>estimatedPlayoutTimestamp</var> +
<var>estimatedNtpDelta</var>).
</li>
</ol>
<p>
In the above calculations, the implementation MAY use smoothed
values for <var>roundTripTime</var> and
<var>estimatedNtpDelta</var> based on multiple RTCP SR samples
over a sliding window.
</p>
</dd>
<dt>
<dfn><code>jitterBufferDelay</code></dfn> of type <span class=
"idlMemberType">double</span>
Expand Down