RTCRemoteOutboundRTPAudioStream rtt calculation

[emanuele bizzarri]

Hi,

I am checking audio rtt calculation for an incoming audio track.

If I use webrtc samples (munge sdp) it seems that roundTripTime is not calculated by the browser

While if I use Google Meet, where the audio is produced server side, then rtt is calculated by the browser

Can you help me to understand this please?

Thank you,

Emanuele

[Philipp Hancke]

RTT calculation is done based on receiver reports.

Meet is using a mechanism called RRTR which Gustavo Garcia described nicely here:

https://www.rtcbits.com/2022/06/webrtc-google-meet.html

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[emanuele bizzarri]

Hi,

I added

a=rtcp-fb:111 rrtr

to the sdp in munge-sdp example and it works properly

Thank you for your suggestion.

Emanuele

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[emanuele bizzarri]

which is the purpose to have rtt stats on receiver side?

Are they currently used?

[Rajneesh Soni]

I have not seen the libwebrtc code. I think receiver can adjust its NACK generation timeout and jitter buffer based on rtt.