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Hi All,</div>
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I've asked in other forums without success. I'm hoping this community (maybe Paul?) can point me in the right direction. It's intellectual curiosity and not business related.<span id="��"></span></div>
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<span style="font-family: Calibri, Arial, Helvetica, sans-serif; font-size: 12pt;">I'm trying to understand which is better for tx/rx of highest quality and lowest latency audio to multple remote peers - uncompressed audio (for example jacktrrip 2 channel 48k
/ 128 frames), or something like opus at 48k 2 channel (for example webrtc using opus)? Granted webrtc opus is typically mono but there are sip clients like baresip which can use opus at 2 channel stereo at 48k. Am I completely wrong in thinking that the two
are nearly identical as far as quality? My focus is within a controlled environment like a flat wired or optical 1g or 10g ethernet.</span></div>
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<span style="font-family: Calibri, Arial, Helvetica, sans-serif; font-size: 12pt;">Does the use of opus (and a sip stack/webrtc) add latency overall, or are there efficiency gains (for example network bandwidth reduction) through its use?</span></div>
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Could this difference, if any, be measured using jack_iodelay? I was thinking about running jack_iodelay through a loop between jacktrip p2p vs baresip+jack+opus at 48k stereo p2p with a loopback cable.</div>
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<span style="font-family: Calibri, Arial, Helvetica, sans-serif; font-size: 12pt;">Thanks,</span><br>
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Matt</div>
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