Re: [webrtc-pc] Prefer variable ptime (audio frame lengths) (#2300)

I am talking about most of the browser implementations including Chromium (they are using NetEq jitter buffer) as well as any WebRTC enabled media servers and transcoding gateways (such as Asterisk or FreeSwitch), or phones connected via some sort of ICE/DTLS-SRTP to SRTP gateway (such as things derived from rtpengine). All of those will have problems. SFU generally would not care. 

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Received on Tuesday, 17 September 2019 00:06:27 UTC