- From: Harald Alvestrand <harald@alvestrand.no>
- Date: Mon, 21 May 2018 06:11:38 +0200
- To: public-webrtc@w3.org
- Message-ID: <99313fd0-ac94-3da1-490c-56aafd909e8d@alvestrand.no>
On 05/19/2018 12:11 AM, Sergio Garcia Murillo wrote: >> >> We should separate transports from encoders (split the RtpSender in >> half) to give more flexibility to apps. > > I agree with that, but given RTP packetization is codec specific, we > can't ignore that. I think we can translate this into requirements language: IF we specify the transport separately from the encoder THEN we need a number of pieces of API, either as JS API or as specifications of "what happens when you couple a transport with an encoder": - If the transport is like RTP, and isn't codec agnostic, it needs information from the encoder about what the codec is, and may need to inform the encoder about some transport characteristics (like max MTU size for H.264 packetization mode 0). - If the transport is congestion controlled (and they all are!), then there needs to be coupling of the transport's congestion control with the encoder's target bitrate - If the transport isn't reliable, and may toss frames, it needs to be coupled back to the encoder to tell it to do things like stop referring to a lost frame, or to go back and emit a new iframe. Handling the data is just one part of handling the interaction between transport and encoder. -- Surveillance is pervasive. Go Dark.
Received on Monday, 21 May 2018 04:12:04 UTC