- From: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
- Date: Mon, 21 May 2018 11:33:13 +0200
- To: public-webrtc@w3.org
- Message-ID: <87795e07-fbf7-7dc6-28b3-7ea93b6ab110@gmail.com>
On 21/05/2018 6:11, Harald Alvestrand wrote: > 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. > Fully agree with Harald. IMHO that is the best way forward too. Best regards Sergio
Received on Monday, 21 May 2018 09:32:56 UTC