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Re: Transport / encoder coupling (Re: Use cases / requirements for raw data access functions)

From: Peter Thatcher <pthatcher@google.com>
Date: Tue, 22 May 2018 14:21:25 -0700
Message-ID: <CAJrXDUEH7ZuV_p+7k6zSfUGTPNxQ82CVjG2g4BYGR-gG7xAoLQ@mail.gmail.com>
To: Harald Alvestrand <harald@alvestrand.no>
Cc: public-webrtc@w3.org
I made a diagram of my thinking of the different possible API layers.  Here
it is:

[image: RTP object stack.png]

This highlights the question: what happens to RtpParameters?  Some are
per-transport, some are per-stream, some are per-encoder (which is
different than per-stream), and some don't make sense any more.

Per-transport:
- Payload type <-> codec mapping
- RTCP reduced size
- Header extension ID <-> URI mapping

Per-stream:
- PTs used for PT-based demux
- priority
- mid
- cname
- ssrcs

Per-encoder:
- dtx
- ptime
- codec parameters
- bitrate
- maxFramerate
- scaleResolutionDownBy

Don't make sense:
- active (just stop the encoder)
- degradation preference (just change the encoder directly)


Providing the per-transport and per-encoder ones is straightforward.  But
what about per-stream parameters?  Where do those go?  Perhaps we'll need
some kind RTP object that works with a encoded media instead of with
unencoded/decoded tracks.  Perhaps EncodedRtpSender and
EncodedRtpReceiver?   And if we have those, do we even need a separate
RtpFrameTransport instead of just sticking an EncodedRtpSender and
EncodedRtpReceiver directly on top of the RtpPacketTransport?




On Tue, May 22, 2018 at 1:00 AM Harald Alvestrand <harald@alvestrand.no>
wrote:

> On 05/21/2018 11:14 PM, Peter Thatcher wrote:
>
>
>
> On Sun, May 20, 2018 at 9:14 PM Harald Alvestrand <harald@alvestrand.no>
> 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.
>>
>
> You are correct that a transport needs to have certain signals like a
> BWE.  It's also useful if it can give info about when a message/frame has
> been either acked or thrown away.
>
>
>
> You are also correct that a high-level RTP transport would need to be told
> how to packetize.  But a low-level RTP transport could leave that up to the
> app.
>
>
> I think we're closing in on saying that the conceptual model of an RTP
> transport is layered: There's a layer that sends frames (which knows about
> codec specific packetization and would tell the encoder that a frame has
> been lost), and there's a layer that sends packets (and doesn't need to
> know about how to break frames into packets).
>
> The RTPFrameTransport would offer an interface similar to what
> QUICTransport is proposed to do, while RTPPacketTransport would offer an
> interface that lets the client set fields like PT. I think SRTP encryption
> is agnostic to packet content, so it could be part of the
> RTPPacketTransport.
>
> merging in another thread on the same subject: unlike our current model
> (and, I think, ORTC), I would think of an RTPTransport as an object that
> handles multiple media streams. An RTPSender would be the object that
> handles one media stream, since that's what we're using the name for today.
> RTPSender stlll seems to have a mission.
>
>
>
>
>
>
>> --
>> Surveillance is pervasive. Go Dark.
>>
>>
>

RTP_object_stack.png
(image/png attachment: RTP_object_stack.png)

Received on Tuesday, 22 May 2018 21:22:05 UTC

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