Re: [rtcweb] Proposal for a JS API for NoPlan (adding multiple sources without encoding them in SDP)

Robin Raymond <robin@hookflash.com> Tue, 18 June 2013 02:54 UTC

Return-Path: <robin@hookflash.com>
X-Original-To: rtcweb@ietfa.amsl.com
Delivered-To: rtcweb@ietfa.amsl.com
Received: from localhost (localhost [127.0.0.1]) by ietfa.amsl.com (Postfix) with ESMTP id D086D11E80C5 for <rtcweb@ietfa.amsl.com>; Mon, 17 Jun 2013 19:54:40 -0700 (PDT)
X-Virus-Scanned: amavisd-new at amsl.com
X-Spam-Flag: NO
X-Spam-Score: -2.598
X-Spam-Level:
X-Spam-Status: No, score=-2.598 tagged_above=-999 required=5 tests=[BAYES_00=-2.599, HTML_MESSAGE=0.001]
Received: from mail.ietf.org ([12.22.58.30]) by localhost (ietfa.amsl.com [127.0.0.1]) (amavisd-new, port 10024) with ESMTP id 4eD2UDSthybY for <rtcweb@ietfa.amsl.com>; Mon, 17 Jun 2013 19:54:38 -0700 (PDT)
Received: from mail-ie0-x22f.google.com (mail-ie0-x22f.google.com [IPv6:2607:f8b0:4001:c03::22f]) by ietfa.amsl.com (Postfix) with ESMTP id 5AEA621F9C22 for <rtcweb@ietf.org>; Mon, 17 Jun 2013 19:54:38 -0700 (PDT)
Received: by mail-ie0-f175.google.com with SMTP id a13so8865635iee.34 for <rtcweb@ietf.org>; Mon, 17 Jun 2013 19:54:38 -0700 (PDT)
X-Google-DKIM-Signature: v=1; a=rsa-sha256; c=relaxed/relaxed; d=google.com; s=20120113; h=message-id:date:from:user-agent:mime-version:to:cc:subject :references:in-reply-to:content-type:x-gm-message-state; bh=7oSltqAMnlNEWENS5ngXAOhQotY79JrxTWeLKqyUnqk=; b=hO6bd4sc5s0ZHHZx9GYh/oWM07chyEl/u2bHc009IPEfkn7Nvvh+QPsRzS/pnz34jT ZfJtCBdVhMq6c0gcc2GtCr+Y/t0o5f3RNbiypCxd55dkisl/eZRSBNwjn1XgHWicfQvN TuvaPGvbIatnQH+k0uxB6QHyYFlzfO0anCt1ZY9/Yn95+wFhgE20RwdgG4STpp4lB9Ws Y4Hk6+qL/a3s1XERZFhuMPiqLv3Fn7tJHj3AHhHMd3PNaoHVcaT2eoR89/dVNM1rM2ch Kl/jPWgN8LHrwtnIalpaRLGFiRyzHjyDKa0cX8Ufvu1zFQpvR7RlHgO+ujN4ELGbAKpR CERw==
X-Received: by 10.50.131.161 with SMTP id on1mr6594192igb.112.1371524077886; Mon, 17 Jun 2013 19:54:37 -0700 (PDT)
Received: from Robins-MacBook-Pro.local (CPE602ad08742f7-CM602ad08742f4.cpe.net.cable.rogers.com. [99.224.116.224]) by mx.google.com with ESMTPSA id y11sm20179622igy.10.2013.06.17.19.54.35 for <multiple recipients> (version=TLSv1 cipher=ECDHE-RSA-RC4-SHA bits=128/128); Mon, 17 Jun 2013 19:54:36 -0700 (PDT)
Message-ID: <51BFCBE9.5070406@hookflash.com>
Date: Mon, 17 Jun 2013 22:54:33 -0400
From: Robin Raymond <robin@hookflash.com>
User-Agent: Postbox 3.0.8 (Macintosh/20130427)
MIME-Version: 1.0
To: Silvia Pfeiffer <silviapfeiffer1@gmail.com>
References: <CAJrXDUHdoxLTsofiwLBdwBNnCCkCBgjSdbmLaXrNEPODMrsSVA@mail.gmail.com> <CAHp8n2m4VwkpbdGE+q73qqij5RDCB4Vb-Ui1LmGSx1zmv8TX2g@mail.gmail.com>
In-Reply-To: <CAHp8n2m4VwkpbdGE+q73qqij5RDCB4Vb-Ui1LmGSx1zmv8TX2g@mail.gmail.com>
Content-Type: multipart/alternative; boundary="------------090100000502080103010206"
X-Gm-Message-State: ALoCoQlEyaBGFTHh17d5+I5Zfc4luJe03ejBRAV1kAedVlQ8cx6fTvUwo01bkVUO4J1yguVDXufA
Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Proposal for a JS API for NoPlan (adding multiple sources without encoding them in SDP)
X-BeenThere: rtcweb@ietf.org
X-Mailman-Version: 2.1.12
Precedence: list
List-Id: Real-Time Communication in WEB-browsers working group list <rtcweb.ietf.org>
List-Unsubscribe: <https://www.ietf.org/mailman/options/rtcweb>, <mailto:rtcweb-request@ietf.org?subject=unsubscribe>
List-Archive: <http://www.ietf.org/mail-archive/web/rtcweb>
List-Post: <mailto:rtcweb@ietf.org>
List-Help: <mailto:rtcweb-request@ietf.org?subject=help>
List-Subscribe: <https://www.ietf.org/mailman/listinfo/rtcweb>, <mailto:rtcweb-request@ietf.org?subject=subscribe>
X-List-Received-Date: Tue, 18 Jun 2013 02:54:41 -0000

I actually need access to the raw controls like SSRC, detailed ICE 
candidate control, encryption keying, etc, and as much control over how 
media should be controlled/behave as possible. Those things are 
extremely important for interoperability and signaling and have specific 
meaning for those who deal with that kind of stuff daily.

However, as a JavaScript developer, you likely need a simple high level 
API. Why should you care about all that low level junk? It's meaningless 
nonsense to you.

Real-Time Communication is hard because there is a lot involved, 
especially if there is any kind of compatibility, advanced features or 
network control. Nobody wants a difficult API for the sake of being 
difficult, especially for a sleek language like JavaScript. It's more 
that it's just necessary if we audio/video people want to do some really 
cool features that work with new or existing platforms, devices and 
services.

But we can have the best of both worlds! There are many people who are 
in this RTC domain who can wrap that lower level API to give very simple 
and easy to understand conceptual APIs that abstract that weird funky 
stuff away. I'd be one of those people to offer such a simpler library 
(along with many others I'm sure).

In some way, it's not all that dissimilar with DOM and CSS3, etc.; 
jQuery and jQuery UI (and other libraries) create wrappers to make 
access and control easier but for those who need to raw control they 
have it. Most people use wrappers and toolkits when they are trying to 
do the standard stuff, but someone put together the APIs originally to 
make common use cases.

On a side note, I agree that SDP should go (and quickly) but it's not 
the format that's the problem. Granted, it is obtuse. But it's not just 
that; it's a monolithic do-everything offer/answer model that offers 
very little control and the API to control the browser's RTC is 
effectively via manipulation of the SDP. Yuck! Even if it were fancier 
and prettier JSON, it would still be an ugly do-everything monolithic 
object with a sketchy offer/answer model that is brittle and offered 
very little real-scenario controls for those whom need it. It's 
absolutely horrible and is in good need of quick deprecation.

-Robin




> Silvia Pfeiffer <mailto:silviapfeiffer1@gmail.com>
> 17 June, 2013 9:00 PM
> Hi Peter, all,
>
> I'm looking at all this from the view of a JS developer and I am
> really excited that there is movement in this space. Having hit my
> head hard against the limitations of the current WebRTC API and being
> forced to learn SDP to achieve some of the less common use cases, I'm
> feeling the pain. I am therefore happy to see that there is a proposal
> for a higher-level API with similarities to the Microsoft's CU-WebRTC
> proposal and I hope that together with Microsoft's input this can
> become a really useful API.
>
> What I would like to see, though, is a bit different from what you've
> proposed. In particular, the MediaFlowDescription object is the one
> object that I believe is supposed to enable JS developers to do "SDP
> hacking" without having to understand SDP. Unfortunately, the way in
> which it is currently written, this API doesn't help a JS developer
> much. There are properties in that object like "ssrc" that mean
> nothing unless you understand SDP.
>
> I would therefore like to recommend making the properties on the
> MediaFlowDescription more semantic - in particular giving them better
> names - such that a JS developer really doesn't have to understand SDP
> to create/change media flow descriptions. Can we find better names for
> id, transportId and ssrc that are more explicitly expressing what
> they stand for and when a JS developer would actually change them?
>
> It would be nice if the MediaFlowDescription was abstract enough such
> that in future it doesn't matter if SDP is actually underneath (with
> offer/answer), but that's not actually my main goal. What I'm after is
> an API that can be used without having to understand what is
> underneath.
>
> Thanks for listening and HTH,
> Silvia.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> Peter Thatcher <mailto:pthatcher@google.com>
> 17 June, 2013 8:57 AM
> Google is in full support of "Plan B" for encoding multiple media 
> sources in SDP, and would like to see the Plan A vs. Plan B decision 
> resolved soon.  Recently, though, a third option, called "NoPlan" has 
> been proposed, but it lacked the details of what a JS API would look 
> like for NoPlan.  Cullen asked to see such an API proposal, and so I 
> have worked with Emil to make one.  He will be adding it to the NoPlan 
> draft, but I will also include it in this email.
>
> Again, Google is in full support of "Plan B".  But if Plan A vs. Plan 
> B cannot be decided, then we support NoPlan with the following 
> additions to the WebRTC JS API as an option that allows implementing 
> either Plan A or Plan B  in Javascript.  And even if Plan A vs. Plan B 
> is resolved, these API additions would still be a big improvement for 
> those WebRTC applications that don't use SDP for signalling.
>
> It is a bit long because I have added many comments and examples, but 
> the actually additions only include two new methods on PeerConnection 
> and a few new dictionaries.  So don't be overwhelmed :).
>
>
>
> Intro: This follows the model of createDataChannel, which has a JS 
> method on PeerConnection that makes it possible to add data channels 
> without going through SDP.  Furthermore, just like createDataChannel 
> allows 2 ways to handle neogitation (the "I know what I'm doing; 
>  Here's what I want to send; Let me signal everything" mode and the 
> "please take care of it for me;  send an OPEN message" mode), this 
> also has 2 ways to handle negotiation (the "I know what I'm doing; 
> Here's what I want to send; Let me signal everything" mode and the 
> "please take care of it for me;  send SDP back and forth" mode).
>
> Following the success of createDataChannel, this allows simple 
> applications to Just Work and more advanced applications to easily 
> control what they need to.  In particular, it's possible to use this 
> API to implement either Plan A or Plan B.
>
> // The following two method are added to RTCPeerConnection
> partial interface RTCPeerConnection {
>  // Create a stream that is used to send a source stream.
>  // The MediaSendStream.description can be used for signalling.
>  // No media is sent until addStream(MediaSendStream) is called.
>  LocalMediaStream createLocalStream(MediaStream sourceStream);
>
>  // Create a stream that is used to receive media from the remote side,
>  // given the parameters signalled from MedaiSendStream.description.
>  MediaStream createRemoteStream(MediaStreamDescription description);
> }
>
>
> interface LocalMediaStream implements MediaStream {
>   // This can be changed at any time, but especially before calling
>   // PeerConnection.addStream
>   attribute MediaStreamDescription description;
> }
>
>
> // Represents the parameters used to either send or receive a stream
> // over a PeerConnection.
> dictionary MediaStreamDescription {
>   MediaStreamTrackDescription[] tracks;
> }
>
>
> // Represents the parameters used to either send or receive a track 
> over // a PeerConnection.  A track has many "flows", which can be grouped
> // together.
> dictionary MediaStreamTrackDescription {
>   // Same as the MediaStreamTrack.id
>   DOMString id;
>
>   // Same as the MediaStreamTrack.kind
>   DOMString kind;
>
>   // A track can have many "flows", such as for Simulcast, FEC, etc.
>   // And they can be grouped in arbitrary ways.
>   MediaFlowDescription[] flows;
>   MediaFlowGroup[] flowGroups;
> }
>
> // Represents the parameters used to either send or receive a "flow"
> // over a PeerConnection.  A "flow" is a media that arrives with a
> // single, unique SSRC.  One to many flows together make up the media
> // for a track.  For example, there may be Simulcast, FEC, and RTX
> // flows.
> dictionay MediaFlowDescription {
>   // The "flow id" must be unique to the track, but need not be unique
>   // outside of the track (two tracks could both have a flow with the
>   // same flow ID).
>   DOMString id;
>
>   // Each flow can go over its own transport.  If the JS sets this to a
>   // transportId that doesn't have a transport setup already, the
>   // browser will use SDP negotiation to setup a transport to back that
>   // transportId.  If This is set to an MID in the SDP, then that MID's
>   // transport is used.
>   DOMString transportId;
>
>   // The SSRC used to send the flow.
>   unsigned int ssrc;
>
>   // When used as receive parameters, this indicates the possible list
>   // of codecs that might come in for this flow.  For exmample, a given
>   // receive flow could be setup to receive any of OPUS, ISAC, or PCMU.
>   // When used as send parameters, this indicates that the first codec
>   // should be used, but the browser can use send other codecs if it
>   // needs to because of either bandwidth or CPU constraints.
>   MediaCodecDescription[] codecs;
> }
>
>
> dictionary MediaFlowGroup {
>   DOMString type;  // "SIM" for Simulcast, "FEC" for FEC, etc
>   DOMString[] flowids;
> }
>
> dictionary MediaCodecDescription {
>   unsigned byte payloadType;
>   DOMString name;
>   unsigned int? clockRate;
>   unsigned int? bitRate;
>   // A grab bag of other fmtp that will need to be further defined.
>   MediaCodecParam[] params;
> }
>
> dictionary MediaCodecParam {
>   DOMString key;
>   DOMString value;
> }
>
>
> Notes:
>
> - When LocalMediaStreams are added using addStream, onnegotiatedneeded 
> is not called, and those streams are never reflected in future SDP 
> exchanges.  Indeed, it would be impossible to put them in the SDP 
> without first resolving if that would be Plan A SDP or Plan B SDP.
>
> - Just like piles of attributes would need to be defined for Plan A 
> and for Plan B, similar attributes would need to be defined here 
> (Luckily,  much work has already been done figuring out what those 
> parameters are :).
>
>
> Pros:
>
> - Either Plan A or Plan B or could be implemented in Javascript using 
> this API
> - It exposes all the same functionality to the Javascript as SDP, but 
> in a much nicer format that is much easier to work with.
> - Any other signalling mechanism, such as Jingle or CLUE could be 
> implemented using this API.
> - There is almost no risk of signalling glare.
> - Debugging errors with misconfigured descriptions should be much 
> easier with this than with large SDP blobs.
>
>
> Cons:
>
> - Now there are two slightly different ways to add streams: by 
> creating a LocalMediaStream first, and not.  This is, however, 
> analogous to setting "negotiated: true" in createDataChannel.  On way 
> is "Just Work", and the other is more advanced control.
>
> - All the options in MediaCodecDescription are a bit complicated. 
>  Really, this is only necessary because Plan A requires being able to 
> specify codec parameters per SSRC, and set each flow on different 
> transports.  If we did not have this requirement, we could simplify.
>
>
> Example Usage:
>
> // Imagine I have MyApp, handles creating a PeerConnection,
> // signalling, and rendering streams.  This is how the new API could be
> // used.
> var peerConnection = MyApp.createPeerConnection();
>
> // On sender side:
> var stream = MyApp.getMediaStream();
> var localStream = peerConnection.createSendStream(stream);
> sendStream.description = MyApp.modifyStream(localStream.description)
> MyApp.signalAddStream(localStream.description, function(response)) {
>   if (!response.rejected) {
>     // Media will not be sent.
>     peerConnection.addStream(localStream);
>   }
> }
>
> // On receiver side:
> MyApp.onAddStreamSignalled = function(streamDescription) {
>   var stream = peerConnection.createReceiveStream(streamDescription);
>   MyApp.renderStream(stream);
> }
>
>
> // In this exchange, the MediaStreamDescription signalled from the
> // sender to the receiver may have looked something like this:
>
> {
>   tracks: [
>   {
>     id: "audio1",
>     kind: "audio",
>     flows: [
>     {
> id: "main",
>       transportId: "transport1",
>       ssrc: 1111,
>       codecs: [
>       {
>         payloadType: 111,
>         name: "opus",
>         // ... more codec details
>       },
>       {
>         payloadType: 112,
>         name: "pcmu",
> // ... more codec details
>       }]
>    }]
>  },
>  {
>     id: "video1",
>     kind: "video",
>     flows: [
>     {
>       id: "sim0",
>       transportId: "transport2",
>       ssrc: 2222,
>       codecs: [
>       {
>         payloadType: 122,
>         name: "vp8"
> // ... more codec details
>       }]
>    },
>    {
>      id: "sim1",
>      transportId: "transport2",
>      ssrc: 2223,
>      codecs: [
>      {
>        payloadType: 122,
>        name: "vp8",
> // ... more codec details
>      }]
>    },
>    {
>      id: "sim2",
>      transportId: "transport2",
>      ssrc: 2224,
>      codecs: [
>      {
>        payloadType: 122,
>        name: "vp8",
> // ... more codec details
>      }]
>    },
>
>    {
>      id: "sim0fec",
>      transportId: "transport2",
>      ssrc: 2225,
>      codecs: [
>      {
>        payloadType: 122,
>        name: "vp8",
>        // ...
>      }]
>    }],
>    flowGroups: [
>    {
>      semantics: "SIM",
>      ssrcs: [2222, 2223, 2224]
>    },
>    {
>      semantics: "FEC",
>      ssrcs: [2222, 2225]
>    }]
>  }]
> }
>
>
> Constructive feedback is welcome :).
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb