Getting WebRTC performance indication - google-chrome

We've developed a professional WebRTC application and are trying to give users an indication of how many streams their PC can handle (2-7). Is there an easy way of figuring this out (in browser or with a separate application)?
It's a conference application we offer to users browsing with Chrome.
Another question, if you work with for example 7 streams, are they divided over the different CPU cores? Or is the whole WebRTC deal included in the process for that browser tab?

WebRTC makes extensive use of threads, so it can utilize more than one core especially in multi-party conferences.
The simplest way to check is to make calls to yourself (each one = 2 calls in a mesh conference). If it's an MCU-style conference (likely with 7 participants), you need to simulate a one-way call (so you're doing one encode), plus decode N additional VP8 streams at "appropriate" resolutions.
This is complicated by Firefox, for example, using content analysis to selectively reduce resolution and/or frame rate of sent-video depending on load and outgoing bandwidth. However, for your case, it's more one of reception.
The short answer, though, is that it's hard to be sure, and will depend on the other senders too.

Related

Basic architecture to serve, stream and consume large audio files to minimize client-side resource consumption and latency

I am trying to build a web application which will need to have audio streaming functionality implemented in some way. Just to give you guys some context: It is designed to be a purely auditive experience/game/idkhowtocallit with lots of different sound assets varying in length and thus file size. The sound assets to be provided will consist of ambient sounds, spoken bits of conversation, but also long music sets (up to a couple of hours). Why I think I won't be able to just host these audio files on some server or CDN and serve them from there is, because the sound assets will need to be fetched and played dynamically (depending on user interaction) and as instantly as possible.
Most importantly, consuming larger files (like the music sets and long ambient loops) as a whole doesn't seem to be client-friendly at all to me (used data consumption on mobile networks and client-side memory usage).
Also, without any buffering or streaming mechanism, the client won't be able to start playing these files before they are downloaded completely, right? Which would add the issue of high latencies.
I've tried to do some online research on how to properly implement a good infrastructure to stream bigger audio files to clients on the server side and found HLS and MPEG-DASH. I have some experience with consuming HLS players with web players and if I understand it correctly, I would use some sort of one-time transformation process (on or after file upload) to split up the files into chunks and create the playlist and then just serve these files via HTTP. From what I understand the process should be more or less the same for MPEG-DASH. My issue with these two techniques is that I couldn't really find any documentation on how to implement JavaScript/TypeScript clients (particularly using the Web Audio API) without reinventing the wheel. My best guess would be to use something like hls.js and bind the HLS streams to freshly created audio elements and use these elements to create AudioSources in my Web Audio Graph. How far off am I? I'm trying to get at least an idea of a best practice.
To sum up what I would really appreciate to get some clarity about:
Would HLS or MPEG-DASH really be the way to go or am I missing a more basic chunked file streaming mechanism with good libraries?
How - theoretically - would I go about limiting the amount of chunks downloaded in advance on the client side to save client-side resources, which is one of my biggest concerns?
I was looking into hosting services as well, but figured that most of them are specialized in hosting podcasts (fewer but very large files). Has anyone an opinion about whether I could use these services to host and stream possibly 1000s of files with sizes ranging from very small to rather large?
Thank you so much in advance to everyone who will be bothered with helping me out. Really appreciate it.
Why I think I won't be able to just host these audio files on some server or CDN and serve them from there is, because the sound assets will need to be fetched and played dynamically (depending on user interaction) and as instantly as possible.
Your long running ambient sounds can stream, using a normal HTMLAudioElement. When you play them, there may be a little lag time before they start since they have to begin streaming, but note that the browser will generally prefetch the metadata and maybe even the beginning of the media data.
For short sounds where latency is critical (like one-shot user interaction sound effects), load those into buffers with the Web Audio API for playback. You won't be able to stream them, but they'll play as instantly as you can get.
Most importantly, consuming larger files (like the music sets and long ambient loops) as a whole doesn't seem to be client-friendly at all to me (used data consumption on mobile networks and client-side memory usage).
If you want to play the audio, you naturally have to download that audio. You can't play something you haven't loaded in some way. If you use an audio element, you won't be downloading much more than what is being played. And, that downloading is mostly going to occur on-demand.
Also, without any buffering or streaming mechanism, the client won't be able to start playing these files before they are downloaded completely, right? Which would add the issue of high latencies.
If you use an audio element, the browser takes care of all the buffering and what not for you. You don't have to worry about it.
I've tried to do some online research on how to properly implement a good infrastructure to stream bigger audio files to clients on the server side and found HLS and MPEG-DASH.
If you're only streaming a single bitrate (which for audio is usually fine) and you're not streaming live content, then there's no point to HLS or DASH here.
Would HLS or MPEG-DASH really be the way to go or am I missing a more basic chunked file streaming mechanism with good libraries?
The browser will make ranged HTTP requests to get the data it needs out of the regular static media file. You don't need to do anything special to stream it. Just make sure your server is configured to handle ranged requests... most any should be able to do this right out of the box.
How - theoretically - would I go about limiting the amount of chunks downloaded in advance on the client side to save client-side resources, which is one of my biggest concerns?
The browser does this for you if you use an audio element. Additionally, data saving settings and the detected connectivity speed may impact whether or not the browser pre-fetches. The point is, you don't have to worry about this. You'll only be using what you need.
Just make sure you're compressing your media as efficiently as you can for the required audio quality. Use a good codec like Opus or AAC.
I was looking into hosting services as well, but figured that most of them are specialized in hosting podcasts (fewer but very large files). Has anyone an opinion about whether I could use these services to host and stream possibly 1000s of files with sizes ranging from very small to rather large?
Most any regular HTTP CDN will work just fine.
One final note for you... beware of iOS and Safari. Thanks to Apple's restrictive policies, all browsers under iOS are effectively Safari. Safari is incapable of playing more than one audio element at a time. If you use the Web Audio API you have more flexibility, but the Web Audio API has no real provision for streaming. You can use a media element source node, but this breaks lock screen metadata and outright doesn't work on some older versions of iOS. TL;DR; Safari is all but useless for audio on the web, and Apple's business practices have broken any alternatives.

Web Audio Streaming/Seeking (Mobile + Desktop) with Tracking

It's been a long time since I've needed to build a streaming media solution for the web since there are so many good services providing basic needs. Ages ago I had done streaming deployments with Red5 and a frontend player like Flow or JW, but despite a lot of searching I haven't been able to determine what the best modern options are to achieve a simple web solution that has good native compatibility with mobile and desktop. I'm hoping someone can suggest a good open source stack that would be a good fit for what I'm trying to do:
Mobile + Desktop Web Audio Support (with the possibility of some video at some point)
Streaming (not live on-air streaming, but the ability to seek and listen without downloading the whole file and only buffering enough to for good quality as to not waste bandwidth)
Track how much of a stream has been listened to (not the amount downloaded)
Protecting streams (this should be pretty easy either using headers or URL patterns that expire the link/connection after a certain point of time by writing that logic into the server side but I wanted to mention it anyways)
Lightweight, maybe 10 to 20 concurrent audio streams for now.
My initial thought, since my knowledge in the streaming space is dated, was to do it with RTMP, RTSP or HLS and something like JWPlayer for the frontend, depending on which protocol seemed to have the best support across the most devices. Or, if the client side support is good I could probably skip the media server and go with psuedo-streaming via the webserver. On the server side I could handle the access control to the streams and on the front end player I could use the API to keep track of the amount of seconds played and just ping that data in a ajax request to keep track of how much has actually been listened too (as opposed to downloaded/buffered).
The thing is I have a feeling that is probably an obsolete way to go about it in 2017. Even though I haven't turned up anything specific yet, I had a feeling perhaps there was a better solution out there utilizing Node and perhaps websockets to accomplish the same thing with a server and player coupled more tightly (server aware of playhead/actual amount played vs front end only) or a solution that leverages newer standards (MPEG-DASH?). Plus, in the past getting the other streaming protocols to be compatible across all devices was a pain and IIRC there is no universal HTML5 support for any given one of the older protocols across all browsers.
So what are the best current approaches to tackle something like this? Is a media server + a front end like JW or Flow still one of the better ways to go or are there better/easier ways to deploy a solution like this?

Low latency (< 2s) live video streaming HTML5 solutions?

With Chrome disabling Flash by default very soon I need to start looking into flash/rtmp html5 replacement solutions.
Currently with Flash + RTMP I have a live video stream with < 1-2 second delay.
I've experimented with MPEG-DASH which seems to be the new industry standard for streaming but that came up short with 5 second delay being the best I could squeeze from it.
For context, I am trying to allow user's to control physical objects they can see on the stream, so anything above a couple of seconds of delay leads to a frustrating experience.
Are there any other techniques, or is there really no low latency html5 solutions for live streaming yet?
Technologies and Requirements
The only web-based technology set really geared toward low latency is WebRTC. It's built for video conferencing. Codecs are tuned for low latency over quality. Bitrates are usually variable, opting for a stable connection over quality.
However, you don't necessarily need this low latency optimization for all of your users. In fact, from what I can gather on your requirements, low latency for everyone will hurt the user experience. While your users in control of the robot definitely need low latency video so they can reasonably control it, the users not in control don't have this requirement and can instead opt for reliable higher quality video.
How to Set it Up
In-Control Users to Robot Connection
Users controlling the robot will load a page that utilizes some WebRTC components for connecting to the camera and control server. To facilitate WebRTC connections, you need some sort of STUN server. To get around NAT and other firewall restrictions, you may need a TURN server. Both of these are usually built into Node.js-based WebRTC frameworks.
The cam/control server will also need to connect via WebRTC. Honestly, the easiest way to do this is to make your controlling application somewhat web based. Since you're using Node.js already, check out NW.js or Electron. Both can take advantage of the WebRTC capabilities already built in WebKit, while still giving you the flexibility to do whatever you'd like with Node.js.
The in-control users and the cam/control server will make a peer-to-peer connection via WebRTC (or TURN server if required). From there, you'll want to open up a media channel as well as a data channel. The data side can be used to send your robot commands. The media channel will of course be used for the low latency video stream being sent back to the in-control users.
Again, it's important to note that the video that will be sent back will be optimized for latency, not quality. This sort of connection also ensures a fast response to your commands.
Video for Viewing Users
Users that are simply viewing the stream and not controlling the robot can use normal video distribution methods. It is actually very important for you to use an existing CDN and transcoding services, since you will have 10k-15k people watching the stream. With that many users, you're probably going to want your video in a couple different codecs, and certainly a whole array of bitrates. Distribution with DASH or HLS is easiest to work with at the moment, and frees you of Flash requirements.
You will probably also want to send your stream to social media services. This is another reason why it's important to start with a high quality HD stream. Those services will transcode your video again, reducing quality. If you start with good quality first, you'll end up with better quality in the end.
Metadata (chat, control signals, etc.)
It isn't clear from your requirements what sort of metadata you need, but for small message-based data, you can use a web socket library, such as Socket.IO. As you scale this up to a few instances, you can use pub/sub, such as Redis, to distribution messaging throughout the servers.
To synchronize the metadata to the video depends a bit on what's in that metadata and what the synchronization requirement is, specifically. Generally speaking, you can assume that there will be a reasonable but unpredictable delay between the source video and the clients. After all, you cannot control how long they will buffer. Each device is different, each connection variable. What you can assume is that playback will begin with the first segment the client downloads. In other words, if a client starts buffering a video and begins playing it 2 seconds later, the video is 2 seconds behind from when the first request was made.
Detecting when playback actually begins client-side is possible. Since the server knows the timestamp for which video was sent to the client, it can inform the client of its offset relative to the beginning of video playback. Since you'll probably be using DASH or HLS and you need to use MCE with AJAX to get the data anyway, you can use the response headers in the segment response to indicate the timestamp for the beginning the segment. The client can then synchronize itself. Let me break this down step-by-step:
Client starts receiving metadata messages from application server.
Client requests the first video segment from the CDN.
CDN server replies with video segment. In the response headers, the Date: header can indicate the exact date/time for the start of the segment.
Client reads the response Date: header (let's say 2016-06-01 20:31:00). Client continues buffering the segments.
Client starts buffering/playback as normal.
Playback starts. Client can detect this state change on the player and knows that 00:00:00 on the video player is actualy 2016-06-01 20:31:00.
Client displays metadata synchronized with the video, dropping any messages from previous times and buffering any for future times.
This should meet your needs and give you the flexibility to do whatever you need to with your video going forward.
Why not [magic-technology-here]?
When you choose low latency, you lose quality. Quality comes from available bandwidth. Bandwidth efficiency comes from being able to buffer and optimize entire sequences of images when encoding. If you wanted perfect quality (lossless for each image) you would need a ton (gigabites per viewer) of bandwidth. That's why we have these lossy codecs to begin with.
Since you don't actually need low latency for most of your viewers, it's better to optimize for quality for them.
For the 2 users out of 15,000 that do need low latency, we can optimize for low latency for them. They will get substandard video quality, but will be able to actively control a robot, which is awesome!
Always remember that the internet is a hostile place where nothing works quite as well as it should. System resources and bandwidth are constantly variable. That's actually why WebRTC auto-adjusts (as best as reasonable) to changing conditions.
Not all connections can keep up with low latency requirements. That's why every single low latency connection will experience drop-outs. The internet is packet-switched, not circuit-switched. There is no real dedicated bandwidth available.
Having a large buffer (a couple seconds) allows clients to survive momentary losses of connections. It's why CD players with anti-skip buffers were created, and sold very well. It's a far better user experience for those 15,000 users if the video works correctly. They don't have to know that they are 5-10 seconds behind the main stream, but they will definitely know if the video drops out every other second.
There are tradeoffs in every approach. I think what I have outlined here separates the concerns and gives you the best tradeoffs in each area. Please feel free to ask for clarification or ask follow-up questions in the comments.

Increasing Google Chrome's max-connections-per-server limit to more than 6

As far as I know, at the current moment, late 2011 the max-connections-per-server limit remains 6. Please correct me if I am wrong. This is bad that we cannot fix this easily as in Firefox. As far as I know this value is hardcoded.
One of the solutions is to download the Chromium's sources and rebuild them. Is there a more easy solution?
Is there any tricky way to hack this without creating a dozen of mirror-domains?
Why I'm asking the question: My task is to create a html-javascript slideshow that will run inside a fullscreened browser, and a huge monitor is hanging on the wall. The javascript is really complicated, it preloads photos and makes a lot of ajax calls to my web services. If WIFI connection is slow, if 6 photos are loading, the AJAX calls fail, the application runs bad. I want a fast solution based, on http or browser or ubuntu tweak something else, because rebuilding the javascript app will take days.
Offtopic: do you know any other things that can be tweaked in my concrete situation?
IE is even worse with 2 connection per domain limit. But I wouldn't rely on fixing client browsers. Even if you have control over them, browsers like chrome will auto update and a future release might behave differently than you expect. I'd focus on solving the problem within your system design.
Your choices are to:
Load the images in sequence so that only 1 or 2 XHR calls are active at a time (use the success event from the previous image to check if there are more images to download and start the next request).
Use sub-domains like serverA.myphotoserver.com and serverB.myphotoserver.com. Each sub domain will have its own pool for connection limits. This means you could have 2 requests going to 5 different sub-domains if you wanted to. The downfall is that the photos will be cached according to these sub-domains. BTW, these don't need to be "mirror" domains, you can just make additional DNS pointers to the exact same website/server. This means you don't have the headache of administrating many servers, just one server with many DNS records.
I don't know that you can do it in Chrome outside of Windows -- some Googling shows that Chrome (and therefore possibly Chromium) might respond well to a certain registry hack.
However, if you're just looking for a simple solution without modifying your code base, have you considered Firefox? In the about:config you can search for "network.http.max" and there are a few values in there that are definitely worth looking at.
Also, for a device that will not be moving (i.e. it is mounted in a fixed location) you should consider not using Wi-Fi (even a Home-Plug would be a step up as far as latency / stability / dropped connections go).
BTW, HTTP 1/1 specification (RFC2616) suggests no more than 2 connections per server.
Clients that use persistent connections SHOULD limit the number of simultaneous connections that they maintain to a given server. A single-user client SHOULD NOT maintain more than 2 connections with any server or proxy. A proxy SHOULD use up to 2*N connections to another server or proxy, where N is the number of simultaneously active users. These guidelines are intended to improve HTTP response times and avoid congestion.
There doesn't appear to be an external way to hack the behaviour of the executables.
You could modify the Chrome(ium) executables as this information is obviously compiled in. That approach brings a lot of problems with support and automatic upgrades so you probably want to avoid doing that. You also need to understand how to make the changes to the binaries which is not something most people can pick up in a few days.
If you compile your own browser you are creating a support issue for yourself as you are stuck with a specific revision. If you want to get new features and bug fixes you will have to recompile. All of this involves tracking Chrome development for bugs and build breakages - not something that a web developer should have to do.
I'd follow #BenSwayne's advice for now, but it might be worth thinking about doing some of the work outside of the client (the web browser) and putting it in a background process running on the same or different machines. This process can handle many more connections and you are just responsible for getting the data back from it. Since it is local(ish) you'll get results back quickly even with minimal connections.

Using WebGL from inside a Web Worker: is it possible ? How?

I opened this matrix multiplication benchmarks and my browser (Firefox 7.0.1) froze until the benchmarks finished (I opened the page in an old Asus EeePC 1000H).
I heard that web workers were invented to separate processing from displaying the web pages. Is it possible to make use of the Web Workers API to make WebGL not stall the whole web browser ?
For the sake of clarity: the benchmark that you linked to does not use WebGL at all. (I should know, I wrote it.) And in the case of that particular benchmark you absolutely could run it in a Web Worker now and it would be perfectly fine.
(Fun fact - Web Workers didn't support TypedArrays when the benchmark was built, and since most of the matrix libraries rely on that it was impractical to run it in a Worker at that time. That has since been fixed.)
Anyway, to answer your original question: No, WebGL cannot run in a worker. The core blocker to this is that in order to get a WebGL context you need to call getContext on a canvas element. Web Workers explicitly disallow DOM access (which is a good thing, BTW!) and as such you'll never be able to access WebGL from a worker.
But that's not as bad as you might think. For one, consider that most all 3D rendering is actually happening in a different thread anyway. Specifically, a whole bunch of threads running on your GPU. The only part the browser has in it is to tell your graphics driver "Hey! Start rendering some triangles using this data!" and then it moves on without waiting for the triangles to actually be rendered. As such, while the draw commands must be executed from the main process, the time it spends blocking that process is (usually) very little.
Of course, that's not what's going to eat up a bunch of your time if you were coding a realtime game. You've got animations, physics, AI, collision detection, pathfinding... there's a lot of non-graphical tasks involved that will eat your CPU alive if you let them. In some case (animation), it's usually just gobs and gobs of matrix math, just like the benchmark you linked to! Fortunately for us, however, that type of processing CAN be done in a Worker, and all we need to communicate back to the main thread is the data required to render the scene.
Yes, this introduces some challenges in terms of synchronization and data transfer, but on the whole it will be vastly preferable to locking up your browser while we try and simulate those 500 boxes colliding.
Yes, on Firefox!
https://hacks.mozilla.org/2016/01/webgl-off-the-main-thread/
We’re happy to announce WebGL in Web Workers in Firefox 44+! Using the new OffscreenCanvas API you can now create a WebGL context off of the main thread.
By default you can't use WebGL in a Web Worker as Toji explained.
You can check out WebGLWorker which is a library that lets you do WebGL stuff in a Web Worker, by transparently proxying commands to the main thread.
Here is a nice blog post that explains how it works.