I have a server that currently streams out RTMP media and I need to play it with minimum delay (2-3 seconds).
Now I'm using CRTMPServer to consume the stream but, since there is no RTMP player without flash dependency, this solutions does not work for me because I need to play it embedded into a web page.
At this point, I tried Nimble Streamer but I only could consume HLS, DASH and SLDP. HLS and DASH has 10 second delay so I discard these approaches.
SLDP has low latency but is a closed protocol and there is only one player (closed but freeware).
On the other hand, there is RTSP but there are no HTML5 players for it. But I couldn't consume RTSP streams from Nimble neither.
Any other ideas?
I do not know why you discarded Nimble Streamer, I've tried it and it has a delay of 2 or 3 seconds.
My tests were with more than 200 attendees in Full HD (1280)
And the results were incredible, it has relatively low consumption of resources, the only thing you have to take into account is the bandwidth.
They have a protocol that allows them to have a delay of 2 or 3 seconds.
Look:
https://es.wmspanel.com/nimble/sldp
So Nimble Stream seems like a good option.
regards
I'm Yury from Softvelum, which is the company behind Nimble Streamer and SLDP technology.
You mention
SLDP has low latency but is a closed protocol and there is only one
player (closed but freeware).
We currently have 3 players - HTML5/JavaScript, Android and iOS, they all are being actively developed at the moment. All 3 players have their SDKs which means you can extend their behavior in any way you like and use in your own apps and web pages. We currently communicate with other existing players in order to extend the reach of SLDP technology.
At the moment when we started creating the SLDP architecture, there were no publicly available technologies to cover low-latency last-mile scenarios while our customers strove for that all the time. That's why we implemented SLDP and our goal is not just to provide the technology but to make it available for broader audience. We cannot accomplish this by ourselves alone, hence the SDK and other initiatives. We're opened for collaboration with any other companies on this market.
If SLDP works fine for your use case, don't hesitate to use further - we have big plans for it and you can be sure it will be supported and improved by our company. Of course, your feedback is highly welcomed.
You can install one webserver with RTMP module like
Nginx+RTMP module
If have Windows 10 Pro then can use Ubuntu WSL(linux for windows) 18.04 bionic
Then can use OBS to can streaming your laptop's desktop or your laptop's webcam or other media resurse to Nginx(which have RTMP module installed)
Then as player=render you can use VLC player or can push the stream to FB,YT ... or other webservice live
OBS--->>--->NgInx(+RTMP module)--->>-->VLC player or FB,YT live
Related
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?
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.
I've got a website and I've been looking for ways to embed a 24/7 webcast. I've looked at options such as Ustream and Justin.TV however, these do not work on mobile devices, which is what I really need.
I don't have that much knowledge on how streaming works but I've read that the streaming Engine Wowza is another option. I also found that HTML 5 player works cross platform and on any mobile device aswell.
If I were to use Wowza would it work with HTML 5 player? And am I even going in the right path with how I can do this. I also have a home dedicated server for streaming to a cloud wouldn't be required.
I'm very amateur just trying to broadcast my television program on my website for viewing. Any advice would help here. Thanks
Wowza can packetize video as http live streaming (HLS) which, although an Apple invention, works on most HTML5-capable browsers except IE11: http://www.jwplayer.com/html5/hls/ . Many players will fall back to using Flash for browsers which don't support native HLS or H.264 encoding. Flash uses http dynamic streaming (HDS) rather than HLS, so you would add that as another packetizer in wowza. (Wowza calls these packetizers "cupertinostreamingpacketizer" and "sanjosestreamingpacketizer" respectively.)
You would then point your preferred HTML5 video player (jwplayer, flowplayer, etc) at the URL http:// your-wowza-server.com:1935/live/yourstreamname/playlist.m3u8 [1]. For Flash fallback in flowplayer you can use the f4m resolver and the http-streaming plugin, as in the first example here, to access the subtly different URL http:// your-wowza-server.com:1935/live/yourstreamname/manifest.f4m. I'm sure something similar applies in players like jwplayer and others.
The main problem with Wowza is how much it costs: for your own server you're looking at around $55 per month per channel [2]. At least during testing, you may find it cheaper to get Wowza on Amazon EC2 devpay: $5/month rental plus an extra couple of cents per hour on your normal EC2 instance costs.
[1] Assuming you're using Wowza's default /live/ application on port 1935
[2] A channel is roughly the number of streams you're sending to the server to be re-broadcast
We developped a custom HTML5 player which we wanted to make compatible with HLS and fragmented mp4 for LIVE events. We started on Zencoder but realize they were not able to do genrate fragmented mp4.
I would like to explore the flash fallback solution and the wowza( probably on AWS) for the packaging.
Would you be available to consult on this project?
We use www.bitcodin.com for event-based or 24/7 live transcoding and streaming. It generates DASH - which can be playback natively in HTML5 using the bitdash MPEG-DASH players - as well as HLS for iOS devices. You can find an example here: http://www.dash-player.com/demo/live-streaming/
we are currently planning to connect all our offices via webcams selectable from a website. So I guess the setup could be something like this:
Webcam 1 \
Webcam 2 - > Streamserver -> Webserver displaying html5 Video
Webcam n /
I've made first tests from streaming from Windows using VLC to a central ffserver, but the results where rather bad (after 1-2 frames the stream stopped).
Do you have any suggestions on
- which stream software I can use on the client/webcam site (Linux or Windows)
- which software I can use on the server side (Linux) for collecting and recoding streams?
Thanks,
Moritz
I would recommend using WebRTC instead of VLC & ffserver.
All you will need for this implementation is a central node server.
And if you did not want to go the webRTC rout you could always just stream right over a socket, but I really think webRTC is the way to go.
from scratch :
http://www.html5rocks.com/en/tutorials/webrtc/basics/
using something made to make it a little simpler :
http://simplewebrtc.com/
What is the best solution to display multiple live streams from a surveillance camera (so low latency is a requirement) in a web application (VideoWall-like)?
Personally I'm thinking about two possible solutions, but I can't choose between them:
1) Develop a custom Firefox plugin that uses ffmpeg to acquire and decode the video streams
2) Rely on HTML5 inserting a layer between the cameras and the web application that transcodes/restreams the video streams (probably using http live streaming)
The requirements are compatibility with H.264 over RTSP and Mpeg-4 over RTP and, of course, low latency and no loss on video quality.
Thanks
Andrea