I have setup nginx in ubuntu machine for rtmp to hls streaming my webcam according to many articles that there are in the web but it has an unacceptable latency of 35-45 secs!!!
HLS is normally about 10s+ latency, and if you want ~100ms latency you MUST use WebRTC, but WebRTC is complex and not only about the good things.
Generally we want low latency, and we also want as more as posssible devices to publish and play streams, and we want CDN to support it if we want to scale out our system.
In short, if 10s latency is ok for you, never try to build a 3s latency system. If 3s is ok to you, never build a 100ms one. There is no free thing in this world.
The baseline latency about protocols:
HLS, about 5~10s latency, friendly to CDN.
HTTP-FLV, about 1~3s latency, only supported by some CDN.
WebRTC, about 0.5~1s latency, few of CDN support it.
SRT, about 0.3~0.5ms latency, only supported by encoder.
There are some issues about the latency:
About players for theses protocol, please read this.
How to benchmark the latency, please read this.
Use WebRTC to do live streaming, coverting RTMP to WebRTC, please read this.
If use FFmpeg, there is a post to reduce the latency.
So it really depends on the use scenario, for example:
If your users only use H5, you could use WebRTC or FLV.
If your users use OBS to publish stream, you could use SRS to covert RTMP to WebRTC, for viewers to play the stream.
If your users are using broadcasting devices, you could use SRT.
users --RTMP/SRT/WebRTC--> SRS --SRT/WebRTC/FLV --> Viewers
If you want more discussion, you could join the community of SRS.
Related
I have a client with a 1-2 thousand viewer audience, with everyday streams, same concurrent number of viewers.
Ive got a server set up for their website etc, but am in the process of figuring out the best way to stream with OBS onto that server, and than re-distribute that stream to clients (as an embed on the website).
Now from the calculations i did, running that kind of concurrent viewers is very problematic, as it forces you into a 10gbit link - which is very expensive, and i would ideally like to fit within 1-2gbps, if possible.
A friend of mine recommended to look into "Multicast" which supossedly uses MUCH less bandwith than regular live streaming options. Is multicast doable? Ive had a NGINX live stream set up on my server by a friend before, but never looked into the config and if multicast is supported within that. Are there any other options? What would you recommend?
Also, the service of that live stream isnt a high profit / organisation type of deal, so any pre-made services just dont make sense, as it would easily cost 40+ dollars per stream, which is just too much for my client.
Thank you for any help!
Tom
Rather than Multicast, P2P is more practical solution on Internet, to save money not bandwidth.
Especially for H5 browser, it's possible to use WebRTC DataChannel to transport P2P data.
But Multicast does not work on internet routers.
Multicast works by sending a single stream across the network to edge points where clients can 'join' the multicast to get an individual stream for them.
It requires that the network supports multicast protocols and the edges align with your users.
It is typically used when an operator has their own IP network for service like IPTV, rather than for services over the internet.
For your scenario, you would usually use an organ server and a CDN - this will usually reduce the load on your own server as the video will be cached on the network and multiple user can access the same 'chunks' of the video.
You can see and AWS example for on demand video here - other vendor and cloud providers have solutions too so this is just an example:
https://docs.aws.amazon.com/AmazonS3/latest/userguide/tutorial-s3-cloudfront-route53-video-streaming.html
You can find more complex On Demand and Live tutorial also but they are likley more that you need: https://aws.amazon.com/cloudfront/streaming/
Exploring P2P may be an option also as Winton suggests - some CDN may also leverage P2P technology internally.
Based on this article, when implementing a WebRTC solution without a server, I assume it means SFU, the bottleneck is that only 4-6 participants can work.
Is there a solution that can work around this? For example, I just want to use Firebase as the only backend, mainly signaling and no SFU. What is the general implementation strategy to achieve at least 25-50 participants in WebRTC?
Update: This Github project shares a different statement. It states "A full mesh is great for up to ~100 connections"
Your real bottleneck with MESH is that each RTCPeerConnection will do its own video encoding in the browser.
The p2p concept naturally includes the requirement that both peers should adjust encoding quality based on network conditions. So, when your browser sends two streams to peers X (good download speed) and Y (bad download speed), the encodings for X and Y will be different - Y will receive lower framerate and bitrate than X.
Sounds reasonable, right? But, unfortunately, mandates separate video encoding for each peer connection.
If multiple peer connections could re-use the same video encoding, then MESH would be much more viable. But Google didn't provide that option in the browser. Simulcast requires SFU, so that's not your case.
So, how many concurrent video encodings can browser perform on a typical machine, for 720p 30 fps video? 5-6, not more. For 640x480 15 fps? Maybe 20 encodings.
In my opinion, the encoding layer and networking layer could be separated in WebRTC design, and even getUserMedia could be extended to getEncodedUserMedia, so that you could send the same encoded content to multiple peers.
So that's the real practical reason people use SFU for multi-peer WebRTC.
If you want to make a conference with 25 people all sending their video, then a regular webrtc setup will not work. Except if you massively lower your video quality. The reason for this is that every participant would need to send 24 seperate streams to every other client. So lets say you stream is 128 KB/s then you will need to have 3MB/s in upload speed available. Which isn't always available. Then also downloading that same amount.
The problem is that isn't scalable. That's why you need an SFU. Then you will only send a single stream and receive from others. The other positive thing about SFUs is that you can use simulcast which adapts the quality of your received streams depending on your network speed.
You can use the Janus gateway or mediasoup for example. Here is an already setup mediasoup video conferencing application that is scalable github repository
I understand in a client-server model gRPC can do a bidirectional streaming of data.
I have not tried yet, but want to know will it be possible to stream audio and video data from a source to cloud server using gRPC and then broadcast to multiple client, all in real time ?
TLDR: I would not recommend video over gRPC. Primarily because it wasn't designed for it, so doing it would take a lot of hacking. You should probably take a look at WebRTC + a specific video codec.
More information below:
gRPC has no video compression
When sending video, we want to send things efficiently because sending it raw could require 1GB/s connectivity.
So we use video compression / video encoding. For example, H.264, VP8 or AV1.
Understand how video compression works (eg saving bandwidth by minimising similar data shared between frames in a video)
There is no video encoder for protobufs (the format used by gRPC).
You could then try image compression and save the images in a bytes field (e.g. bytes image_frame = 1;, but this is less efficient and definitely takes up unnecessary space for videos.
It's probably possible to encode frames into protobufs using a video encoder (e.g. H.264) and then decode them to play in applications. However, it might take a lot of hacking/engineering effort. This use case is not what gRPC/protobufs is designed for and not commonly done. Let me know if you hack something together, I would be curious.
gRPC is reliable
gRPC uses TCP (not UDP), which is reliable.
At a glance, reliability might be handy, to avoid corrupting data or lost data. However, depending on the use case (realtime video or audio calls), we may prefer to skip frames if they are dropped or delayed. The losses may be unnoticeable or painless to the user.
If the packet is delayed, it will wait for the packet before playing the rest. (aka. out of order delivery)
If the packet is dropped, it will resend it (aka. packet loss)
Therefore, video conferencing apps usually use WebRTC/RTP (configured to be unreliable)
Having mentioned that, looks like Zoom was able to implement Video-over-WebSockets, which is also a reliable transport (over TCP). So it's not "game-over", just highly not recommended and a lot more effort. They have moved over to WebRTC though.
Data received on the WebSockets goes into a WebAssembly (WASM) based decoder. Audio is fed to an AudioWorklet in browsers that support that. From there the decoded audio is played using the WebAudio “magic” destination node.
I've tried the Wowza streaming engine but even at low video quality there's a 3 second delay. Are there any standard way to set this up with minimal delay?
The technology and standards certainly exist. Look at video conferencing: minimal delay, perfect A/V sync and great for changing network condition.
Apple's FaceTime is a prime example.
I doubt the delay is in the up-link phone to Wowza - more likely in the transcoding and packaging.
Every re-streamer (Wowza and others) use DASH or HLS to get to the client which makes your video stream look like lot's of little files with a 1-3 second duration. This leverages existing cache and CDN infrastructure but introduces seconds of delay.
If your target delay is below a few 100 milliseconds you have to do something like RTP/RTSP.
I'm trying to figure out the best way to stream video and I'm stuck on choosing my technology stack. Eventually what I want is to stream video (for now it's strictly h264 streaming) as efficient as possible. I'm new to streaming video and am quite confused choosing the transport protocol. I want to implement a streaming service where the packet loss is favored over latency, what would be the best transport protocol I could use?
I've been working with RTMP and still can't get the latency less than 30sec for 720p via 50mbit wifi connection. I couldn't find any performance comparisons between transport protocols so either I haven't been looking in the right places or no such comparison can be done without specifying the information about the data being transfered. In any case, could anyone shed some light on the issue?
I don't know if it makes any difference but I'm using NGINX-based media streaming server, and I'm open to any other suggestions for the media server.
Thanks
For low latency you could choose a protocol like RTP/RTSP which is intended for realtime. Now if you want to further reduce latency - turn off B-frames on your H.264 encoder.
WebRTC may be another option.
A) RTMP
Drawback:
1) For CDN, It will be costly since it work without caching.
2) From point of view of security, RTMP Encrypted is somewhat flawed, as it is prone to Man-in-the-Middle Attacks. RTMP toolkit - RTMPDump can be used easily to download RTMP streams.
3) RTMP streams your video in chunks it never loads ahead of time and viewers on slow speed connections might experience a bit of buffering as they wait for the rest of the file to stream.
4) You could use rtmp only with Flash plugin.
5) RTMP works on port 1935 only (Blocked by most corporate).
B)HLS (from Apple)Adaptive Streaming
Can be used with live + stored video streaming
Drawback:
This is platform dependent(good for IOS only)
C) Progressive /Pseudo Streaming
Download video in chunks before request.
Platform Independent.
Supports all video formats and served from http protocol.
Use pre trans-coded video to work with par with adaptive streaming.
**Nginx has support for all above methods, ngx_http_mp4_module