I am new to WebRTC technology.
I want to create a video chat / video conferencing with a transmitter and many followers (more 1000).
Example:
I read a lot of documentations :
https://medium.com/linagora-engineering/scalability-in-video-conferencing-part-1-276f52b4acac
https://webrtcglossary.com/sfu/
But I still don't know what is the best solution (in my case) between Selective Forwarding Unit (SFU) and Multiploint Control Unit (MCU).
Can you help me to understand?
I think the best way is MCU but I am not sure.
Second question:
Can you suggest some sources and links that can help me to set up such an architecture. Currently my project works perfectly in Peer To Peer (Mesh) but it is not the right solution. I have absolutely no idea how to set this up.
Thank you so much
It is possible to implement this using an SFU. The more peers are connected, the more you would need processing power to handle those new peers. This could be done by using more threads and/or forwarding requests to another machine.
With mediasoup it is possible have control over this. With this tool you have routers where peers can connect to to get the stream. A router works on a worker which has a limited amount of receiving peers (depending on cpu capacity). Now to allow more peers you can forward the stream to other routers which can expand the total capacity.
useful links:
https://mediasoup.org/documentation/v3/scalability/#one-to-many-broadcasting
https://mediasoup.org/documentation/v3/mediasoup/design/#architecture
https://mediasoup.discourse.group/t/scalability-in-mediasoup-example/793/2?u=dirvann
Related
I am new to webrtc development, I would like to create a conference room with multiples connections. I have read about webrtc and peer communication. I would like to know if google meet used one peer connection for every participants one another. If so, how can they handle 250 participants without draining the browser resources?
For group calls, Google Meets does not use peer-to-peer, if that was the case, it would drain one's bandwidth very quickly and browser resources quickly.
Although I am not exactly sure about Meets' infrastructure, they most likely use a Selective Forwarding Unit (SFU) to be able to handle lots of connections, since it's common practice for this sort of application.
The SFU works as a webrtc client server-side, and it basically just forwards the stream to clients that are requesting it browser-side.
Does Google WebRTC Native implementation has support for SFU?
Does Google WebRTC Native implementation support for integrating custom/hardware encoder/decoder?
Not without alteration.
Internally WebRTC's internal audio/video pipelines are directly tied to encoder/decoders.
PeerConnectionFactory allows you to provide a video decoder/encoder factory, so you can short circuit the logic here, and grab the encoded frames, mock up a stream, and feed them directly into it as a relay, creating a new PeerConnection and setting those streams onto it.
The audio end is more difficult. There isn't a codec factory, so you will have to short circuit the logic there probably by alteration of libwebrtc.
The final question is RTCP termination, and how to override the mechanisms for quality/bandwidth control to not create a "One goes out, they all go out." situation.
Since libwebrtc will be the SFU, it will receive RTCP feedback from its remote peer for the content it is proxying, and vice versa.
For a 1-1 situation, it needs to be able to forward the RTCP feedback to the remote peer.
For multipoint, it needs to perform some logic to determine if one of the peers is problematic, and stop sending it video, switch off its video feed, or attempt to switch to a lower bitrate video stream. Basically it needs to act as a conduit that attempts to predict why/how packet loss is occurring, and keep as many audio/video feeds operating normally at at the highest possible quality for each peer.
How exactly to hijack the RTCP feedback mechanisms in libwebrtc, I think that again will likely require some customization/hooks into libwebrtc
I think it will be easier to try with GStreamer implementation of WebRTC. Although it is still in "Bad Plugins" it is way easier to get or provide encoded audio and video. Actually it is implemented in that in mind - to make implementation of MFU and SFU easier.
I have a question regarding how to enable Multihop in LoRa (that is to communicate between two end devices without the LoRaWAN gateway). I have tried doing it using transparent bridging but it won't work.
Although it works with LoRaBlink the issue is flooding. If the number of devices increases the channel utilization as well as the performance goes down rapidly.
Can someone please suggest if there is any other way to do it or how to do it efficiently through LoRaBlink?
Thanks
If you check the wiki of Radiohead library, you will find RHRouter and RHMesh under topic Managers with the following description:
RHRouter Multi-hop delivery of RHReliableDatagrams from source node to destination node via 0 or more intermediate nodes, with manual, pre-programmed routing.
RHMesh Multi-hop delivery of RHReliableDatagrams with automatic route discovery and rediscovery.
There are raw LoRa libraries for a mesh network. It's implemented on the Pycom devices, and the library for it is called PyMesh. The technology is based on Thread by Thread groop.
i would like build an only-audio conference system based on webrtc for a game, but i must avoid lag.
for example i can use https://github.com/muaz-khan/WebRTC-Experiment/tree/master/RTCMultiConnection
How many clients can be connected in the same time?and about bandwidth?
I think that the upload will be a limit increasing the amount of connected client.
Could i make a Tree of clients, so each client have only 2 or 3 connections?
The server can be in any language because i will use vert.x
regards
There are two scenarios:
1: Peer-to-Peer
In this model; maximum peer connections limit per page on chromium is 256.
2: Peer-to-Server
In this model; you can use media server to relay stream over unlimited peers.
In 1st model; you can face bandwidth/cpu usage issue.
In 2nd model; all such things are handled by the media server.
If you're planning to setup peer-to-peer video-conferencing; you'll use mesh model. It is suggested to limit conferencing to 5 users only. Otherwise, you'll face issues like audio-lost, echo and obviously bandwidth/cpu huge usage!
In broadcasting p2p scenarios; you can relay i.e. forward remote streams to overcome burden from single peer.
I'm trying to create an embedded outdoor display of bus arrival times at my university. I'd like the device to utilize my school's secured WiFi network to show arrival time updates determined from a server script I have running.
I was hoping to get some advice on the high-level operation of this thing -- would it be better for the display board to poll a hosted database via the WiFi network or should I have a script try to communicate with the board directly over 802.11? (Push or Pull?)
I was planning to use a Wifly or WIZnet ethernet board in combination with a wireless access hub. Mostly inspired by this project: http://www.circuitcellar.com/Wiznet/winners/001166.html Would anyone recommend something else over one of the WIZnet boards? I saw SPI/UART options and thought these boards could work with an AVR platform.
And out of curiosity -- if you were to 'cold start' this device (ie, request a bus arrival time by pushing the display's on button) you might expect it to take 10-20 seconds to get assigned an IP and successfully connect to the database, does that sound right?
I'd go pull. In fact, I'd have outdoor display make http or https requests of the server. That way the server could tell it how long to show a given set of data before polling for a new one using standard http page expiration.
I think pull would make it easier to have multiple displays, and to test your server as well. I've also got a gut feeling that this would make your display more secure. Someone would have to hack your server to hijack your display.
There's a very cool looking Arduino-targetted product called the WiShield. Seems super easy to use and he provides some source code. It uses SPI for host communication. If you're not interested in going the Arduino route, I'm sure the source code wouldn't be too hard to port to something like avr-gcc. Check it out, might save you some time and headaches for $55. Worth checking out anyway.