I am working on webRtc and i am facing so many problems like
streaming latency and video hang issues.
When 2 people connected to the same room(many-to-many) the videos stream fine and users receive and send the video in normal manner but when more users connect to the room the streaming become slow and even it hangs.
How to fix this issue? or webrtc is not capable to handle more users? because there are so many bandwidth and cpu usage issues and in the testing ,maximum 3 connections work fine after that it hangs.
Related
I am trying out the WebRTC examples of MuazKhan. It is working perfectly fine when the broadcaster is on AWS/Azure(or any other network) and the receiver is through my phones 4G network. But as soon as I switch to my broadband from 4G, the video stream is unable to connect and I the video player keeps on trying. Therefore I assumed that the problem is of the router NATting and will be resolved if I use a tried-n-tested TURN service such as Xirsys.
Sadly, even after using their TURN servers, I still am blocked with my broadband connection issue as mentioned above. Here are a few queries that I wanted to discuss:-
This issue seems to be due to NATting through my broadbands routers. Shouldn't using the TURN server solve it?
How can I verify if even the TURN servers are getting used and its not just the STUN servers.
Can this issue be due to the Signalling Server?
Do i need to enable some specific protocols in my router to make this work?
What else could be responsible for the issue that I should debug?
I'm building a webrtc project, and i need to:
route specific stream to specific users
record the streams on the server
I know this is typically the job of an SFU (selective forwarding unit)
However, before finding out about SFUs, I had previously started using browsers running on servers (i tested both chrome and firefox...using firefox for now), and it seems to be working.
I run my javascript and create peer connections and add the relevant streams just like i would on the clients.
I was even able to successfuly achieve multi-server hierarchy this way.
The only downside is see right now, is that the browsers decode the streams, which i believe would cause cpu overhead which i would not see using a proper SFU.
However, my project generally does 1 to many streaming (or rather few to many), and i need server side recording (which would cause an SFU to decode the streams anyway)
So, my question is..
Why is using a browser as an SFU for webrtc a bad idea? I haven't seen a lot of people doing this, so there must be a reason
Thank you
I've created a webrtc p2p app using socket io for signalling.
This work perfectly with one connection pair. When there are more than one connection pair, the video works well but the audio isn't hear on the other side.
I've deployed TURN server, it works as intended.
I'm not able to isolate the issue as in where does this issue lie.
I've been searching on google with no luck. It's been 3 days now
It would be great if anyone would point me to a right direction
We have developed a 1-1 video conferencing solution using RTMP , RTMFP technologies using Adobe. We are making use of Flash Media Development Server for recording the web streams. We have tested our application locally and it seems to work good.
Thus, We deployed the application to our server but the latency is very high (like 15 seconds). I mean when i talk , the other party gets my voice after 15 - 20 seconds.
Could someone explain me what is the real problem behind this ?
Do i have to purchase this https://secure1.influxis.com/webrootv10/custom_enterprise/index.aspx
Application.xml
http://www.2shared.com/document/PfMTpQsJ/Application.html
Thanks.
I've got some distance learning software, yesterday I've got near 1000 users watching video simultaneously. My client told me that server was down and memory was exceeded.
It asp.net website and server language is C#. I'm using Response.TransmitFile(...) for streaming videos. Is there any way to simulate 1000 simultaneously video stream situation to figure out what's going on, cause when I'm testing the website everything works fine.
Do not ever use your web server as video streaming server, there is no chance you can succeed.
I personally moved my videos to Amazon Web Services and I'm very satisfied with that.