I am looking for monitor WebRTC based audio/video calls and conferences to integrate one of my already existing SIP monitoring Product. Can i have some suggestions to monitor the WebRTC sessions.
Thanks in advance and waiting for response.
Thanks & Regards
Sudhakar K.
https://www.callstats.io/
This tool is built specifically for monitoring the webRTC sessions.
Related
I've been reading about Janus, looked at the examples. I'm looking for a webRTC component that I can use in the following way:
Receive RTP video packets from some external sender
Become a WebRTC peer and connect to an external WebRTC signaling server, STUN, TURN, the usuals
Send the incoming RTP packets as a coherent video via the WebRTC peer connection to some other peer on a browser on the Internet
Is Janus the right tool? Maybe there are other tools? Would appreciate some directions..
Thanks!
I am not sure about Janus.
You can achieve these functionalities with LM Tools (lmtools.com) with easy configuration. It can receive RTP packets from external sender and can send those packets as per WebRTC specification to other peer.
Please note LM tools is not free software like JSON, though you can have free trial for 1 month.
Disclaimer: I work for LM Tools.
Can Janus WebRTC server implement server-side peer?
Yes, can can do that. What you are looking for is an RTP forwarding and you will get more context, and expert opinion from their community friendly google group page.
I hope you are looking for a Gateway solution.
(RTP/RTCP separate streams are converted to webrtc RTP/RTCP mux)
For this you need to make changes in the Janus code or use some plugin supporting RTP/RTSP.
Current Janus server relays RTP/RTCP and messages between browsers.
https://janus.conf.meetecho.com/docs/
I want to study relations of ICE/STUN/TURN protocol and network packets under operation of WebRTC. I found many testing products. Because I want to observe ICE/STUN/TURN mode of operation step by step, those testing products can not help me. I want to found a tool like pjsip icedemo tool (http://www.pjsip.org/pjnath/docs/html/ice_demo_sample.htm ).Have anyone used that tool?
I think that best test tools are Firefox and Chrome.
Turn on the logs for WebRTC and ICE and you will able to see all the details.
I've got WebRTC peer to peer working but when I want to broadcast a single camera to multiple clients obviously peer to peer isn't suitable.
I've found solutions like
http://lynckia.com
and
http://www.medooze.com/products/mcu/webrtc-support.aspx
But the first I can't get setup (and it seems to have cross browser issues)
the second just feels like we're hitting a nail with a nuclear missile.
All I need is a relay, I don't need to decode / recode streams.
I just need
The Broadcaster to connect to the server (peer to peer)
The clients to connect to the server (peer to peer)
The server to relay the stream from the broadcaster to the clients.
Is there any software out there that offers this solution that I've missed? is there an alternative working and scalable alternative?
Thanks
Jitsi Video Bridge works pretty much exactly how you describe.
On your server you can run Janus, to which your broadcaster can provide a stream via RTP.
Have a look at an example configuration file.
After writing a configuration file which defines how the server receives stream from the broadcaster, you should be able to launch janus in the background via a command line interface tool:
$ janus --daemon --config=config_file.conf
Also, see streaming test demo.
Note: I have not tested this thoroughly.
Have a look at this github-repo inspired from muaz khan's WebRTC p2p scalable broadcast. This can work great on LAN. On internet, I am not sure how well it can work as of now though we are improving it on the go.
If you just want to broadcast from a peer to a set of peers, if they don't care about the latency, the best solution is to covert WebRTC to live streaming, without transcoding just muxing:
Peer(Publisher) ---WebRTC--> Server --RTMP/HLS/DASH--> Peers/Players
If this works good for you, SRS is able to covert WebRTC to live streaming.
Because live streaming allows you to use CDN or TCP to deliver the streams, and the latency is about 3~5s, so this solution is only available when Peers/Players never need to communicate to the Peer(Publisher).
If you want all those peers to talk to each other, it's very complex and need a WebRTC SFU cluster to do this, there will be a huge number of streams. For example, if allows 100 peers to talk to each other, there will be 100x100 = 10k streams.
It's too complicated, so I don't think there is good open-source solution right now(at 2022.02).
Firstly I just want to know that what is the difference between freeSwitch and Red5?
As I have very good working experience with red5 and I have made many app's that streamed video/audio using Red5.
But now I am not able to understand that If Red5 can do video/audio conferencing or streamed live video then what is the use of FreeSwitch in conferencing or in other things.
I want to make a app in PHP or rails or Django(Python) where users can record their voice by participating in conference and while recording, the voice of all users will be broadcast to other members.
So now I really want to know what will be the right solution to do this?
FreeSWITCH is primarily a telephony application server. So, it is oriented on solving the telephony tasks. There is also support for WebRTC, and some work is being done for video conferencing.
What you can easily do with FreeSWITCH, is allowing users join your Red5 conference from the telephony network.
I wanted to perform http streaming from red5 server so as to use it to broadcast for Iphone. Can u guys suggest a way to do it. Is there a way to play rtmp stream from red5 in Iphone. Any help is appreciable
I was involved in the discussion you linked. But I don't think that the project team has yet committed the HTTP Streaming components. There is also no documentation about it.
So if there is a possibility to do that you probably will need somebody from red5 developers to hire.
Sebastian