Is it possible to communicate with RTMFP from serverside without flash server?
Thanks ;)
Presumably you just need to communicate with the Flash player so that's not related to RTMFP and is easily accomplished.. Did I misunderstand?
RTMFP is a p2p communication method, but you need to have a server in order to exchange the ID of the p2p participants.
Related
I have a surveillance Camera and i want to stream the live feed to multiple clients in the same network using WebRTC. To save compute resource and bandwidth requirement i want to do multicasting on WebRTC feed. Is this possible?
WebRTC doesn't support multicast. You might want to look into tools like webtorrent.
I have implemented basic WEBRTC peer connection between two clients and uses RTCPeerConnection and getUserMedia APIs for it. I am fetching Audio tracks only from the stream.
Even though my signalling works, ice agents are shared, streams are shared as well.
The audio comes out distorted completely.
However, if I utilize a VPN on one device the other device can hear the audio coming from the device with VPN very clearly with no hiccups.
NOTE:
I tried this with only stun and with a combination of stun and turn servers(viagenie).
Can someone tell me how do i debug the issue, what to look for?
Is my ISP involved in messing around with the media stream? (I live in the UAE.)
Thanks in advance!
i think the problem is UAE ISP block the STUN/TURN PORT (3478,5766 ...)
you can try to use 443/80 port?
In a cellular environment, the ip address assigned for a phone changes dynamically.
So i wonder how WebRTC deals with this issue.
Does WebRTC has some feature called 'Path Migration' like QUIC ?
Is ICE Restart what i am finding?
There is an RFC8016 TURN about mobility to reuse TURN session in case of IP change, but it is AFAIK not implemented in the actual WebRTC stack.
AFAIK WebRTC does not handle yet this case.
You could detect the error on client side and restart ICE agent.
Can I use the UDP protocol to implement the communications between two directly connected PCs?
Yes, you can.
The only downfall could be you can't be sure of proper delivery of messages(as in case of TCP).
You'll need to develop a very good application layer code, which would kind of mimic the functionalities of TCP , to achieve a reliable communication.
As far as I know, there are many applications using SIP : ekiga, linphone, even skype, … Are those device all able to work with each other? I mean, if I register with, say, linphone, will someone on skype be able to ring me?
Skype is not using SIP. But if you interconnect any SIP system then there are plugin available like SipTheeSkype, SIP to Skype Gateway, Skype with asterisk & some other.
By using that plugin you can interact with Skype network from your SIP network.
As far as concern to other SIP client they are all interact with each other if they don't have any proprietary header check to register with specific server only.