How do i stop echo in a webrtc call. Video is working and sound but there is echo hence we cannot talk to each other.
Am using jssip+asterisk16
Echo cancellation is the responsibility of the point in the system, remote from where the echo is heard, that transitions from high latency (VoIP, or mobile phone), to low latency (analogue, or circuit switched). In the typical WebRTC setup, that is not within Asterisk, but you haven’t sufficiently described your network to work out where it is.
Ona a pure SIP system, it would be the remote SIP device.
Thanks @david551 for your answer. just to decribe my solution:
- asterisk server on the cloud(meaning on public IP)
- on the same server there is apache server to serve the webrtc pages onto which the users who open the webpages are behind NAT on ther local networks which have internet
- My main sip device is the webbrowser which connects to the server using webrtc
- How can i cancel echo on the webbrowser?
- am using jssip javascript library for my webrtc client
Your described configuration cannot suffer from echo. To suffer from echo you need something that can echo the audio back. That can generally only happen if there are two telephone like devices, one at each end.
I’d therefore suggest you have missed some important detail.
(Actually it is possible for an echo canceller to create echo, if it believes there is an echo, when there isn’t, but I don;t see how that would happen, either, and Asterisk will not be doing any echo cancellation in such a configuration.