Asterisk sip ring sound issue

I don’t think WebRTC uses TLS.

10000 - 25000 tcp is NOT necessary! So go ahead and take that out

thank you i got the solution as u said i change the port range 10000 to 25000 and ice support enabled and strict enabled by default ice support and strict rtp options enabled but i disabled before now again i enabled those and changed the port range it’s working fine

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