Hello All,
I am very new on asterisk and WebRTC. I am trying to make a sample audio video chat demo but i am getting below listed errors.
– SIP/AL00101-00000005 is ringing
[Apr 4 07:45:20] ERROR[62003][C-00000003]: pjproject:0 <?>: icess0x7fcd48016f28 …Error sending STUN request: Operation not permitted
– SIP/AL00101-00000005 answered SIP/AL00102-00000004
[Apr 4 07:45:20] ERROR[62206][C-00000003]: pjproject:0 <?>: icess0x7fcd60014728 …Error sending STUN request: Operation not permitted
– Channel SIP/AL00101-00000005 joined ‘simple_bridge’ basic-bridge <817c12f8-6806-49b1-9b36-74b82e9ffaf6>
– Channel SIP/AL00102-00000004 joined ‘simple_bridge’ basic-bridge <817c12f8-6806-49b1-9b36-74b82e9ffaf6>
[Apr 4 07:45:20] ERROR[61562]: pjproject:0 <?>: icess0x7fcd48016f28 …Error sending STUN request: Operation not permitted
[Apr 4 07:45:20] ERROR[61562]: pjproject:0 <?>: icess0x7fcd60014728 …Error sending STUN request: Operation not permitted
> 0x7fcd48013f50 – Strict RTP learning after ICE completion
[Apr 4 07:45:21] ERROR[61562]: pjproject:0 <?>: icess0x7fcd60014728 …Error sending STUN request: Operation not permitted
[Apr 4 07:45:21] ERROR[61562]: pjproject:0 <?>: icess0x7fcd60014728 …Error sending STUN request: Operation not permitted
> 0x7fcd60011b70 – Strict RTP learning after ICE completion
Thanks in Advance.