WEBRTC audio problem


Hi, I have a UCM6510 with WAN interface directly connected with public IP address. I have external SIP clients working perfecly, but I have problem with WEBRTC. I can login successfully but I have no audio on both sides of the call.

I usually activate the NAT Settings when the PBX behind a router, but in the case the PBX is the router, so I not enabling any NAT features. In other hand, the external SIP clients are working fine, so I imagine it is OK.

My only problem is the WEB RTC, so, can someone give me a tip about what to do about it ? Is it needed any information for this forum ?

Thanks, Gilson


I mostly sure it is buged. If there is NAT then UCM do not use proper IP in SDP so voice cannot start :confused:


Sure, but the public IP Address is directly on WAN interface, there is no NAT. If it is a bug, a lot o people should be complaing about this problem. Do I have to activate the NAT on UCM6510 even if it is directly on WAN ? Is there a channel that I can report a problem to GS ? Thanks for your @marcin.hyjek


helpdesk https://helpdesk.grandstream.com
Well for not nat there is no problem, i only see this for nat.
Only few people use WebRTC, 99,99% is SIP.