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.


Was there a solution for this audio problem? We have several clients needed this to work and we are testing from our office on their system and no audio in either direction. They have some remote hard phones now working with no audio issues…make no sense to me.