Hi to all,
I’m working on asterisk webrtc and now I have some NAT issue.
I’m using Asterisk 11.6, Sipml5 and eyebeam. Webrtc calls from sipml5 to softphone are Ok and I see in the RTP logs this : (VIA ICE) . so ICE support is enabled.
When I make a call from sipml5 to softphone in the internet, I don’t have any audio. I noticed that Asterisk does not change the private address of client in SDP.
I’m using nat=force_rport,comedia and externaddr=my public ip. I didn’t use any STUN server config for server or client.
So are there any problem in Asterisk 11.6 NAT Traversal mechanism ?