Hi,
I’m using WebRTC with asterisk and I having a problem when I’m behind a NAT.
Locally all works fine.
Witch logs or configurations could I post where? I’m a completely newbie when using asterisk 
I’m calling in a local network using a sipml5 client and a microsip to an asterisk in the internet!
in the asterisk console I see the following error:
“
<------------->
— (13 headers 45 lines) —
Using INVITE request as basis request - 12cf758d-ee23-f5de-35f5-f521cd029422
Found peer ‘10002’ for ‘10002’ from 79.169.75.39:53745
== Using SIP RTP CoS mark 5
Found RTP audio format 111
Found RTP audio format 103
Found RTP audio format 104
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 106
Found RTP audio format 105
Found RTP audio format 13
Found RTP audio format 126
[Feb 26 12:15:38] WARNING[475][C-00000009]: sip/sdp_crypto.c:173 sdp_crypto_activate: Could not set SRTP policies
[Feb 26 12:15:38] WARNING[475][C-00000009]: sip/sdp_crypto.c:173 sdp_crypto_activate: Could not set SRTP policies
Found unknown media description format opus for ID 111
Found unknown media description format ISAC for ID 103
Found unknown media description format ISAC for ID 104
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found unknown media description format CN for ID 106
Found unknown media description format CN for ID 105
Found audio description format CN for ID 13
Found audio description format telephone-event for ID 126
[Feb 26 12:15:38] WARNING[475][C-00000009]: chan_sip.c:10454 process_sdp: Rejecting secure audio stream without encryption details: audio 56936 RTP/SAVPF 111 103 104 0 8 106 105 13 126
”
in module show I see that res_srtp.so is loaded
res_srtp.so Secure RTP (SRTP) 0
What can I do?
Thanks for the help.