No audio using Asterisk 13.4 + SIPml5

I’m trying to make a call between two sipml5 endpoint using asterisk 13.4
Called endpoint rings, but no audio path has been established.
I got the following error:

== WebSocket connection from ‘’ for protocol ‘sip’ accepted using version ‘13’
– Registered SIP ‘6000’ at
== Using SIP RTP CoS mark 5
[Jul 23 18:45:01] ERROR[990][C-0000000a]: pjsip:0 <?>: icess0x7ffdf01 …Error sending STUN request: Invalid argument

Can you please help?