I have compiled sip.js with webrtc for mobile. I am able to register the sip user for both android and ios platform over web socket and I am able to make outgoing calls without any issue but I am not able to receive incoming calls over on my webrtc end point.
when ever I am sending incoming calls to webrtc end point, I am getting the below error in my asterisk console.
chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
Do I need to add STUN server configuration for webrtc end points to make incoming work as it is not able to relay packet to end point which is behind the router or firewall ?