we are working with FreeSWITCH as webrtc server and Firefox as client with sip.js
we notice that when we receiving a call, the rtp takes about 60 sec to establish with firefox, where with chrome and edge it takes maximum 5 sec.
the STUN candidates returns correct result, with ip / port / relatedIP / relatedPort
when the webrtc server sends the call to the browser, this one takes 60 sec to answer to the server, which renegociates the port for audio.
for the same requests, Chrome answers within 10 sec maximum
Do you know if FF behaviour is normal ? could we configure the answer delay/timeout in FF ?
with stun, we get a list of candidate for the browser and send the call to the browser
a=candidate:1 1 UDP 1686052863 199.xxx.xxx.xx 57036 typ srflx raddr 10.1.0.2 rport 65228
setting remote audio ice addr to index 2 199.xxx.xxx.xx:57036 based on candidate
Activating RTP audio ICE: 6sfdsgfd:dyOdUp8kdkql01l1 199.xxx.xxx.xx:57036
after 60 seconds, we have this and then we have audio (whereas with chrome it takes 10 sec to get this CHANGING AUDIO)
Auto Changing audio stun/rtp/dtls port from 199.xxx.xxx.xx:57036 to 199.xxx.xxx.xx:65228
we have this behaviour only when going through a VPN (so probably using PAT or multiple NAT)
when using a simple home connection, we do not have any issue