Call audio drops after 5 minutes over WebRTC/TURN

Asterisk Asterisk Support
1

I have an issue that is fairly difficult to reproduce, but I’m hoping there’s someone who has any general thoughts on what could cause it. When I connect to Asterisk over WebRTC on a connection sufficiently complex to require TURN, things work great for 5 minutes and then no audio is received. The connection is still active, but the call goes silent. RTP logs show two-way transmission for the first 5 minutes, then one-way after that.

ICE is configured and working, and I’m connecting to Asterisk over HTTP (Sipml5 JavaScript client). The way I can reproduce it back home is to enable my soft firewall. However, the reason it’s difficult to reproduce is since I am away from my usual office (holiday travel), when I enable my Mac firewall here, I am not routed through TURN, and things work fine.

Any general thoughts? I would provide logs but am not able to reproduce the issue right now, unfortunately. I can provide my Asterisk configuration if that helps.

Thank you!

2

Here is my sip.conf:

[general]
realm=1.2.3.4
udpbindaddr=0.0.0.0:5060
transport=udp,ws,wss
useragent=Sipml5
icesupport = yes
session-timers=refuse
nat=auto_force_rport,auto_comedia

[WebRTCClient]
callerid="WebRTCClient" <100>
md5secret = mypass
type=friend
host=dynamic
context=webrtc
hasiax = no
hassip = yes
encryption = yes
avpf = yes
icesupport = yes
videosupport=no
directmedia=no
qualify = yes
canreinvite = no
3

This problem has been resolved. I was using Numb as my TURN server (numb.viagenia.ca), and I worked with them to track down the bug. It was an issue on the TURN server with permissions not being refreshed.