Incoming call on WebRtc no longer then 60 seconds!

Hi guys! Help please - all our incoming calls have a limit on 60 seconds - i dont know why - can you help me?
all internal/external peers are online and pinging
everything seems good
but call fails after 60 seconds - why?

  1. Sometime carrier have a limited on the length of a call if no RTP packets is received. Make sure you have two way audio.
  2. Check NAT Rules to make sure the firewall is not closing the connection.

Just some ideas.

I have mostly nothing on rtp.conf:


I have 2 way audio - during this 60 seconds everybody hears each other
Nat has no problems - i have this effect on all 20 users in our crm system !
Can you help me?

You need to provide the logging from Asterisk. At the moment we don’t even know which side cleared the call/

Sixty seconds is of the right order of magnitude for a missing ACK.

Also, you haven’t indicated the version of Asterisk that you are using. WebRTC is a moving target and you need to always use the very latest version.

Now i switch to 17.5.1 and making a new tests

make everything ok with incoming calls - why? may be i should change some other settings?

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