RTP keepalives

[Georges Buisson]

Hello,

Anyone has dealt with RTP & firewall issues on a hosted 3CX where call is put on hold (here it was a Yealink phone) for some time and users can no longer get audio upon resuming the call ?

When a call is put on hold, the SIP re-invites will set media to "inactive" (no RTP flowing both ways) and depending on the UDP timeout defined on the firewall, the state may expire and if this happens then there is no way the user can resume his call. Yes, this specific firewall seems pretty strict and is probably not re-using the same port on the global side when RTP resumes in the outbound direction from the Yealink.

Found some references... there is an RFC for RTP keepalives (RFC 6263) and the Asterisk SIP channel driver has an option for this called rtpkeepalive.

It's a long shot, but perhaps there a hidden option in 3CX for doing RTP keepalives ?

 

[cobaltit]

You should only need keep alives if your port forwarding isn't correct. Are you passing the firewall test?

 

[complex1]

What you can try is enable keep alive in 3CX.
(Settings > Network > Firewall > Enable Keep Alives)

 

[Georges Buisson]

Thanks for the answers.
@cobaltit : Yes firewall test passes, and all is working well other than that timeout for long call holds.

@complex1 : I believe this option will only affect SIP and not RTP... unless someone else can confirm they see periodic RTP keepalives packets sent out with this enabled.