====== Common SIP problems ======
===== Phone cannot register =====
Turn off ICE, STUN, uPnP and other “tricks” that your SIP phone might try to use to get through ordinary firewalls. As the Internet Gate is SIP transparent such “tricks” are harmful and unnecessary - and might even actually stop SIP traffic from getting through the firewall!

===== Phone doesn't stop ringing after caller hangup =====
Some SIP clients fail to send correct CANCEL requests to terminate phone ringing.\\
Enable **Strip route information from 180 responses** on [[web GUI:Advanced SIP settings]] page for a workaround.

===== Phone goes silent in middle of call =====
It could be the phone doesn't support mid-session port changes. Enable **Reuse port numbers within session** on [[web GUI:Advanced SIP settings]] page for a workaround.

===== Call transfer problems =====
In SIP, call transfers should be performed by the endpoints (SIP UA), but handling this is not a mandatory requirement according to the standard so some SIP UA:s may not support it. If you have problems with call transfers Internet Gate can perform the call transfer locally on behalf of an endpoint, using a [[B2BUA]]. You configure **Handling of Call Transfer** on the [[web GUI:Advanced SIP settings]] page.

===== Media doesn't go through VPN tunnel =====
If for example media from an ITSP goes directly to a remote caller, not through the IPSec tunnel she has to Internet Gate, you need to enable **Relay media for calls over VPN** on [[web GUI:Advanced SIP settings]] page to force media from ITSP to Internet Gate so it can forward it through the IPSec tunnel to the caller.