SIP from UDP to TCP- no audio

Hello dear community,
Please advise - when using standard chan_sip settings with UDP transport everything works perfectly, but when I change extension setting to use TCP (Transport -> All TCP Primary) and change transport on client side - extension registers normally over TCP, but making call from the extension to for example announcement results in no audio and “Retransmission timeout reached…” errors on asterisk. What could be the problem?
Using latest asterisk distributive with latest freepbx installed on centos 6.
Thanks in advance.