Hi, I'm having some difficulties making freeswitch talk to gmail chat. Signalling and text chat work both ways, but have troubles with audio. I'm behind NAT and use STUN to traverse it. In a problematic case I observe the following error in debug mode in the console, where user1 initiated the conversation with user2 <iq to="user1@XXX/fsvega0EC80FEA" id="313" type="error" from="user2@YYY /VtokiPhn75120521"> <ses:session type="candidates" id="3759110558" initiator="user1@XXX/fsvega0EC80FEA" xmlns:ses="http://www.google.com/session"> <ses:candidate name="rtp" address="xxx.xxx.xxx.xxx" port="23751" username="bSYbFfMybRj8D3xH" password="ri5AOommjwum60f8" preference="1.0" protocol="udp" type="stun" network="0" generation="0"></ses:candidate> <ses:candidate name="rtcp" address="xxx.xxx.xxx.xxx" port="23752" username="tVI0bVcGkAl7kdQT" password="ay23gfVzUPdaKDW4" preference="1.0" protocol="udp" type="stun" network="0" generation="0"></ses:candidate> </ses:session> <error type="modify"> <sta:bad-request xmlns:sta="urn:ietf:params:xml:ns:xmpp-stanzas"></sta:bad-request> <sta:text xml:lang="en" xmlns:sta="urn:ietf:params:xml:ns:xmpp-stanzas"></sta:text> </error> </iq> It appears that user2 would like to modify the session, but mod_dingaling does not take any action and I do not see any RTP traffic with wireshark. Is this expected? A second question is: In the session setup log I see mod_dingaling advertising a port which is outside the range specified with the configuration variables rtp-start-port and rtp-end-port in switch.xml. Do the latter only work with SIP? Thanks for any advise, Fede