If you Look at the sessions that are getting created for SIP and RTP you will be able to see if the NAT is being properly applied or not. Now, the SIP INVITE packet will have the internal address in the SDP payload, and PA will have to NAT the SDP payload using the SIP NAT 1:1 NAT rule ( If no App override is in place) One way Audio, would usually mean, either the PA is not doing the correct NAT on the Payload which would mean the RTP session that are opened up will have the Internal Address, in which case the remote end will not be able to route the response correctly. Or If there is application Override in place, then the PA will not do a NAT at all on the INVITE payload, which would mean that the RTP session ( Audio) will get opened up with the Private address of the call initiator. The best way to verify if the PA is doing the NAT correctly, is to capture This traffic at receive and transmit stage on the PA, then look at the INVITE packet ( layer 7 payload ) to see if the Proposed IP in the Transmit capture is NATted by comparing it to the receive capture INVITE payload. Note: If the clients are NOT NAT AWARE ( if the clients are not configured to know their NATted IP), then application override will not solve the problem. You can upload the captures here along with SIP and RTP session details and I can take a look. Hope that helps. Aditya
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