During the testing procedure, I found that when both sip UAs who are located behind the same NAT cloud want to establish voice or video connection, there is not neccesary to bridge them with rtpproxy. Only in situations where one of UA sits behind NAT or each UA sits behind different NAT clouds, that need a RTPProxy to bridge their media stream. What I mean is SER can determine the use of RTPproxy or not through the registration of sip UA. In location table, there are recieved and contact fields. But ser don't fill the recieved field, I think it is very userful for NAT.
Glad to hear your instructions
Best Regards
Sun Zongjun