If you are saying the port in the m= line that Asterisk sends from is not the source port seen by the peer, when Asterisk subsequently sends media for that stream, this is almost certainly the result of a NAT router thinking that the source port number is already in use, for another call, and dynamically allocating a new port number. If so, this has to be fixed in the router.
That would be valid in SIP, but is not something that Asterisk can do, although it can handle the case when the other side is doing it, as long as you enable symmetric media.
I’d say this has to be a rogue router, unless you have a hardware fault.
Note that you haven’t captured any outbound RTP, and you haven’t captured the port numbers for the inbound RTP. Also, you haven’t told us whether this is the current or the deprecated driver, and which version of Asterisk.
Where was the capture taken, and is this the same call as the one from which the INVITE was taken? What port numbers are reported by “rtp set debug on” and “pjsip set logger on”, which are the preferred sources for this information (and assuming you are using currently supported drivers).
Note that the Issabel package names and versions don’t mean a lot, although I assume the 16.7.0 on the asterisk package is the correct asterisk version (somewhat out of date).
Problem solved. It was related with the range of port allowed in firewall. When INVITE choose one not allowed then assign other and the communication do not work.