Hi folks, I recently started to work with SIPp and was wondering if I could use it to test terminal mobility scenarios, that is due to an IP change in the UA terminal sendig a re-INVITE with the updated IP contact information.
My first attempt was to establish a typical call with media support, then call a external function that basically changes the IP address of the eth interface and then trigger the re-INVITE procedure. As I expected, the UPD socket broke due to the address change and SIPp could no send SIP messages after IP address change. RTP packets were sent through the new interface though with checksum errors. The issue is related how sockets are created, to my understanding while RTP packets are sent in RAW mode the ISP packets use the same UDP socket for the session (in default transport mode). At this point my questions for you. - Has anyone gained some valuable experiences with related activities. - Can anyone share a working INVITE / re-INVITE scenario. Any hint or bearing on how to best proceed on this issue would be really appreciated. Best regards, Christian ------------------------------------------------------------------------- This SF.net email is sponsored by DB2 Express Download DB2 Express C - the FREE version of DB2 express and take control of your XML. No limits. Just data. Click to get it now. http://sourceforge.net/powerbar/db2/ _______________________________________________ Sipp-users mailing list [email protected] https://lists.sourceforge.net/lists/listinfo/sipp-users
