Hi,
I have a problem with NAT usage. First I thought I needed STUN but the rport
thing did it so much better and now it works fine doing a register. The connect
fields looks perfect but the problem now is that the keep-alive mechanism does
not work.
As I understand it Sofia-SIP send OPTIONS to itself to keep the NAT happy. The
problem is that both SIP (IMS actually) servers I have tested with reply using
a different source port as the sent to. I try to illustrate where is goes wrong:
Sofia SIP (from port X to port 5800) -> REGISTER -> NAT (from port Y to port
5800) -> Server // Fine so far
... Authentication and things
Server (from port 5800 to port Y) -> 200 OK -> NAT (from port 5800 to port X)
-> Server // Fine so far
Sofia SIP (from port X to port 5800) -> OPTIONS -> NAT (from port Y to port
5800) -> Server // Fine so far
Server (from port 5801 to port Y) -> 200 OK -> NAT (5801 is not 5800 so NAT
throws this away)
After a while Sofia-SIP prints:
outbound(015D0100): FAILED to validate <sip:ROUTERIP:Y;transport=udp>
outbound(015D0100): FAILED with 408 Request Timeout
It there any workaround for this? I cannot change the NAT or SIP Server
behaviors. I really like it to work independent of SIP server assuming that it
supports rport.
Regards,
Bengt Werstén
------------------------------------------------------------------------------
Create and Deploy Rich Internet Apps outside the browser with Adobe(R)AIR(TM)
software. With Adobe AIR, Ajax developers can use existing skills and code to
build responsive, highly engaging applications that combine the power of local
resources and data with the reach of the web. Download the Adobe AIR SDK and
Ajax docs to start building applications today-http://p.sf.net/sfu/adobe-com
_______________________________________________
Sofia-sip-devel mailing list
Sofia-sip-devel@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel