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

Reply via email to