[asterisk-users] what is rinstance parameter in sip header

[Nasir Javaid]

hello

i was wondering what is the use of "rinstance" in SIP Headers. I noticed
that this parameter is visible only when there is NAT invloved.

I am experiencing one way audio when dialing a registered user by his
IP:port. I this case "rinstance" parameter is missing.

when i dial "SIP/username" audio is fine but when i dial SIP/x.x.x.x:port
there is one way audion. Also please tell me what can go wrong by dialing by
ip:port.??

Best regards,

Nasir Javaid
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