SIP OPTIONS after call

General
1

Hi
I use janus as sip - webrtc gateway.
After first INVITE sended I saw some SIP OPTIONS messages that sends by Janus on URI in username with which webrtc user registers to make call.
This messeges is looks like

OPTIONS sip:username@example SIP/2.0
v:SIP/2.0/UDP ip:port;rport;branch=z9hG4bKm7X7708cjH7Sg
Max-Forwards:0
f:sip:username@example.com;tag=K6Hma44rrBSSp
t:sip:username@example.com
i:iLf5gs2IkIgnkyNrno1C7f
CSeq:75468240 OPTIONS
s:KEEPALIVE
Accept:application/vnd.nokia-register-usage
l:0

For what these messages sends?
Thank you for your help!

2

Would like to know the purpose of the SIP OPTIONS as well and how to disable it.

Thanks

3

That’s something the Sofia SIP stack does, probably to detect support for features to subscribe to. I don’t know if it can be disabled.

4

The real issue is that Janus generates the OPTIONS with ‘max-forward’ set to 0, hence, my sip server rightfully response with response code 483 ‘Too many hops’ (instead of 200OK) and Janus doesn’t like this and never refreshes the registration. Is there any configuration file where I can replace 0 with the standard 70?

Thanks