I'm not exactly sure what you seeking. RFC 3261 discusses "Anonymous". Privacy is also discussed within RFC 3323 and RFC 3325.
Remote-Party-ID is defined within an expired draft which never became an RFC. Any device can send a valid SIP message to another device; however the receiver based upon RFCs and services can decide how to handle the request. > -----Original Message----- > From: [email protected] > [mailto:[email protected]] On > Behalf Of Rashid Shakil > Sent: Tuesday, December 16, 2008 12:21 PM > To: [email protected] > Subject: [Sip-implementors] Remote Party ID (RPID) missing > for Restrictedcall (caller ID hide) > > Hello, > > Quick question please ...Can any SIP peer allowed to send an > INVITE for restricted call without calling FROM information > with no "Remote Party ID". If you look at the following > information calling From number is restricted and "Remote > Party ID (RPID)" is missing as well therefore no way to find > out where the call is originating from. Is this acceptable if > yes can you please reference a draft where I can read this in detail ? > > ============================================================== > ============== > INVITE sip:[email protected]:5060 SIP/2.0 > Via: SIP/2.0/UDP > 6.6.19.72:5060;branch=z9hG4bK61ed25f40d9a00534ea564d93dcad9e7-0 > From: "Anonymous" > <sip:[email protected]:5060>;tag=824f5ba36d11a58f9978aa130ae1eefd > To: <sip:[email protected]:5060> > Call-ID: [email protected] > CSeq: 34992 INVITE > Max-Forwards: 70 > Allow: > INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRA > CK,OPTIONS > Accept: application/sdp, application/isup, application/dtmf, > application/dtmf-relay, multipart/mixed > Contact: "Anonymous" <sip:[email protected]:5060;transport=udp> > Anonymity: uri > Supported: timer,100rel > Session-Expires: 1800 > Min-SE: 1800 > Content-Length: 287 > Content-Disposition: session; handling=required > Content-Type: application/sdp > > v=0 > o=Sonus_UAC 145400 14540000 IN IP4 62.62.96.28 s=SIP Media > Capabilities c=IN IP4 62.62.96.28 t=0 0 m=audio 43996 RTP/AVP > 0 18 100 a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:100 telephone-event/8000 > a=fmtp:100 0-15 > a=sendrecv > a=maxptime:20 > ================================================================== > > Regards, > > Rashid Shakil > > > > _______________________________________________ > Sip-implementors mailing list > [email protected] > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > _______________________________________________ Sip-implementors mailing list [email protected] https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
