The incoming INVITE offers only RTP, and you have configured mandatory SRTP. So the call will not be successful. Have you tried to configure optional SRTP, or only RTP? Regards, Dan On Sat, 2012-10-06 at 17:02 +0100, Josh wrote: > I have the following problem which I hope the more knowledgeable on here > could shed some light on: > > Between my UA and the SIP server I have set up SRTP/TLS connection on > and I can register normally with that enforced. When somebody (over the > PSTN network using a regular phone number) tries to ring me, I get the > above error. > > I did a bit of research and found that possible causes could either be > wrong codec or the enforcement of SRTP (I have it as "Mandatory" in my > UA), but I am still not 100% sure and thought to ask here. The error I > am getting from my UA is shown above, though the underlying problem > seems to be with the "SIP/2.0 488 Not Acceptable Here" message in packet > 4 below. This is the full trace of what is happening (I masked my IDs > and the relevant IP subnets): > > ========== > ---------- > 15:12:06 Packet 1/6, 976 bytes > ---> > UDP port 5060 INVITE sip:me at provider.voip > audio->212.x.x.x:12706 > UDP port 5060 > ---------- > INVITE sip:me at provider.voip SIP/2.0 > Via: SIP/2.0/UDP 85.xx.xx.xx;branch=z9hG4bK12fd.d78b9b71.0 > Via: SIP/2.0/UDP > 212.x.x.x:5060;received=212.x.x.x;branch=z9hG4bK0679c35a;rport=5060 > From: "+49xxxx" <sip:+49xxxx@212.x.x.x>;tag=as1012c966 > To: <sip:me at provider.voip> > Contact: <sip:%2b49xxxx at 212.x.x.x> > Call-ID: 39dacf63231a416e13b2d2804a433687 at 212.x.x.x > CSeq: 102 INVITE > User-Agent: Provider-VOIP > Max-Forwards: 68 > Date: Sat, 06 Oct 2012 15:12:06 CET > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > Supported: replaces > Content-Type: application/sdp > Content-Length: 260 > Thor-Scope: lookup-destination > Thor-Origin: sip:212.x.x.x:5060 > > v=0 > o=root 1888 1888 IN IP4 212.x.x.x > s=session > c=IN IP4 212.x.x.x > t=0 0 > m=audio 12706 RTP/AVP 0 8 3 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=sendrecv > ========== > ---------- > 15:12:06 Packet 2/6, 1150 bytes > <--- > UDP port 5060 INVITE sip:me at provider.voip:52222 > audio->81.x.x.x:57958 > UDP port 5060 > ---------- > INVITE sip:me at provider.voip:52222;transport=tls;ob SIP/2.0 > Record-Route: <sip:81.x.x.x;lr;ftag=as1012c966;did=05.76f75113> > Via: SIP/2.0/UDP 81.x.x.x;branch=z9hG4bK12fd.65070351.0 > Via: SIP/2.0/UDP 85.xx.xx.xx;branch=z9hG4bK12fd.d78b9b71.0 > Via: SIP/2.0/UDP > 212.x.x.x:5060;received=212.x.x.x;branch=z9hG4bK0679c35a;rport=5060 > From: "+49xxxx" <sip:+49xxxx@212.x.x.x>;tag=as1012c966 > To: <sip:me at provider.voip> > Contact: <sip:%2b49xxxx at 212.x.x.x> > Call-ID: 39dacf63231a416e13b2d2804a433687 at 212.x.x.x > CSeq: 102 INVITE > User-Agent: Provider-VOIP > Max-Forwards: 68 > Date: Sat, 06 Oct 2012 15:12:06 CET > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > Supported: replaces > Content-Type: application/sdp > Content-Length: 260 > Thor-Scope: relay-destination > Thor-Target: sip:178.x.x.x:52222;transport=tls > > v=0 > o=root 1888 1888 IN IP4 212.x.x.x > s=session > c=IN IP4 81.x.x.x > t=0 0 > m=audio 57958 RTP/AVP 0 8 3 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=sendrecv > ========== > ---------- > 15:12:06 Packet 3/6, 481 bytes > ---> > UDP port 5060 100 Giving a try for INVITE > UDP port 5060 > ---------- > SIP/2.0 100 Giving a try > Via: SIP/2.0/UDP 81.x.x.x;branch=z9hG4bK12fd.65070351.0 > Via: SIP/2.0/UDP 85.xx.xx.xx;branch=z9hG4bK12fd.d78b9b71.0 > Via: SIP/2.0/UDP > 212.x.x.x:5060;received=212.x.x.x;branch=z9hG4bK0679c35a;rport=5060 > From: "+49xxxx" <sip:+49xxxx@212.x.x.x>;tag=as1012c966 > To: <sip:me at provider.voip> > Call-ID: 39dacf63231a416e13b2d2804a433687 at 212.x.x.x > CSeq: 102 INVITE > Server: SIP Thor on OpenSIPS XS 1.8.0 > Content-Length: 0 > ========== > ---------- > 15:12:06 Packet 4/6, 722 bytes > ---> > UDP port 5060 488 Not Acceptable Here for INVITE > UDP port 5060 > ---------- > SIP/2.0 488 Not Acceptable Here > Via: SIP/2.0/UDP 81.x.x.x;branch=z9hG4bK12fd.65070351.0 > Via: SIP/2.0/UDP 85.xx.xx.xx;branch=z9hG4bK12fd.d78b9b71.0 > Via: SIP/2.0/UDP > 212.x.x.x:5060;rport=5060;received=212.x.x.x;branch=z9hG4bK0679c35a > Record-Route: > <sip:85.xx.xx.xx:5061;transport=tls;lr;r2=on;ftag=as1012c966;did=05.c25aa83> > Record-Route: <sip:85.xx.xx.xx;lr;r2=on;ftag=as1012c966;did=05.c25aa83> > Record-Route: <sip:81.x.x.x;lr;ftag=as1012c966;did=05.76f75113> > Call-ID: 39dacf63231a416e13b2d2804a433687 at 212.x.x.x > From: "+49xxxx" <sip:+49xxxx@212.x.x.x>;tag=as1012c966 > To: <sip:me at provider.voip>;tag=-FBQxyWnTqZHHrOenpJvRCDUnZwCgDYh > CSeq: 102 INVITE > Content-Length: 0 > ========== > ---------- > 15:12:06 Packet 5/6, 660 bytes > <--- > UDP port 5060 488 Not Acceptable Here for INVITE > UDP port 5060 > ---------- > SIP/2.0 488 Not Acceptable Here > Via: SIP/2.0/UDP 85.xx.xx.xx;branch=z9hG4bK12fd.d78b9b71.0 > Via: SIP/2.0/UDP > 212.x.x.x:5060;rport=5060;received=212.x.x.x;branch=z9hG4bK0679c35a > Record-Route: > <sip:85.xx.xx.xx:5061;transport=tls;lr;r2=on;ftag=as1012c966;did=05.c25aa83> > Record-Route: <sip:85.xx.xx.xx;lr;r2=on;ftag=as1012c966;did=05.c25aa83> > Record-Route: <sip:81.x.x.x;lr;ftag=as1012c966;did=05.76f75113> > Call-ID: 39dacf63231a416e13b2d2804a433687 at 212.x.x.x > From: "+49xxxx" <sip:+49xxxx@212.x.x.x>;tag=as1012c966 > To: <sip:me at provider.voip>;tag=-FBQxyWnTqZHHrOenpJvRCDUnZwCgDYh > CSeq: 102 INVITE > Content-Length: 0 > ========== > ---------- > 15:12:06 Packet 6/6, 393 bytes > ---> > UDP port 5060 ACK sip:me at provider.voip > UDP port 5060 > ---------- > ACK sip:me at provider.voip SIP/2.0 > Via: SIP/2.0/UDP 85.xx.xx.xx;branch=z9hG4bK12fd.d78b9b71.0 > From: "+49xxxx" <sip:+49xxxx@212.x.x.x>;tag=as1012c966 > Call-ID: 39dacf63231a416e13b2d2804a433687 at 212.x.x.x > To: <sip:me at provider.voip>;tag=-FBQxyWnTqZHHrOenpJvRCDUnZwCgDYh > CSeq: 102 ACK > Max-Forwards: 69 > User-Agent: SIP Thor on OpenSIPS XS 1.8.0 > Content-Length: 0 > ========== > > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >
