Hi Nitin, There could be multiple reasons on why the terminating UA (or UAS) is behaving without sending the response. The SDP looks okay. I would say may be its not liking multipart content type or its not able to decode application/isup.
But typically, all TU have to be designed in a way that it will receive the messages liberally and deliver strictly (aka RFC 3261). I would recommend to take trace/logs in the terminating UA to see what's going wrong. In either way, it should have sent negative response (400 etc). Regards, Somesh -----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of ext PUNJABI BOY Sent: Tuesday, May 01, 2012 8:03 PM To: [email protected] Subject: Re: [Sip-implementors] Termination is Not replying to INVITE at ALL Dear All, Could anyone please help me out on the below problem please. Thanks, Nitin On Tue, May 1, 2012 at 8:37 AM, PUNJABI BOY <[email protected]> wrote: > Dear All, > > I need the help from in one of my sip call scenario. Call flow is as below. > > UA -> 187.28.162.5 -> SBC Realm 1 (200.182.99.75) -> Machine SIP Realm 1 > (187.60.58.133) -> SBC Realm 2 (187.60.52.80) -> Termination Provider Realm > 1 (200.216.239.196). > > > However the problem i am seeing rightnow is that whenever my SBC is > sending the call from from *second last entity to Termination Provider > REALM1 then the termination provider realm is not replying to the INITIAL > invite at all.* > > > Although i checked the content type value in message body and seeing > something odd, not sure if this is the valid headers or not as below. > > > UA--->to SBC(And in this SBC recieved the INVITE from UAC and forwarded to > next hope(Which means SBC Ingress realm and it sends the 100 Trying) > > > ===== > > Content-Length: 455 > > Content-Type: multipart/mixed;boundary=ssboundary > > > --ssboundary > > Content-Length: 249 > > Content-Type: application/sdp > > ===== > > > Also SDP > > ====== > > m=audio 17774 RTP/AVP 18 8 4 2 > > a=rtpmap:18 G729/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:4 G723/8000 > > a=rtpmap:2 G726-32/8000 > > a=fmtp:18 annexb=yes > > --ssboundary > > Content-Length: 30 > > Content-Type: application/isup;version=itu-t92+ > > > ..h. > > .. > > ......%..... > > ....#v.C. > > --ssboundary-- > > ===== > > > *Now from here the situation got worst where Termination Realm is not > replying to MSX at all, where also i can see another VIA header is added by > MSX(could by due to proxy)* > > > ====== > > INVITE sip:[email protected];user=phone SIP/2.0 > > Max-Forwards: 66 > > Session-Expires: 3600;refresher=uac > > Min-SE: 600 > > Supported: timer, 100rel > > To: <sip:[email protected]:5060;user=phone> > > From: <sip:[email protected]>;tag=3544361906-29701 > > Call-ID: [email protected] > > CSeq: 1 INVITE > > Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, > SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH > > Record-Route: <sip:187.60.58.133> > > Via: SIP/2.0/UDP 187.60.58.133:5060;branch=z9hG4bK60c2d2733c5e07c55fee0d1 > > Via: SIP/2.0/UDP 200.182.99.75:5060 > ;branch=z9hG4bK9d9c30880484a3108ccd4d447b928f24 > > Contact: <sip:[email protected]:5060> > > Call-Info: > <sip:200.182.99.75>;method="NOTIFY;Event=telephone-event;Duration=1000" > > Content-Type: multipart/mixed;boundary=1140858378-1335373106-29811 > > Content-Length: 459 > > > --1140858378-1335373106-29811 > > Content-Type: application/SDP > > > v=0 > > o=RISNEXT01 24425908 24425908 IN IP4 200.182.99.75 > > s=sip call > > c=IN IP4 200.182.99.92 > > t=0 0 > > m=audio 47666 RTP/AVP 18 8 4 2 > > a=rtpmap:18 G729/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:4 G723/8000 > > a=rtpmap:2 G726-32/8000 > > a=fmtp:18 annexb=yes > > > --1140858378-1335373106-29811 > > Content-Type: application/ISUP;version=itu-t92+ > > ====== > > Could anyone please check and help me to investigate the issue. For the > reference i also attached both legs of ethereal traces. > > Thanks, > Nitin > _______________________________________________ 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
