Hi, I have met a problem that run the PJSIP symbian_ua_gui example. I bulid a sisx file with Carbide C/C++ and install in NOKIA N73. I make a call with this application, it can register successfully. But, after the confirmed by callee, it's strange that the caller receive the BYE request and the application have to exit this call. The log message as below: RX 373 bytes Response msg 200/REGISTER/cseq=25494 (rdata0x71da04) from UDP 210.52.252.108:5060: SIP/2.0 200 OK CSeq: 25494 REGISTER Via: SIP/2.0/UDP 10.14.230.182:5060;rport;branch=z9hG4bKPjnghsLkjHbXoDxcByoZGf0jFBPBP4itTN From: <sip:wsz@210.52.252.108>;tag=gk3RNJJp7Ne.Hqasm4FBJ5tg0NjnTCC9 Call-ID: c66ZHx5fG5HFuROZqsKPq3P5Yy8J.J4e To: <sip:wsz at 210.52.252.108>;tag=281242092106 Contact: <sip:wsz at 10.14.230.182:5060>;expires=30 Expires: 30 Content-Length: 0 --end msg--sip:wsz at 210.52.252.108: registration success, status=200 (OK), will re-register in 30 secondsKeep-alive timer started for acc 1, destination:210.52.252.108:5060, interval:15sRegistration success!RX 633 bytes Response msg 200/INVITE/cseq=14946 (rdata0x71da04) from UDP 210.52.252.108:5060: SIP/2.0 200 OK CSeq: 14946 INVITE Via: SIP/2.0/UDP 10.14.230.182:5060;rport;branch=z9hG4bKPjUL89AEql5o50wLomdkDK5Tpp9TnyaIKw From: sip:wsz@210.52.252.108;tag=oMiS2VsfsgIMlTWFdT0v7HcC9RZ7O9X- Call-ID: MOTwWfT8guaDRb35fe-SNYEntSNolOtM To: sip:013880767030 at 210.52.252.108;tag=2812410921537011883127973 Contact: <sip:210.52.252.108:5060;transport=udp> Content-Type: application/sdp Content-Length: 224 v=0 o=VoipSwitch 8972 8972 IN IP4 210.52.252.108 s=VoipSIP i=Audio Session c=IN IP4 210.52.252.108 t=0 0 m=audio 7972 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv --end msg--Call 0 state=CONNECTING VAD temporarily disabledEncoder stream startedDecoder stream startedMedia updates, stream #0: GSM (sendrecv)Port 1 (sip:013880767030 at 210.52.252.108) transmitting to port 0 (Symbian Audio)Port 0 (Symbian Audio) transmitting to port 1 (sip:013880767030 at 210.52.252.108)TX 377 bytes Request msg ACK/cseq=14946 (tdta0x74d540) to UDP 210.52.252.108:5060: ACK sip:210.52.252.108:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.14.230.182:5060;rport;branch=z9hG4bKPju9qRLRFifXPfgBJYle6hofuMLJFP09BC Max-Forwards: 70 From: sip:wsz@210.52.252.108;tag=oMiS2VsfsgIMlTWFdT0v7HcC9RZ7O9X- To: sip:013880767030 at 210.52.252.108;tag=2812410921537011883127973 Call-ID: MOTwWfT8guaDRb35fe-SNYEntSNolOtM CSeq: 14946 ACK Content-Length: 0 --end msg--Call 0 state=CONFIRMED RX 322 bytes Request msg BYE/cseq=2 (rdata0x71da04) from UDP 210.52.252.108:5060: BYE sip:wsz at 210.52.252.108;tag=oMiS2VsfsgIMlTWFdT0v7HcC9RZ7O9X- SIP/2.0 CSeq: 2 BYE Via: SIP/2.0/UDP 210.52.252.108:5060 From: sip:013880767030@210.52.252.108;tag=2812410921537011883127973 Call-ID: MOTwWfT8guaDRb35fe-SNYEntSNolOtM To: sip:wsz at 210.52.252.108;tag=oMiS2VsfsgIMlTWFdT0v7HcC9RZ7O9X- Content-Length: 0 --end msg--TX 294 bytes Response msg 200/BYE/cseq=2 (tdta0x751c50) to UDP 210.52.252.108:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 210.52.252.108:5060;received=210.52.252.108 Call-ID: MOTwWfT8guaDRb35fe-SNYEntSNolOtM From: <sip:013880767030@210.52.252.108>;tag=2812410921537011883127973 To: <sip:wsz at 210.52.252.108>;tag=oMiS2VsfsgIMlTWFdT0v7HcC9RZ7O9X- CSeq: 2 BYE Content-Length: 0 Can you help me to explain the reason why the server or the callee send the BYE request to the caller, result that the call be stopped? Thanks very much! Best regards. Bob Soong _________________________________________________________________ Windows Live: Make it easier for your friends to see what you?re up to on Facebook. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_2:092009 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20100103/8942da41/attachment.html>
