Hi Mathieu, I'm really not a SIP expert, but my reading of this is that the bad contact address causes the PRACK to not be sent and the other endpoint keeps sending the 183 and finally gives up and sends a BYE. But the contact address looks OK. Can you do an nslookup of pcgw-0005.xxx.wiz.net and see if that works on your machine? Regards, Bill On 2/26/2015 9:59 AM, Mathieu Trinc wrote: > Hi Bill, > > The error occurs when PJSIP tries to resolve a hostname present in the Contact field of the first '183 Statut progress' message received: > > Contact: <sip:lucentNGFS-000116 at pcgw-0005.xxx.wiz.net:5060;encoded-parm=QbkRBthOEgsTXgkTBA0HHiUrKz1CQEFCQkVDNgQMGAlsMTcgK2ghOyAnOCs.ITogYX9jZmR4NjsxblJGQAQEGF5VSh5dSBNMDQERXFJYX1o_> > > Could this error explain why I keep receiving these messages? > > I set the log verbosity level to 6 but there is not much information (Is there a way to get more detailed information?): > > 09:46:21.703 pjsua_core.c !.pjsua version 2.3 for win32-6.1/i386/msvc-18.0 initialized > 09:46:21.711 main.c Ready: Success > 09:46:22.002 pjsua_acc.c !....sip:from at foo.org: registration success, status=200 (OK), will re-register in 300 seconds > 09:46:40.171 pjsua_app.c !..Turning sound device ON > 09:46:40.222 pjsua_app.c .......Call 0 state changed to CALLING > 09:46:42.919 pjsua_app.c !.....Call 0 state changed to EARLY (183 Session Progress) > 09:46:42.976 tsx005CBDEC ......Failed to send Request msg PRACK/cseq=18743 (tdta005CAD80)! err=70018 (gethostbyname() has returned error (PJ_ERESOLVE)) > 09:46:43.494 pjsua_app.c .....Call 0 state changed to EARLY (183 Session Progress) > 09:46:44.593 pjsua_app.c .....Call 0 state changed to EARLY (183 Session Progress) > 09:46:46.694 pjsua_app.c .....Call 0 state changed to EARLY (183 Session Progress) > 09:46:50.795 pjsua_app.c .....Call 0 state changed to EARLY (183 Session Progress) > 09:46:53.696 pjsua_app.c .....Call 0 is DISCONNECTED [reason=487 (Request Terminated)] > 09:46:53.696 pjsua_app_comm ..... > [DISCONNCTD] To: sip:to at foo.org;tag=54596251-54ef31642d05c34a-gm-po-lucentPCSF-152762 > Call time: 00h:00m:00s, 1st res in 2701 ms, conn in 0ms > #0 audio PCMA @8kHz, sendrecv, peer=85.31.200.0:48546 > SRTP status: Not active Crypto-suite: > RX pt=8, last update:00h:00m:00.684s ago > total 508pkt 80.0KB (100.4KB +IP hdr) @avg=59.4Kbps/74.5Kbps > pkt loss=1 (0.2%), discrd=0 (0.0%), dup=0 (0.0%), reord=0 (0.0%) > (msec) min avg max last dev > loss period: 20.000 20.000 20.000 20.000 0.000 > jitter : 0.000 0.291 1.500 0.125 0.210 > TX pt=8, ptime=20, last update:00h:00m:00.818s ago > total 33pkt 5.2KB (6.6KB +IP hdr) @avg=3.9Kbps/4.8Kbps > pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%) > (msec) min avg max last dev > loss period: 0.000 0.000 0.000 0.000 0.000 > jitter : 123.500 275.938 428.375 428.375 56.755 > RTT msec : 154.000 165.500 177.000 177.000 11.500 > 09:46:54.696 pjsua_app.c .Turning sound device OFF > 09:46:57.188 pjsua_acc.c !.....sip:from at foo.org: unregistration success > > > Thanks ! > > > Mathieu > > > > Sent: Thursday, February 26, 2015 at 2:45 PM > From: "Bill Gardner" <billg@xxxxxxxxxxxx> > To: pjsip at lists.pjsip.org > Subject: Re: Strange behavior when making a call (format fix) > Hi Mathieu, > > I think you should track down the source of the PJ_ERESOLVE error. Can > you send a complete log? > > Regards, > > Bill > > On 2/26/2015 6:29 AM, Mathieu Trinc wrote: >> (Sorry for the previous email which was in HTML format) >> >> Hello everyone, >> >> I'm trying to use pjsua to make a call. The registration process works fine but when I make a call I notice a strange behavior: the INVITE message is sent, I receive a "Status: 100 Trying" message >> and then I keep receiving "Status: 183 Session Progress" messages until the recipient answers. The communication is then established but I think some SIP messages are missing (for instance there is no 200 OK message) >> and this leads to an incorrect state (I do not know when the call is established, how long it lasts, etc.). >> >> (I tried to make the same call with a different softphone (XLite) and I get a different behavior: >> INVITE - 100 Trying - 183 Session Progress - 180 Ringing. And when the recipient answers I get 200 OK) >> >> Any help to understand what is going on and how to get the standard behavior would be greatly appreciated. >> >> Here is the log displayed in pjsua and the details of some SIP messages (slightly modified for privacy concerns): >> >> 02:31:35.649 pjsua_app.c !..Turning sound device ON >> 02:31:35.699 pjsua_app.c .......Call 0 state changed to CALLING >> 02:31:39.446 pjsua_app.c !.....Call 0 state changed to EARLY (183 Session Progress) >> 02:31:39.448 tsx0060CD0C ......Failed to send Request msg PRACK/cseq=17029 (tdta0060BCA0)! err=70018 (gethostbyname() has returned error (PJ_ERESOLVE)) >> 02:31:39.981 pjsua_app.c .....Call 0 state changed to EARLY (183 Session Progress) >> 02:31:41.081 pjsua_app.c .....Call 0 state changed to EARLY (183 Session Progress) >> 02:31:43.181 pjsua_app.c .....Call 0 state changed to EARLY (183 Session Progress) >> 02:31:47.280 pjsua_app.c .....Call 0 state changed to EARLY (183 Session Progress) >> >> // The recipient answers and we can communicate >> >> // I decide to hang up (if it is the recipient that hangs up I get this after a few seconds: Call 0 is DISCONNECTED [reason=500 (Server Inernal Error)]) >> h >> >> 02:31:54.231 pjsua_app.c .....Call 0 is DISCONNECTED [reason=487 (Request Terminated)] >> 02:31:54.232 pjsua_app_comm ..... >> [DISCONNCTD] To: sip:to at foo.org;tag=54596251 >> -54ee770c1f6962c8-gm-po-lucentPCSF-055498 >> Call time: 00h:00m:00s, 1st res in 3750 ms, conn in 0ms >> #0 audio PCMA @8kHz, sendrecv, peer=85.31.200.0:47386 >> SRTP status: Not active Crypto-suite: >> RX pt=8, last update:00h:00m:04.702s ago >> total 706pkt 112.6KB (140.8KB +IP hdr) @avg=60.9Kbps/76.2Kbps >> pkt loss=0 (0.0%), discrd=0 (0.0%), dup=0 (0.0%), reord=0 (0.0%) >> (msec) min avg max last dev >> loss period: 0.000 0.000 0.000 0.000 0.000 >> jitter : 0.000 0.206 1.250 0.250 0.137 >> TX pt=8, ptime=20, last update:00h:00m:04.838s ago >> total 34pkt 5.4KB (6.8KB +IP hdr) @avg=2.9Kbps/3.6Kbps >> pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%) >> (msec) min avg max last dev >> loss period: 0.000 0.000 0.000 0.000 0.000 >> jitter : 82.000 225.063 368.125 368.125 56.755 >> RTT msec : 175.000 233.500 292.000 175.000 56.755 >> 02:31:55.237 pjsua_app.c .Turning sound device OFF >> >> INVITE sip:to at foo.org SIP/2.0 >> Via: SIP/2.0/UDP 192.168.1.25:5060;rport;branch=z9hG4bKPja40ebf7032174a939b836f6a4030dfd9 >> Max-Forwards: 70 >> From: sip:from@xxxxxxx;tag=bc2c850cf705496cabe58b4a071794fb >> To: sip:to at foo.org >> Contact: <sip:from at 192.168.1.25:5060;ob> >> Call-ID: 635050fa9ef7442990b7809c34a2328b >> CSeq: 17028 INVITE >> Route: <sip:p-cscf.wiz.net:5060;lr> >> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS >> Supported: replaces, 100rel, timer, norefersub >> Session-Expires: 1800 >> Min-SE: 90 >> User-Agent: PJSUA v2.3 win32-6.1/i386/msvc-18.0 >> Content-Type: application/sdp >> Content-Length: 473 >> v=0 >> o=- 3633906695 3633906695 IN IP4 192.168.1.25 >> s=pjmedia >> b=AS:84 >> t=0 0 >> a=X-nat:0 >> m=audio 4000 RTP/AVP 98 97 99 104 3 0 8 9 96 >> c=IN IP4 192.168.1.25 >> b=TIAS:64000 >> a=rtcp:4001 IN IP4 192.168.1.25 >> a=sendrecv >> a=rtpmap:98 speex/16000 >> a=rtpmap:97 speex/8000 >> a=rtpmap:99 speex/32000 >> a=rtpmap:104 iLBC/8000 >> a=fmtp:104 mode=30 >> a=rtpmap:3 GSM/8000 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:9 G722/8000 >> a=rtpmap:96 telephone-event/8000 >> a=fmtp:96 0-16 >> >> SIP/2.0 100 Trying >> Call-ID: 635050fa9ef7442990b7809c34a2328b >> Via: SIP/2.0/UDP 192.168.1.25:5060;received=192.168.1.25;branch=z9hG4bKPja40ebf7032174a939b836f6a4030dfd9;rport=5060 >> To: <sip:to at foo.org;user=phone> >> From: <sip:from@xxxxxxx;user=phone>;tag=bc2c850cf705496cabe58b4a071794fb >> CSeq: 17028 INVITE >> Date: Thu, 26 Feb 2015 01:29:48 GMT >> Server: Alcatel-Lucent-HPSS/3.0.3 >> Content-Length: 0 >> >> SIP/2.0 183 Session Progress >> Call-ID: 635050fa9ef7442990b7809c34a2328b >> Via: SIP/2.0/UDP 192.168.1.25:5060;received=192.168.1.25;branch=z9hG4bKPja40ebf7032174a939b836f6a4030dfd9;rport=5060 >> To: <sip:to at foo.org;user=phone>;tag=54596251-54ee770c1f6962c8-gm-po-lucentPCSF-055498 >> From: <sip:from@xxxxxxx;user=phone>;tag=bc2c850cf705496cabe58b4a071794fb >> CSeq: 17028 INVITE >> Require: 100rel >> Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,PRACK,INFO,REFER,UPDATE >> Contact: <sip:lucentNGFS-000116 at pcgw-0005.xxx.wiz.net:5060;encoded-parm=QbkRBthOEgsTXgkTBA0HHiUrKz1CQEFCQkVDNgQMGAlsMTcgK2ghOyAnOCs.ITogYX9jZmR4NjsxblJGQAQEGF5VSh5dSBNMDQERXFJYX1o_> >> Content-Type: application/sdp >> RSeq: 1 >> Server: Alcatel-Lucent-HPSS/3.0.3 >> Content-Length: 220 >> v=0 >> o=LucentPCSF 113972553 113972553 IN IP4 .wiz.net >> s=- >> c=IN IP4 85.31.200.0 >> t=0 0 >> m=audio 47386 RTP/AVP 8 101 >> a=rtpmap:101 telephone-event/8000 >> a=sendrecv >> a=ptime:20 >> a=maxptime:30 >> >> Thank you ! >> >> Mathieu >> >> _______________________________________________ >> 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[http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org] > > _______________________________________________ > Visit our blog: http://blog.pjsip.org[http://blog.pjsip.org] > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org[http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org] > > _______________________________________________ > 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
