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
