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 at wavearts.com>
To:?pjsip at lists.pjsip.org
Subject:?Re: [pjsip] 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
>
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