Strange behavior when making a call (format fix)

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(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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