Hi,
I am prototyping an application using pjsua2 Java APIs on the Android platform.
I have a question about Call#getState at onIncomingCall timing.
1. MyAccount#onIncomingCall(OnIncomingCallParam p) called.
2. new MyCall(this, p.getCallId())
The Call#getState value of the above Call object is pjsip_inv_state.PJSIP_INV_STATE_NULL.
I expected pjsip_inv_state.PJSIP_INV_STATE_INCOMING.
I am not in trouble, but I felt a bit strange.
Is this a specification?
I am using PJSIP tags/2.7.2.
I do not know whether it will be helpful, but paste the log below.
# The last line is the called callback's logging.
06-22 18:36:02.323 V/PJSIP#2.7.2(10005): pjsua_acc.c Sending 2 bytes keep-alive packet for acc 0 to 192.168.75.101:5060 [10029@#write 584]
06-22 18:36:02.341 V/PJSIP#2.7.2(10005): tdta0xd044b064 Destroying txdata raw [10029@#write 584]
06-22 18:36:15.642 V/PJSIP#2.7.2(10005): sip_endpoint.c Processing incoming message: Request msg INVITE/cseq=102 (rdata0xd0870014) [10029@#write 584]
06-22 18:36:15.643 V/PJSIP#2.7.2(10005): pjsua_core.c .RX 930 bytes Request msg INVITE/cseq=102 (rdata0xd0870014) from UDP 192.168.75.101:5060:
06-22 18:36:15.643 V/PJSIP#2.7.2(10005): INVITE sip:204@192.168.75.1:62013;ob SIP/2.0
06-22 18:36:15.643 V/PJSIP#2.7.2(10005): Via: SIP/2.0/UDP 192.168.75.101:5060;branch=z9hG4bK5be093eb;rport
06-22 18:36:15.643 V/PJSIP#2.7.2(10005): Max-Forwards: 70
06-22 18:36:15.643 V/PJSIP#2.7.2(10005): From: "?Ä?c ?À?õ?F [my Windows]" <sip:201@192.168.75.101>;tag=as41c77a10
06-22 18:36:15.643 V/PJSIP#2.7.2(10005): To: <sip:204@192.168.75.1:62013;ob>
06-22 18:36:15.643 V/PJSIP#2.7.2(10005): Contact: <sip:201@192.168.75.101:5060>
06-22 18:36:15.643 V/PJSIP#2.7.2(10005): Call-ID: 45f6de177a9ed89460bef071758d8478@192.168.75.101:5060
06-22 18:36:15.643 V/PJSIP#2.7.2(10005): CSeq: 102 INVITE
06-22 18:36:15.643 V/PJSIP#2.7.2(10005): User-Agent: Asterisk PBX 13.18.3~dfsg-1ubuntu4
06-22 18:36:15.643 V/PJSIP#2.7.2(10005): Date: Fri, 22 Jun 2018 09:36:17 GMT
06-22 18:36:15.643 V/PJSIP#2.7.2(10005): Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
06-22 18:36:15.643 V/PJSIP#2.7.2(10005): Supported: replaces, timer
06-22 18:36:15.643 V/PJSIP#2.7.2(10005): Content-Type: application/sdp
06-22 18:36:15.643 V/PJSIP#2.7.2(10005): Content-Length: 303
06-22 18:36:15.643 V/PJSIP#2.7.2(10005):
06-22 18:36:15.643 V/PJSIP#2.7.2(10005): v=0
06-22 18:36:15.643 V/PJSIP#2.7.2(10005): o=root 324205171 324205171 IN IP4 192.168.75.101
06-22 18:36:15.643 V/PJSIP#2.7.2(10005): s=Asterisk PBX 13.18.3~dfsg-1ubuntu4
06-22 18:36:15.643 V/PJSIP#2.7.2(10005): c=IN IP4 192.168.75.101
06-22 18:36:15.643 V/PJSIP#2.7.2(10005): t=0 0
06-22 18:36:15.643 V/PJSIP#2.7.2(10005): m=audio 10032 RTP/AVP 0 8 3 101
06-22 18:36:15.643 V/PJSIP#2.7.2(10005): a=rtpmap:0 PCMU/8000
06-22 18:36:15.643 V/PJSIP#2.7.2(10005): a=rtpmap:8 PCMA/8000
06-22 18:36:15.643 V/PJSIP#2.7.2(10005): a=rtpmap:3 GSM/8000
06-22 18:36:15.643 V/PJSIP#2.7.2(10005): a=rtpmap:101 telephone-event/8000
06-22 18:36:15.643 V/PJSIP#2.7.2(10005): a=fmtp:101 0-16
06-22 18:36:15.643 V/PJSIP#2.7.2(10005): a=maxptime:150
06-22 18:36:15.643 V/PJSIP#2.7.2(10005): a=sendrecv
06-22 18:36:15.643 V/PJSIP#2.7.2(10005):
06-22 18:36:15.643 V/PJSIP#2.7.2(10005): --end msg-- [10029@#write 584]
06-22 18:36:15.646 V/PJSIP#2.7.2(10005): pjsua_call.c .Incoming Request msg INVITE/cseq=102 (rdata0xd0870014) [10029@#write 584]
06-22 18:36:15.649 V/PJSIP#2.7.2(10005): tsx0xd04f3864 ...Transaction created for Request msg INVITE/cseq=102 (rdata0xd0870014) [10029@#write 584]
06-22 18:36:15.651 V/PJSIP#2.7.2(10005): tsx0xd04f3864 ..Incoming Request msg INVITE/cseq=102 (rdata0xd0870014) in state Null [10029@#write 584]
06-22 18:36:15.652 V/PJSIP#2.7.2(10005): tsx0xd04f3864 ...State changed from Null to Trying, event=RX_MSG [10029@#write 584]
06-22 18:36:15.653 V/PJSIP#2.7.2(10005): dlg0xd0873864 ....Transaction tsx0xd04f3864 state changed to Trying [10029@#write 584]
06-22 18:36:15.656 V/PJSIP#2.7.2(10005): dlg0xd0873864 ..UAS dialog created [10029@#write 584]
06-22 18:36:15.659 V/PJSIP#2.7.2(10005): dlg0xd0873864 ..Module mod-invite added as dialog usage, data=0xd0454a14 [10029@#write 584]
06-22 18:36:15.661 V/PJSIP#2.7.2(10005): dlg0xd0873864 ...Session count inc to 3 by mod-invite [10029@#write 584]
06-22 18:36:15.662 V/PJSIP#2.7.2(10005): inv0xd0873864 ..UAS invite session created for dialog dlg0xd0873864 [10029@#write 584]
06-22 18:36:15.664 V/PJSIP#2.7.2(10005): dlg0xd0873864 ...Session count inc to 3 by mod-pjsua [10029@#write 584]
06-22 18:36:15.665 V/PJSIP#2.7.2(10005): pjsua_media.c ..Call 0: initializing media.. [10029@#write 584]
06-22 18:36:15.671 V/PJSIP#2.7.2(10005): pjsua_media.c ...RTP socket reachable at 192.168.200.2:40004 [10029@#write 584]
06-22 18:36:15.672 V/PJSIP#2.7.2(10005): pjsua_media.c ...RTCP socket reachable at 192.168.200.2:40005 [10029@#write 584]
06-22 18:36:15.675 V/PJSIP#2.7.2(10005): srtp0xd0391800 ...SRTP keying SDES created [10029@#write 584]
06-22 18:36:15.676 V/PJSIP#2.7.2(10005): pjsua_media.c ...Media index 0 selected for audio call 0 [10029@#write 584]
06-22 18:36:15.677 V/PJSIP#2.7.2(10005): pjsua_media.c ...Call 0: media transport initialization complete: Success [10029@#write 584]
06-22 18:36:15.680 D/Extension#0.1(10005): MyCall{Id=0, isActive=true, hasMedia=false, Digest=CallInfo{Id=0, Role=PJSIP_ROLE_UAS, AccId=0, LocalUri=<sip:204@192.168.75.1;ob>, localContact=<sip:204@192.168.75.1:62013;ob>, RemoteUri="?Ä?c ?À?õ?F [my Windows]" <sip:201@192.168.75.101>, RemoteContact=<sip:201@192.168.75.101:5060>, CallIdString=45f6de177a9ed89460bef071758d8478@192.168.75.101:5060, Setting=CallSetting{Flag=4, ReqKeyframeMethod=0, AudioCount=1, VideoCount=0, isEmpty=false}, State=PJSIP_INV_STATE_NULL, StateText=NULL, LastStatusCode=PJSIP_SC_NULL, LastReason=}} [10029@Facade#onIncomingCall 455]
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