pjsua2 - reinvite sends audio inactive

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hi all,
I figure it out what was wrong on my side. I don't understand documentation
very well. I was trying send reinvite with:
 CallOpParam prm = new CallOpParam();
 prm.setOptions(pjsua_call_flag.PJSUA_CALL_UNHOLD.swigValue());
 call.reinvite(prm);
and good way is
 CallOpParam prm = new CallOpParam();
 CallSetting opt = prm.getOpt();
 opt.setAudioCount(1);
 opt.setVideoCount(0);
 opt.setFlag(pjsua_call_flag.PJSUA_CALL_UNHOLD.swigValue());
 call.reinvite(prm);

sorry for spam :)
br
Andrzej

2015-03-17 14:23 GMT+01:00 frogersik <frogersik at gmail.com>:

> I paste response but anyway its something wrong here.
> hold:
> pjsua_call.c !Putting call 0 on hold
> pjsua_core.c  ....TX 918 bytes Request msg INVITE/cseq=3 (tdta0x7a0a5318)
> to UDP xx.x.x.xx:5062:
>     INVITE sip:xx.x.x.xx:5062 SIP/2.0
>     Via: SIP/2.0/UDP 192.0.0.4:6000
> ;rport;branch=z9hG4bKPjaVYBN1fg7dfx-Iy2jSVxm16YHYmP1qsCR
>     Max-Forwards: 70
>     From: sip:4xxxxxxxxxxx@xxxxxxxxxxxx
> ;tag=aVYBN1fg7dfx-Iy2jSVxm16YHYmP1qsCR
>     To: sip:+4yyyyyyyyyyy at neofon.tp.pl
> ;tag=52560051-14265979364676-gm-po-lucentPCSF-064891
>     Contact: <sip:4xxxxxxxxxxx at 192.0.0.4:6000>
>     Call-ID: aVYBN1fg7dfx-Iy2jSVxm16YHYmP1qsCR
>     CSeq: 3 INVITE
>     Allow: INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
> REFER, MESSAGE, OPTIONS
>     User-Agent: PJSIP/2.3-svn
>     Content-Type: application/sdp
>     Content-Length:   313
>     v=0
>     o=- 3635586736 3635586737 IN IP4 192.0.0.4
>     s=pjmedia
>     b=AS:84
>     t=0 0
>     a=X-nat:0
>     m=audio 4000 RTP/AVP 9 8 0 101
>     c=IN IP4 192.0.0.4
>     b=TIAS:64000
>     a=rtcp:4001 IN IP4 192.0.0.4
>     a=rtpmap:9 G722/8000
>     a=rtpmap:8 PCMA/8000
>     a=rtpmap:0 PCMU/8000
>     a=rtpmap:101 telephone-event/8000
>     a=fmtp:101 0-16
>     a=sendonly
>     --end msg--
>
> reinvite:
> pjsua_call.c !Sending re-INVITE on call 0
> pjsua_core.c  ....TX 918 bytes Request msg INVITE/cseq=4 (tdta0x7a0b8638)
> to UDP xx.x.x.xx:5062:
>     INVITE sip:xx.x.x.xx:5062 SIP/2.0
>     Via: SIP/2.0/UDP 192.0.0.4:6000
> ;rport;branch=z9hG4bKPjaVYBN1fg7dfx-Iy2jSVxm16YHYmP1qsCR
>     Max-Forwards: 70
>     From: sip:4xxxxxxxxxxx@xxxxxxxxxxxx
> ;tag=aVYBN1fg7dfx-Iy2jSVxm16YHYmP1qsCR
>     To: sip:+4yyyyyyyyyyy at neofon.tp.pl
> ;tag=52560051-14265979364676-gm-po-lucentPCSF-064891
>     Contact: <sip:4xxxxxxxxxxx at 192.0.0.4:6000>
>     Call-ID: aVYBN1fg7dfx-Iy2jSVxm16YHYmP1qsCR
>     CSeq: 4 INVITE
>     Allow: INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
> REFER, MESSAGE, OPTIONS
>     User-Agent: PJSIP/2.3-svn
>     Content-Type: application/sdp
>     Content-Length:   313
>     v=0
>     o=- 3635586736 3635586738 IN IP4 192.0.0.4
>     s=pjmedia
>     b=AS:84
>     t=0 0
>     a=X-nat:0
>     m=audio 4000 RTP/AVP 9 8 0 101
>     c=IN IP4 192.0.0.4
>     b=TIAS:64000
>     a=rtcp:4001 IN IP4 192.0.0.4
>     a=rtpmap:9 G722/8000
>     a=rtpmap:8 PCMA/8000
>     a=rtpmap:0 PCMU/8000
>     a=rtpmap:101 telephone-event/8000
>     a=fmtp:101 0-16
>     a=inactive
>     --end msg--
>
> it should be sendrecv instead inactive?
>
> 2015-03-17 12:43 GMT+01:00 frogersik <frogersik at gmail.com>:
>
>> hi, if I use call.reinvite with any param, messege will be send but audio
>> will be always in inactive state. here is log from app:
>> Response msg 200/INVITE/cseq=4 (rdata0x7a71a424) from UDP
>> xx.xx.xx.xx:5062:
>>     SIP/2.0 200 OK
>>     Via: SIP/2.0/UDP
>> kk.kk.kk.kk:6000;received=212.160.230.3;branch=z9hG4bKPj401k1aYWjzp6ATDoPyrkt2P3fds6BMN6O0C-PWjoO;rport=6000
>>     From: sip:48xxxxxxxxx@xxxxxxxxx
>> ;tag=401k1aYWjzp6ATDoPyrkt2P3fds6BMN6O0C
>>     To: sip:+48xxxxxxxxx at server.pl
>> ;tag=536caa4d-1426592075550893-gm-po-lucentPCSF-034732
>>     Call-ID: 401k1aYWjzp6ATDoPyrkt2P3fds6BMN6O0C
>>     CSeq: 4 INVITE
>>     Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
>>     Supported:
>>     Accept:
>> application/dtmf-relay,application/media_control+xml,application/sdp
>>     Contact: <sip:XX.XX.xx.xx:5062>
>>     Content-Type: application/sdp
>>     Content-Length: 253
>>     Server: Alcatel-Lucent-HPSS/3.0.3
>>     v=0
>>     o=xxxxxxx.pl
>>     s=-
>>     c=IN IP4 0.0.0.0
>>     t=0 0
>>     m=audio 16778 RTP/AVP 9 101
>>     c=IN IP4 0.0.0.0
>>     b=TIAS:64000
>>     a=rtpmap:9 G722/8000
>>     a=rtpmap:101 telephone-event/8000
>>     a=fmtp:101 0-16
>>     a=inactive
>>
>
>
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