Hi ,
my scenario was like below , those signal was printed by proxy server .
INVITE sip:87262 at 192.168.17.151 SIP/2.0
Via: SIP/2.0/TCP 192.168.23.200:2535;rport;branch=z9hG4bKPj2c40f9ed7fb641fdb73ce8f59c947adb;alias
Max-Forwards: 70
From: sip:87261@192.168.17.151;tag=6588e8b2d8f943dfb5700cf4dee0e8f4
To: sip:87262 at 192.168.17.151
Contact: <sip:87261 at 192.168.23.200:2535;transport=TCP;ob>;+sip.ice
Call-ID: be6d14cfd3de40bcb0d4a8d204f9c33a
CSeq: 22670 INVITE
Route: <sip:192.168.17.151:5060;transport=tcp;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
Content-Type: application/sdp
Content-Length: 705
*******(the sdp part)
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.23.200:2535;rport=5060;received=192.168.17.151;branch=z9hG4bKPj2c40f9ed7fb641fdb73ce8f59c947adb;alias
Call-ID: be6d14cfd3de40bcb0d4a8d204f9c33a
From: <sip:87261@192.168.17.151>;tag=6588e8b2d8f943dfb5700cf4dee0e8f4
To: <sip:87262 at 192.168.17.151>
CSeq: 22670 INVITE
Content-Length: 0
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 192.168.23.200:2535;rport=5060;received=192.168.17.151;branch=z9hG4bKPj2c40f9ed7fb641fdb73ce8f59c947adb;alias
Call-ID: be6d14cfd3de40bcb0d4a8d204f9c33a
From: <sip:87261@192.168.17.151>;tag=6588e8b2d8f943dfb5700cf4dee0e8f4
To: <sip:87262 at 192.168.17.151>;tag=e15600f5b36c43c987c017702c00df04
CSeq: 22670 INVITE
Contact: <sip:87262 at 192.168.23.138:5858;transport=TCP;ob>;+sip.ice
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.23.200:2535;rport=5060;received=192.168.17.151;branch=z9hG4bKPj2c40f9ed7fb641fdb73ce8f59c947adb;alias
Call-ID: be6d14cfd3de40bcb0d4a8d204f9c33a
From: <sip:87261@192.168.17.151>;tag=6588e8b2d8f943dfb5700cf4dee0e8f4
To: <sip:87262 at 192.168.17.151>;tag=e15600f5b36c43c987c017702c00df04
CSeq: 22670 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Contact: <sip:87262 at 192.168.23.138:5858;transport=TCP;ob>;+sip.ice
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length: 507
*****(sdp part)
those signal are seems ok, and the UAC received the OK signal.
i check the detail invite signal it has router hdr , but the rest of signal message without router hdr , but my proxy still got it ,
so i guess the 100 Try and 180 ring use the same tcp connection ,but when UAC response ACK it seem use another connection ?
???? Harald Radke
????? 2015-03-17 16:43
???? pjsip
??? Re: [pjsip]??: PJSIP UA2 API lost ack?
Hi,
could you post INVITE, any provisional answer and the OK as well?
Regards,
Harry
Gesendet: Dienstag, 17. M?rz 2015 um 09:36 Uhr
Von: zhuyongwen at made-in-china.com
An: pjsip <pjsip at lists.pjsip.org>
Betreff: [pjsip] ??: PJSIP UA2 API lost ack?
i have debug the pjsip source file ,found the pjsip had sent out ACK like below :
ACK sip:87262 at 192.168.23.138:5774;transport=TCP;ob SIP/2.0
Via: SIP/2.0/TCP 192.168.23.200:2503;rport;branch=z9hG4bKPj32d7f3c7b48c414bb2426f2bc4eef931;alias
Max-Forwards: 70
From: sip:87261@192.168.17.151;tag=a30e9922b5a44411bcc832acf0557f58
To: sip:87262 at 192.168.17.151;tag=1b2d0905aed44dd1b8a35f3e6243554b
Call-ID: f8cd6035b2d74e92a6fe2f7043313c7c
CSeq: 32447 ACK
Content-Length: 0
but in my application i use tcp connection with 192.168.17.151 as my proxy ?
in this ack signal, obviously it miss the router hdr , so the ack messge can't sent out to proxy ?
so my question is , why the other signal will include the router hdr, why ack without the hdr ? it's a pjsip bug ?
???? zhuyongwen at made-in-china.com
????? 2015-03-17 15:37
???? pjsip
??? PJSIP UA2 API lost ack?
hi all,
My application used pjsip ua2 api to deal with sip message, in the scenario i answer the coming call ,and voice bridge was connect success, but the UAS will timeout for a moment .
i check the sip signal message, when answer 200 ok , UAC should send ACK signal, but it seem doen't send out by pjsip ?
is there something i missing when i answer the call ? or UAC should do something after received 200 ok signal ?
best regards!
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