qiao236349845 / doubango

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how to receive and handle the "UPDATE" message in SIP? #46

Closed GoogleCodeExporter closed 9 years ago

GoogleCodeExporter commented 9 years ago
What steps will reproduce the problem?
1.make a call to a phone which is binding with other number, and it send back a 
update message. 
2.
3.

What is the expected output? What do you see instead?
ignore the update package,hand off the phone.

What version of the product are you using? On what operating system?
version:doubangoSVN578 ,os:android2.2.

Please provide any additional information below.
how to response the "UPDATE" message?
thanks,sir.

Original issue reported on code.google.com by akari9...@163.com on 20 Jun 2011 at 2:01

GoogleCodeExporter commented 9 years ago
You can receive UPDATE message in java layer but you cannot respond to it 
unless you handle it in ANSI-C code.
It's relatively easy to change the FSM in "tsip_dialog_invite.c" if you want to 
respond to UPDATE from java.

From java:
Subclass "SipCallback::onInviteEvent" and listen for "tsip_i_request" event.

Original comment by boss...@yahoo.fr on 20 Jun 2011 at 8:02

GoogleCodeExporter commented 9 years ago
Hi,sir.

There are some waring came out,when i received a UPDATE msg(RFC3261), the debug 
msg as follow:
1)tsip_parser_header.c 5693
TSK_DEBUG_WARN("parse_header_Content_Disposition NOT IMPLEMENTED. Will be added 
as Dummy header.");
2)tsk_fsm.c 190行
TSK_DEBUG_WARN("State machine: No matching state found.");

how to fix this problem? can you give me some tips or any direction . thx a lot.

My UPDATE msg header as follow.thank you again.
pmC^@E0$=G}MkrcUPDATE sip:+8659188244165@125.77.107.242:29352;transport=udp 
SIP/2.0
Via: SIP/2.0/UDP 61.131.4.71:5060;branch=z9hG4bKcb1c46c6407450ce0b5btaN0
To: <sip:+8659188244165@fj.ctcims.cn>;tag=95259306
From: 
<sip:+8659188244180@fj.ctcims.cn>;tag=ztesipdUiBYP89RlYA663*1-1-20481*ebhj.1
Call-ID: 8ae81141-88c8-6050-22e6-689ff94f8b98
CSeq: 1001 UPDATE
Max-Forwards: 13
Contact: <sip:61.131.4.71:5060;zte-did=1-1-20481-2044-12>
Supported: timer
User-Agent: ZTE Softswitch/1.0.0
Session-Expires: 3600
Content-Type: application/sdp
Content-Length: 206
Content-Disposition: session

v=0
o=ZTE 135 1120747413 IN IP4 61.131.4.74
s=phone-call
c=IN IP4 61.131.4.74
t=0 0
m=audio 39204 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
a=ptime:20
a=rtpmap:8 PCMA/8000/1

Original comment by akari9...@163.com on 21 Jun 2011 at 3:48

GoogleCodeExporter commented 9 years ago
Here's my call packet file.

Original comment by akari9...@163.com on 22 Jun 2011 at 9:09

Attachments:

GoogleCodeExporter commented 9 years ago
Have been resolved in revision 610.
thanks a lot.

Original comment by akari9...@163.com on 7 Jul 2011 at 9:56

GoogleCodeExporter commented 9 years ago
you use 
ZTE softswitch from china ,h��
�ha...

Original comment by openser@yeah.net on 15 Jul 2011 at 8:17

GoogleCodeExporter commented 9 years ago
We got the same problem with ZTE IMS core used by a European Telco operator.
They are using reINVITE when the request get forked.

Original comment by boss...@yahoo.fr on 18 Jul 2011 at 2:05