crash APS-Direct

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Hi all, 

I?m successful solved disconnect call after 24.9 minutes,
disable timer session in freeswitch and establish call 8 hours, problem not
occur.



Thanks all.


From: e-mohhassan@xxxxxxxxxxx
To: pjsip at lists.pjsip.org
Date: Sun, 19 Apr 2009 12:24:53 +0000
Subject: Re: crash APS-Direct










thanks Fabio,

in my tests, INVITE Message contain Session-Expires: 120 (Keep alive message between
client and server)

FULL MESSAGE:

INVITE sip:126 at 192.168.0.6:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.0.3;rport;branch=z9hG4bKKtD3vFZ3Qr7Fp

Max-Forwards: 70

From: <sip:125@192.168.0.3>;tag=XvSNHN74Q2e5r

To: <sip:126 at 192.168.0.3>;tag=mb7-7oZqmhhClf1Ug.7f9JJIG11iglYC

Call-ID: wCnoaaKG0VNZUsIu7D7mPEC2sMaN3KuO

CSeq: 113708767 INVITE

Contact:
<sip:mod_sofia at 192.168.0.3:5060;transport=udp>

User-Agent: FreeSWITCH-mod_sofia/1.0.1-exported

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH

Supported: 100rel, timer, precondition, path, replaces

Session-Expires: 120

Min-SE: 120

Content-Type: application/sdp

Content-Disposition: session

Content-Length: 344

 

v=0

o=- 4689126742759932103 2121988458920195298 IN IP4 192.168.0.3

s=pjmedia

c=IN IP4 192.168.0.4

t=0 0

a=X-nat:0

m=audio 4000 RTP/SAVP 102 101

a=rtpmap:102 speex/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=rtcp:4001 IN IP4 192.168.0.4

a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:1ahJYSwy4DYbmm9y2UG0L/drhbBNNRC9Avt8czlN

 

Is keep alive message response from destroy call ? because if
lost keep alive message, my be server to send bye to caller and callee.

I need some suggestions to help me solved this problem.   



Date: Thu, 16 Apr 2009 21:04:13 +0200
From: lists@xxxxxxxxxxxxxxx
To: pjsip at lists.pjsip.org
Subject: Re: crash APS-Direct






  
  


Most probablere there's a memory leak somewhere in the code or some
counter overflowing.



To track down where the bug is please follow this information:

http://www.newlc.com/Tracking-down-memory-leaks.htm

https://developer.symbian.com/wiki/pages/viewpage.action?pageId=432l



Memory leak on symbian are one of the worst things that can be there.



Mohamed, benny, you are the only ones who can make it more stable.



Mohamed, can you provide much details by debugging the memory leaks and
at the same time report the bug with full log details?



Fabio



mohamed hassan wrote:

  
  
  
  
  
  
  
  

  
  Hi all,
  Still I?m working in call disconnect problem,
tried some
tests:
   First, caller and
callee PJSIP symbian_ua(last realse 23-03-2009) and call disconnect  in Symbian_ua after 24.9 minutes in 5 tests.
  Second,  other test
callee and caller QJSimple(0.5) based PJSIP , also call disconnect in
QJSimple
not after const time usually  (2,10,12
minutes), callee stop send data and receive from caller, in some case
callee
crashed otherwise call continue but no any voice in caller or callee in
6 tests.
  
  Last, I try other test two X-lite client call
continued up
to one hour.
  Is found any person meeting same problem in call
up to 30
minutes in PJSIP?
  Thanks.



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