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. _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx&mkt=en-us -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20090419/cf47a693/attachment-0001.html>
