Re: [Linphone-users] linphonec: segmentation fault

2008-01-15 Thread Francois-Xavier KOWALSKI 

Simon,

Recent kernels do notify application segfaults on the kernel console 
(via a handler of the associated MMU processor trap). This is a way for 
administrators that something is going wrong on the machine. Otherwise 
the crash is only visible from the user's context  the disk may fill-up 
silently with core dumps.


The exact format in the syslog is kernel-dependent  hence differs from 
Debian  RHEL (the 2 OS's I have observed this message on).


There is a similar messagefor un-aligned memory access.

--FiX

Simon Morlat wrote:

Hi,

Can you please run the same test from gdb ?
I'm a but surprised with this segfault message coming from the kernel.

Simon

Le Wednesday 09 January 2008 18:21:46 Gerard Robin, vous avez écrit :
  

Hello,
linphonec craches (on cli):
my system:
laptop acer 5102wlmi (amd64)
OS: Debian Sid
linphone: version: 2.0.1-3

~$ linphonec
Friend EUGENIOhome sip:[EMAIL PROTECTED] is Gone
8-

Ready
Warning: video is disabled in linphonec.
linphonec caRegistration on sip:ekiga.net successful.
linphonec call 500
linphoneec[7498]: segfault at  rip 2af858f467b9 rsp
7fff526c8c30 error 4

~$ strace -c linphonec
--8--
Process 4299 detached
% time seconds  usecs/call callserrors syscall
-- --- --- - - 
   48.760.000470 470 1   execve
   24.900.000240   0   560   read
   12.550.000121   1   103   mmap
   10.370.000100   25146 access
3.420.33   0   291   gettimeofday
0.000.00   022   write
0.000.00   07937 open
0.000.00   052   close
0.000.00   0 5 2 stat
0.000.00   038   fstat
0.000.00   042   mprotect
0.000.00   0 4   munmap
0.000.00   0 3   brk
0.000.00   020   rt_sigaction
0.000.00   0 9   rt_sigprocmask
0.000.00   021   ioctl
0.000.00   0 2   pipe
0.000.00   0 1   select
0.000.00   011socket
0.000.00   010   connect
0.000.00   0 1   bind
0.000.00   011   getsockname
0.000.00   010   setsockopt
0.000.00   0 1   clone
0.000.00   0 2   fcntl
0.000.00   0 1   chmod
0.000.00   0 1   getrlimit
0.000.00   0 1   arch_prctl
0.000.00   015 2 futex
0.000.00   0 1   set_tid_address
0.000.00   0 1   set_robust_list
-- --- --- - - 
100.000.000964  137087 total

do other users of linphonec have the same issue ?

With xwindow linphone starts fine but I can't hear my correspondant and
him can't hear me neither.

Thanks for any advice.






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--
Francois-Xavier FiX KOWALSKI /_ __Tel:+33 (0)4 76 14 63 27
HP OpenCall Software  - EMTS  / //_/Fax:+33 (0)4 76 14 51 62
Media-Processing Engineering/  http://www.hp.com/go/opencall
  i n v e n t




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Re: [Linphone-users] New softphone demo from antisip!

2008-01-15 Thread Klaus Darilion 

Hi Aymeric and Asterisk users.

FYI: I've just tested the softphone with a Asterisk and a 3G video 
call(nokia 6630): In direction 3G--SIP the video is fine. In direction 
SIP-3G* the video is there, but bad quality (block artefacts ...).


regards
klaus

*make sure to set upload bandwidth to 64kbit



Aymeric Moizard schrieb:


Dear users,

Most of your already know that I've started a company named antisip 3 
years ago. Since that time, I have improved a lot my osip2 and eXosip2 
projects and have put a lot of effort in contributing to oRTP  the 
mediastreamer2 projects.


This helped me much for my commercial project named amsip which now 
offer a complete SIP sdk including presence, instant messaging, as well 
as a good set of audio  video codecs on all major platforms.


Among capabilities, you can find in amsip/eXosip2/mediastreamer2:
* amazing NAT traversal capabilities based on the ICE specification
  allowing many calls to be peer to peer even if both correspondant
  are behing a NAT!
* video conferencing
* performant H264 codec using intel primitives
* bandwidth negotiation to adapt video framerate  compression.

But text explanation are usually not enough! That's why I'm proud
to announce the first version of my own softphone.

http://sip.antisip.com/download/emansip-setup/emansip-setup-v411-rc10.exe

You have to create a new account on sip.antisip.com if you don't
already have any:

http://sip.antisip.com/

Once you have created an account, you'll receive a mail where you have
to confirm your account creation before you can use the service. If
you don't receive the mail, please ask me and I'll confirm myself
your account.

Current features for this softphone:
* calls
* encryption (TLS  SRTP)
* music on hold, mute, record converstation
* presence
* video calls (configure your upload/download bandwidth)
* audio conference
* version limited to speex/PCMU/PCMA/gsm
* version limited to H263-1998 video codec
* Try streaming files in conversation!!!
* Try video conference (configure your upload bandwidth to minimum value!)

I'm working on adding Instant Messaging: it will appear in a very
few days.

As it's the first official version of this softphone, it will certainly
contains a few bugs. I hope you'll be kind enough to report them to me!

It's time for me to wish you a happy new year and success for
all your software developments,

For any business, support related questions, you can call me at 
sip:[EMAIL PROTECTED]


tks to all,
Aymeric MOIZARD / ANTISIP
amsip - http://www.antisip.com
osip2 - http://www.osip.org
eXosip2 - http://savannah.nongnu.org/projects/exosip/



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