Re: [Linphone-developers] [Q] Bug of linphone-1.0.1?

[Clement Chen]

Hi, Simon,
    The attached is the patch for linphone-1.0.1 to provide file
stream playing features FYI. During a call, a file stream or voice
stream can be switched dynamically. (DTMF strings will not be played
locally when the file stream is used) When a file stream ends playing,
a call back function can be triggered to report this information.
Currently I only register one for linphonec. The GUI version of
linphone may use another call back function instead, since it may show
that information in a GUI component.
    I didn't seem to modify codes relating to SDP processing... and
this error also occured when the original linphone-1.0.1 build is
used. My OS is fedora core 3, libosip is libosip2-2.2.0, and speex
1.1.7. I used "--enable-alsa --with-speex=/usr/local" options to
configure the make file.

Information for other libraries:
jack (from its CVS repository)
libsamplerate-0.1.2-1
libsamplerate-devel-0.1.2-1
libsndfile-1.0.11-1.1.fc3
libsndfile-devel-1.0.11-1.1.fc3

    To use the attached patch, here are the steps:
1. gunzip linphone-1.0.1.patch.gz
2. patch -d <parent dir. of your linphone-1.0.1 original source> -p0 <
./linphone-1.0.1.patch
    for example, if your source is in /usr/local/src/linphone-1.0.1,
you should run
    patch -d /usr/local/src -p0 < ./linphone-1.0.1.patch

Thanks a lot! :)

Clement.

On 5/20/05, Simon Morlat <address@hidden> wrote:
> Hello,
> 
> For the rtp errors, the root cause is here:
> 
> |WARNING| <sdphandler.c: 494> Could not parse remote offer.
> 
> So the remote rtp port could be read, thus linphone could not set destination
> address for rtp packets. I don't know why there is this error; what version
> of osip are you using ? Did you modify some code around SDP processing ?
> 
> Could you please send a patch for the modifications you've made ? I'll see if
> I can integrate it into the main tree.
> 
> Simon
> 
> Le Jeudi 19 Mai 2005 14:12, Clement Chen a écrit:
> > Hi,
> >    I used linphone-1.0.1 to test the following scenario and found
> > oRTP errors in the debugging messages. Should I file a bug against it?
> >
> > linphone #A : sip:address@hidden:5060
> > linphone #B : sip:address@hidden:5060
> >
> > Test step:
> > 0. Launch #A and #B with "-d 6 -l /tmp/test.log" options
> > 1. #A calls #B, #B answers, then #A terminates the call after seconds.
> > 2. #B calls #A, #A answers, then oRTP errors are reported in the log file.
> >
> > The attached "log_phoneB.txt" is a log for linphone #B showing the
> > errors FYI. (error message starts from line 402)
> >
> > Thanks a lot!
> >
> > Clement.
> >
> > BTW, for testing purposes, I myself made some code modifications with
> > both linphonec (linphone CLI) and sipomatic for linphone-1.0.1 to
> > provide the following features:
> >  - sipomatic :
> >    * '-f_16k' option : setup the 16000hz annouce file
> >    * '-e' option : play the annouce file "endlessly" until call is
> > terminated. - linphonec :
> >    * '[play]' section for the configuration file :
> >      = 'file_8000hz' option : setup the 8000hz annouce file
> >      = 'file_16000hz' option : setup the 16000hz annouce file
> >      = 'file_record' option : setup the record file path
> >    * '-f' option : play the annouce file after a call is established,
> > instead of using audio stream from the soundcard
> >    * '-e' option : play the annouce file "endlessly"
> >    * 'v' or 'voice' command : switch audio source to the soundcard
> >    * 'p' or 'play' command : switch audio source to the annouce file
> >    * 'e' or 'endless' command : enable/disable playing annouce file
> > endlessly * 's' or 'status' command : show rtp status for a switched stream
> > * 'rec_file <file2rec>' command : change the record file path
> >
> >    Note that only remote peer will hear the content of the annouce file.
> >
> > Please let me know if any of you are interested in them or think them
> > worth being committed into the repository. Thanks.
>

linphone-1.0.1.patch.gz
Description: GNU Zip compressed data