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Re: [Linphone-users] libosip


From: Simon MORLAT
Subject: Re: [Linphone-users] libosip
Date: Fri, 07 Nov 2003 09:51:58 +0100
User-agent: Mozilla/5.0 (X11; U; Linux i686; en-US; rv:1.5) Gecko/20031030 Thunderbird/0.3

Hello,

I'm not an expert with VPN, but I guess it may have a bad influence on the network latency. Linphone-0.12.0 is now known to have numerous bugs when using the speex codec ( which is usually the default ). The "voice quality degraded" may be a symptom of those bugs. First you should then force both linphones to use other codecs: GSM Mulaw and Alaw are well supported by linphone. Otherwise you 'll need to try linphone-0.12.1pre which is downloadable in the unstable/ directory. All those bugs have been fixed in it. For the segfault, I would suggest that you try running linphone within gdb, and send me a gdb report ("bt full" command in gdb). 0.12.1pre fixes a segmentation fault that sometimes happen after a couple of seconds after the beginning of the call. 0.12.1pre also fixes an abort() that produces in alsalib (the effects are nearly the same as the segfault, you get the gnome crash window). So if you are using alsa drivers with the alsa card selected in the sound selection widget (not /dev/dsp), you should try 0.12.1pre. Note that in 0.12.1pre it is highly recommended, when you have alsa-drivers, to select the alsa card instead of /dev/dsp (which is the oss-emulation layer provided by alsa, that I have seen recently that it sometimes works bad).
I hope it will help you, don't hesitate to come back to me.
Simon



Ged Haywood wrote:

Hello again,

I have sites in France and England linked by the VPN.  I have set up
linphone over the link but it doesn't seem to be working properly.
The VPN is provided by OpenVPN and it seems to be very robust.  The
compiler at both sites is gcc 3.2.3 with glibc 2.3.1, kernel 2.4.22.

From each site I can run linphone to connect to sipomatic running at
the other site, although if the contact is repeated a second time, the
robot's voice quality seems to be degraded until sipomatic is stopped
and restarted.  The linphone connection appears to terminate normally
after each call.

Using linphone at one site I can make contact with linphone running at
the other site.  The dial and ring tones are fine, and I can answer
the call, and speak a few words which are reproduced clearly at the
*French* site, although there is a considerable delay (seconds) in the
voice message.  The user at the French site sees a number of messages
about RTP packets being rejected during this time.

Unfortunately then linphone segfaults at the English site.  It is
always this site which suffers the segfault.  No sound is reproduced
except the tone announcing the incoming call.  The tone is itself
clearly reproduced and stops as it should when the call is answered.

The computer at the French site is a PII 500, Crystal 3DFX sound card.
Linphone 0.12; libosip 0.9.7; oss sound drivers.  Unfortunately the
Alsa drivers do not support this card.  At this site linphone does not
segfault.

At the English site is an Athlon 1300, on-board (VIA8233) sound.  It's
all the same software, but I have also tried the Alsa sound drivers,
both versions 0.9.6 and 0.9.8, with exactly the same results, and
kernel 2.6.0-test5 again with the same results.  Linphonec also
segfaults.

Where should I start looking?

73,
Ged.









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