Voice and Video GSoC Project

[Diego Costantini]

I tested a bit with debug option and a segmentation fault happened after a
long ring eventually accepted. (I attached logs of both clients)
A lot of crashes (especially the Hangup one) today are not happening...

I also have a question. I noticed that voice is not received immediately. I
mean that I tell something (not received), then in the debug window appears:
- media: connecting pad: success
and after that, audio is correctly delivered.

I know that this happens sometimes with media gateways in SIP even though I
don't understand why, since the session should be already negotiated in
advance.
Why here?

Thanks,
Diego

-----Original Message-----
From: Maiku [mailto:cmaiku at gmail.com] 
Sent: Dienstag, 3. Juni 2008 18:50
To: Diego Costantini
Cc: devel at pidgin.im
Subject: Re: Voice and Video GSoC Project

On Tue, Jun 3, 2008 at 8:38 AM, Diego Costantini
<diego.costantini at gmail.com> wrote:
> I apologise, I used the gstreamer0.10-plugins-farsight from ubuntu, not
from
> sid.
>
> I noticed that if the libraries are not exactly those you mentioned, it
> probably won't work. For example, I was using the experimental repository,
> so newer libraries, then one of the clients got in the chat window:
> Error creating session.
> You have rejected the call.
>
> This happens only on the client on that machine, also when it is the call
> initiator. When the other initiates the call, he waits on Calling... while
> the other already rejected.
>
> Sometimes they also crashed, but I couldn't reproduce it.
>
> Anyway I made another clean install and now they look like calling each
> other.
> - When they manage to start the call, they get stuck when I press Hangup
and
> I have to force quit Pidgin.
> - When I call, one client rejects the call, if I call again but this time
I
> accept, the same client crashes.
>
> Since I am working on virtual machines, I have audio card configured but
no
> real output to check how it works, and I noticed with wireshark that
during
> calls they are exchanging rtp packets every 2 or 3 seconds, both at ports
> 7079. This doesn't resemble the tons of RTP packet I was used to see with
> SIP.
>
> Of course these are not complains but bugs I found until now and you don't
> need to search for, just fix them when you'll have time :)
>
> Anyway, the taste of calling with Pidgin is already very sweet, keep up
the
> great work ;)

If you check the debug log, either running pidgin with the -d switch
or going to Help->Debug Window, when you get the "Error creating
session" message, it should say in the debug output something like
"media: Error creating session:" and then the reason why. I'd be
curious to see what the reason is.

As far as testing the sound, there are two progress bar widgets to the
right of the hangup button which display the sound level. The one on
the left is the local audio level going out to the other client and
the right being the input from the other client, which would normally
be played through your speakers. If you could maybe switch your audio
input to the sound the OS is making, and then have the OS make sounds,
you could hopefully see those bars moving. I don't know if your
virtual machine/OS would support such a thing, but it's worth a shot.

Good luck, and thanks for testing. :)
-Maiku
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