HAHA!!!
More very very interesting news!!!
I just wrote a Siren encoder/decoder/RTP payloader/RTP depayloader
for gstreamer and guess what! it worked! And when I say it
worked, I'm not just talking about the fact that I was able
to send and receive Siren (less than 1.6KB/s of bandwidth)
but I'm also talking about the fact that now, WLM *ACCEPTS*
our audio!!! it plays it!!! So.. our initial problem was not
that it needed ICE authentification on the socket, but it
actually just didn't play PCMA/PCMU (although it could send
it).. it only played Siren codec!
Instructions are the same as before... (but you'll need my
siren gst elements... I'll soon commit that into
gst-plugins-farsight).
But here's the new TODO :
TODO :
1 - clean the ugly hacky functions I used for testing...
2 - a proper configure/makefile/runtime dependency check...
3 - a proper UI
I'm about to clean those functions right now before I go to
bed, which means we'll just have the makefile stuff and the
UI to take care of...)
I'll fix the makefiles tomorrow and I'll leave the UI for
you guys! :p
KaKaRoTo
On Tue, Apr 08, 2008 at 08:35:46PM -0400, Youness Alaoui wrote:
> On Tue, Apr 08, 2008 at 08:33:08PM -0400, Youness Alaoui wrote:
> > Hi,
> > A bit more news.. I have just tested out the Audio stuff
> > between my aMSN and Tom's aMSN and even though we're both
> > firewalled (and obviously not in the same network) it
> > worked! I could hear him very well..
> > It is currently using PCMA/PCMU (almost uncompressed audio)
> > which uses about 10KB/s of your bandwidth.. I'll soon try to
> > write siren gst elements so we should get less than 1.6KB/s
> > of bandwidth with that codec...
> >
> > With my latest commit (r9707) things have changed a little..
> > here's how you can do it :
> > 1 - sending an invite :
> > open the console and type : createSip; inviteSip $email
> > 2 - receiving an invite :
> > open the console, you should see the invite printed in
> > there, like this :
> > Received INVITE : 6935BADC0FC1B544C33B62B465969E09 - {{{}..
> > (removed the rest of the line).. all you have to do is type :
> > acceptSIP 6935BADC0FC1B544C33B62B465969E09
> > and that's it... (notice that the 'id' 6935BADC0FC1B544C33B62B465969E09 was
> > taken from the 'Received INVITE' line...)
> >
> > Notice though that to be able to do this you will first need
> > this :
> > 1 - gstreamer
> > 2 - gst-plugins-base
> > 3 - gst-plugins-good
> > 4 - gst-plugins-bad
> > 5 - gst-plugins-farsight
> > 6 - farsight2 with ./configure --disable-python
> > --with-transmitter-plugins=rawudp
> > 7 - go to amsn/utils/farsight and type 'make'
> > Make sure you have the latest version of all these
> > components, you can try disabling some stuff if you don't
> > need them (like for farsight2)
> >
> >
> > This will work bidirectional between two aMSNs but with WLM
> > you can only receive and not send.. That should be fixed
> > soon..
> >
> > TODO :
> > 1 - use of ICE for having WLM receive our stream
> > 2 - add a Siren encoder/decode/payloader/depayloader for gstreamer
> > 3 - clean the ugly hacky functions I used for testing...
> > 4 - a proper configure/makefile/runtime dependency check...
> > 5 - a proper UI
> >
> > That's it! have fun!
> >
> > KaKaRoTo
> >
>
> Oh sorry, forgot something... you first need to do 'source
> sip.tcl'.. so the instructions become :
>
> First, you must open the console and type 'source sip.tcl' then ...
> 1 - sending an invite :
> type : createSip; inviteSip $email
> 2 - receiving an invite :
> One you receive the invite, you should see the invite
> printed in there, like this :
> Received INVITE : 6935BADC0FC1B544C33B62B465969E09 - {{{}..(removed the rest
> of the line).. all you have to do is type :
> acceptSIP 6935BADC0FC1B544C33B62B465969E09
> and that's it...
>
> KaKaRoTo
>
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