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
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