The dependency on mISDN (loopback interface) is completely removed
from GSM and SIP interfaces.
The built in bridge of LCR now forwards audio data between these
interface instances or between these instances and other instances.
Additionally both GSM BS and SIP support direct forwarding of RTP
traffic between other SIP endpoint and OpenBSC, so no traffic is
forwarded by the LCR itself. This is done by forwarding RTP peer
informations between these interface instances.
This is required to bridge traffic beween non-mISDN handlers,
such as GSM, SIP and voice box. Also it bridges traffic between
mISDN handlers and non-mISDN handlers. It is the fundamental step
to get rid of mISDN (loop interface) for non-mISDN handlers.
This is required to bridge audio e.g. between SIP and GSM without
using mISDN. There will be no limitations on 'b-channels' anymore.
Still GSM and SIP requires mISDN, but this will be changed later.
With that bridge I cleaned up some code and also removed the
MESSAGE_DATA, which is not required anymore.
This support is just a simple peer-to-peer support for basic calls.
Currently it requires mISDN_l1loop interface, as every non-ISDN
interface does. Later it will be possible to use it without.
chan_lcr can be handled as an interface. This way it is possible to (e.g.):
- make a SIP phone become an LCR extension with all LCR features.
- make conference calls. (untested)
- perform parallel ringing. (ISDN phone and SIP phones can ring in
parallel.)
- do voice recoding.
It is still also possible to link chan_lcr directly without interface
(as before).
Documentation/howto for that will follow.
trace of vbox
better beep after announcement
announcements without beep
recording of answering machine (vbox) works, as well as call recording.
modified: README
modified: action_vbox.cpp
modified: port.cpp
modified: port.h
modified: todo.txt
modified: vbox.cpp