Commit Graph

4 Commits

Author SHA1 Message Date
Andreas Eversberg d928442c51 Forward DTMF as message directly from GSM BS to SIP.
In case rtp-bridge is used, tones cannot be generated. Instead,
a message is forwarded to SIP endpoint, so it generates it itself.
2012-01-15 10:51:58 +01:00
Andreas Eversberg 5463e1b62a Added bridgin support for GSM and SIP
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.
2012-01-15 09:42:35 +01:00
Andreas Eversberg 877a2dfd52 Adding bridge between protocol handlers (ports)
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.
2012-01-14 18:36:26 +01:00
Andreas Eversberg 863bc64219 Adding basic SIP support, using Sofia-SIP stack
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.
2012-01-13 06:24:21 +01:00