- Register works in both ways
- STUN works as client
- Authentication to remote endpoints only
- Early audio (183) works in both directions
- Caller ID works in both directions
Note: The implementation is only a small subset of many SIP features.
* Multiple network instances are now possible to attach multiple networks
* Early audio handling fixed
* Number type can be given from base station (setup / setup confirm)
* Equal callref for different GSM-MS instances are handled correctly
The remote application interface does not allow any bchannel to be
exported or imported. Audio traffic via socket interface is used instead.
The joinremote instance became obsolete and is removed.
The remote action (routing) became obsolete, use interface.conf instead.
The handling of loopback device became obsolete and was removed
The chan_lcr does not rely on mISDN anymore, that means:
- can be used with GSM and without mISDN at all.
- chan_lcr can be used as internal extension of LCR (e.g. SIP phone)
(chan_lcr can be handled as any other interface)
- no loopback device to be used anymore.
Since LCR does not put hands on any RTP frame when directly bridged between
OpenBSC and SIP, it will now allow all speech codecs that are commonly supported
by MS and remote SIP endpoint.
It must be noted that OpenBSC must support forwarding the codec types that
MS and remote SIP endpoints support.
Currently LCR negotiates the following codecs for GSM:
- Full Rate
- EFR
- AMR
- Half Rate
Call received on an interface can directly be forwarded to a given
destination interface, instead of routing the call through PBX
application. This way calls can be forwarded without going through
route.conf.
Currently only SIP and GSM destinations are supported. Also there
are no tones generated, if one side provides no tones, but the
other wants to receive them.
The keyword "bridge <output interface>" in interface.conf is used.
Without that keyword, incomming calls are handled as usual.
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 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.
If no specific interface is given for the 'extern' rule in the
routing.conf, or if call is forwarded (settings), then only
interfaces marked with 'extern' flag are used.
You need to set this flag when upgrading to this version. See
default/interface.conf for hint.
modified: README
modified: apppbx.cpp
modified: default/interface.conf
modified: interface.c
modified: interface.h
Added limitation option for maximum dialed digits. If dial string exceeds
that limit, overlap-dialing is used to complete dial string.
Siemens EWSD (APS V16) only allows 20 digits at a time.
modified: README
modified: chan_lcr.c
modified: default/interface.conf
modified: dss1.cpp
modified: dss1.h
modified: ie.cpp
modified: interface.c
modified: interface.h
Andreas Eversberg
Andreas Eversberg