Because EFR/AMR/HR codecs use dynamic RTP payload types, it is essential
to forward the actual media types between endpoints too. These media
types are used for negotiation of codecs. A dynamic payload type is
used as given by remote peer. Locally generated payload types are used
when offering codecs to remote peer.
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
If a bridge is enabled, tones (e.g. hangup tone) will have priority
over the bridge. The bridge will continue to forward audio, after
tone is removed. (e.g after beeing on hold music)
In this case the frame is dropped, but audio of the last frame is repeated
with a reduced level. The level is reduced again an again until a new
valid frame is received. This way there is no silent gap in the audio
stream.
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.
There is no linking of any osmocomBB source code required. In order
to use osmocomBB or OpenBSC, just enable the interface, as described
in defaults/interface.conf. At osmocomBB/mobile or at OpenBSC, just
use the option "-m" to enable the socket interface.
Enable GSM at LCR with "./configure --with-gsm-ms --with-gsm-bs".
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.
gsm.cpp: In member function .void Pgsm::bchannel_send(unsigned int,
unsigned int, unsigned char*, int).:
gsm.cpp:249: error: overloaded function with no contextual type
information
make[1]: *** [gsm.o] Error 1
I hope that this will fix the compiler problem.
Fix of conference release bug.
Calls can now be forwarded during alerting phase via "*3#".
modified: README
modified: apppbx.cpp
modified: configure
modified: configure.ac
new file: default/openbsc.cfg
modified: dss1.cpp
modified: gsm.cpp
modified: joinpbx.cpp
No more patch is required, just link openbsc directory to LCR source directory and run "configure":
cd lcr
ln -s path_to_openbsc/openbsc .
modified: Makefile.am
modified: Makefile.in
deleted: bootstrap.c
deleted: bootstrap.h
modified: configure
modified: configure.ac
modified: default/gsm.conf
modified: gsm.cpp
modified: gsm.h
modified: gsm_conf.c
Andreas Eversberg
Harald Welte
Andreas Eversberg