fixup sip
SIP: Register, STUN and authentication support... - 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
Make tones-dir option available for all interface (interface.conf)
GSM: Fixes to GSM interface (multiple networks) * 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
Add essential option to enable and prefer half rate calls to mobile Without it might not be possible to use TCH/H, unless OpenBSC would support late assignment.
Added option to change DTMF decoding threshold level If not given, the DSP modules' default value is used, rather than setting it to 0. This was a bug.
Add FXS support This requires FXS support to mISDN too.
Removed complete bchannel handling from chan_lcr 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.
Allow setting IP:port for peers of SIP interfaces.
Adding negotiation of speech codecs between GSM and SIP when using rtp-bridge 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
Adding switch to compile LCR without mISDN support Disable: --without-misdn Enable: --with-misdn Otherwise it will be enable automatically, if mISDN user is installed.
Fixed dead pointer problem when handling interfaces In order to get the pointer to the currently existing interface, a new function is used, to resolve interface by name.
Adding simple bridge application to forward calls without PBX app. 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.
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.
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.
Adding shutdown option to interface.conf This way an interface can be disabled by just one keyword and not by uncommenting all lines of it.
[gsm] Make LCR work with current Osmocom-BB. Osmocom-BB is still developed, and this only works with the jolly/voice branch. Audio is not yet transmitted, so it is not quite usefull yet.
Adding interface support for remote app (chan_lcr). 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.
Splitted GSM support into BS (network) and MS (mobile) part.
fix last remnant of "extern" vs. incorrect "external" confusion, correct spelling Patch against current git. Signed-off-by: Andreas Mohr <andi@lisas.de> Signed-off-by: Andreas Eversberg <jolly@eversberg.eu>