GSM: Minor unused variable fix
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
GSM: Add audio frame type for uncompressed 16 bit frame It is usefull for connecting MNCC to other networks than GSM.
GSM: A breakdown of MNCC socket causes all calls to be released correctly
Experimental crypto feature: Support for libvootp
gsm: Implement the size checking of the hello packet
gsm: Verify the MNCC_VERSION of the BSC/MS and close the socket on mismatch The BSC/MS will send a Hello packet that includes the version number, make LCR verify this version number and close the socket in case it does not match a supported version.
Add support for TCH/H and half rate codec
AMR codec support
Add AMR codec, for supporting EFR transcoding The AMR codec is added, but at this point only EFR payload is supported.
Add GSM full rate codec to LCR's source repository There is no more need to download a seperate version of GSM full rate (06.10) codec anymore.
Fix: Allow recording of audio for SIP/remote/GSM interfaces too
Cleanup: Make interface name be part of Port class
Added support for all GSM codecs to GSM and SIP interface Untested!
Removed obsolete #include directive.
Allow dynamic RTP payload types when bridging between SIP and OpenBSC. 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.
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
GSM now receives tones during bridge 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)
Adding handling of bad GSM audio frames 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.
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.