MNCC<->SIP bridge; attaches to OsmoMSC to interface with external SIP VoIP telephony
https://osmocom.org/projects/osmo-sip-conector
Keith Whyte
6d398fea1d
GSM 04.08 10.5.4.11 (Table 10.85) states: Coding standards other than the standard defined for the GSM PLMNS shall not be used if the cause can be represented with the GSM standardized coding. This patch adds cause coding GSM PLMS (3) and sets cause location to "public network serving the local user" (2) This prevents UE that pay attention to this from treating all call termination as an error and paves the way to adding correct cause mapping from Table 10.86 Also replaces use of magic numbers with enum constants. Change-Id: I5d3fe3f0c9e8de26dd0c73b10b7e4fc63dff3952 |
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contrib | ||
debian | ||
doc/examples | ||
src | ||
tests | ||
.gitignore | ||
.gitreview | ||
COPYING | ||
Makefile.am | ||
README.asciidoc | ||
configure.ac | ||
git-version-gen | ||
osmoappdesc.py |
README.asciidoc
Osmo SIP Connector ================== Simple utility to map MNCC to SIP and SIP to MNCC. The VTY interface can be used to make configurations. The code doesn't have any RTP or transcoding support. Call identities can be either the MSISDN or the IMSI of the subscriber. Requirements of Equipment ^^^^^^^^^^^^^^^^^^^^^^^^^ * DTMF need to be sent using SIP INFO messages. DTMF in RTP is not supported. * BTS+PBX and SIP connector+PBX must be in the same network (UDP must be able to flow directly between these elements) * No handover support. * IP based BTS (e.g. Sysmocom sysmoBTS but no Siemens BS11) * No emergency calls Limitations ^^^^^^^^^^^ * PT of RTP needs to match the one used by the BTS. E.g. AMR needs to use the same PT as the BTS. This is because rtp_payload2 is not yet supported by the osmo-bts software. * AMR SDP file doesn't include the mode-set params and allowed codec modes. This needs to be configured in some way.