MNCC<->SIP bridge; attaches to OsmoMSC to interface with external SIP VoIP telephony https://osmocom.org/projects/osmo-sip-conector
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Keith Whyte 503d8fdb64 Add SIP <-> MNCC Cause Map
Implements a cause_map, mapping the SIP status codes
to GSM48_CC_CAUSE_* that are defined in libosmocore.
The map at the same time implements the Q.850 cause texts
that are subsequently used in the SIP Reason: header.

Implements two functions cause2status() and status2cause()
to read the map and return the the mapped status.

The mapping mostly follows the implemention in the LCR program,
so that any implementation relying on that mapping should
continue to work as expected with osmo-sip-connector.

Change-Id: Id82be8603a30a6aec28fc0258236c1746973db58
2018-08-31 16:22:17 +02:00
contrib contrib: jenkins.sh: Disable doxygen in libosmocore build 2018-02-20 20:23:10 +01:00
debian debian/rules: Don't overwrite .tarball-version 2018-08-06 11:19:22 +02:00
doc/examples vty: Work on configuration of the MNCC to SIP gateway 2016-03-21 15:39:41 +01:00
src Add SIP <-> MNCC Cause Map 2018-08-31 16:22:17 +02:00
tests distcheck/tests: Add the referenced osmoappdesc.py for testing 2016-04-24 22:28:35 +02:00
.gitignore Use release helper from libosmocore 2017-08-26 05:56:55 +00:00
.gitreview Add git review config 2017-08-25 18:38:05 +02:00
COPYING Initial commit for a MNCC to SIP gateway (and maybe auth GW too) 2016-03-21 09:54:37 +01:00
Makefile.am Use release helper from libosmocore 2017-08-26 05:56:55 +00:00
README.asciidoc Write down some of the limitations of the current setup 2016-03-26 16:31:00 +01:00
configure.ac Logging: Log mncc_names in mncc_data() 2018-05-24 16:56:41 +02:00
git-version-gen Fix git-version-gen 2017-10-28 18:20:00 +02:00
osmoappdesc.py osmoappdesc: Fix VTY prompt to use OsmoSIPcon, not old OsmoMNCC 2018-03-27 15:51:02 +02:00

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.