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 6d398fea1d Add Cause to DISCONNECT and RELEASE Requests
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
2018-05-28 15:20:26 +00:00
contrib contrib: jenkins.sh: Disable doxygen in libosmocore build 2018-02-20 20:23:10 +01:00
debian Bump version: 1.1.0.11-11a0-dirty → 1.1.1 2018-05-06 17:45:55 +02:00
doc/examples vty: Work on configuration of the MNCC to SIP gateway 2016-03-21 15:39:41 +01:00
src Add Cause to DISCONNECT and RELEASE Requests 2018-05-28 15:20:26 +00:00
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.