MNCC<->SIP bridge; attaches to OsmoMSC to interface with external SIP VoIP telephony
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Vadim Yanitskiy bd66804082 mncc: rework passing GCR over the MNCC interface
Using *unpacked* 'struct osmo_gcr_parsed' in the MNCC PDUs makes
the protocol even more complicated than it currently is, and
moreover complicates implementing MNCCv8 in the ttcn3-sip-test.

Replace 'struct osmo_gcr_parsed' in 'struct gsm_mncc' with a
fixed-length buffer, which is supposed to hold the Global Call
Reference encoded as per 3GPP TS 29.205.

Check / indicate presence of GCR using the MNCC_F_GCR flag.

Change-Id: Iaff46732948f8f5d03e42f17c35cbac8a80af49b
Fixes: Id40d7e0fed9356f801b3627c118150055e7232b1
Related: OS#5164, OS#5282
2021-10-27 17:03:42 +03:00
contrib Bump version: → 1.5.0 2021-02-23 13:42:08 +01:00
debian debian/control: remove dh-systemd build-depend 2021-09-01 16:07:07 +02:00
doc debian: create -doc subpackage with pdf manuals 2019-05-29 12:14:22 +02:00
src mncc: rework passing GCR over the MNCC interface 2021-10-27 17:03:42 +03:00
tests distcheck/tests: Add the referenced for testing 2016-04-24 22:28:35 +02:00
.gitignore .gitignore: Get rid of new autofoo tmp files 2021-02-02 16:42:12 +01:00
.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 EXTRA_DIST: debian, contrib/* 2020-05-22 13:46:54 +02:00
README.asciidoc Write down some of the limitations of the current setup 2016-03-26 16:31:00 +01:00 Bump version: → 1.5.0 2021-02-23 13:42:08 +01:00
git-version-gen Fix git-version-gen 2017-10-28 18:20:00 +02:00 switch to python 3 2019-12-11 09:43:11 +01:00


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

* 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


* 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.