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 b603272787 sdp.c Send octet-align in fmtp
rfc4867 8.2:

      octet-align: Permissible values are 0 and 1.  If 1, octet-aligned
               operation SHALL be used.  If 0 or if not present,
               bandwidth-efficient operation is employed.

We don't have any support for AMR BE mode, but if we don't
send this the other end expects BE mode and can't decode the stream

Change-Id: I938758ac4ec55db9223e3da6c3c277e8fa670055
2017-09-03 18:30:53 +00:00
contrib Fix build after recent libosmocore update 2017-09-01 10:19:18 +02:00
debian debian: Add dh-autoreconf required by the debian packaging 2016-03-31 20:04:54 +02:00
doc/examples vty: Work on configuration of the MNCC to SIP gateway 2016-03-21 15:39:41 +01:00
src sdp.c Send octet-align in fmtp 2017-09-03 18:30:53 +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
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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
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git-version-gen Initial commit for a MNCC to SIP gateway (and maybe auth GW too) 2016-03-21 09:54:37 +01:00
osmoappdesc.py distcheck/tests: Add the referenced osmoappdesc.py for testing 2016-04-24 22:28:35 +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.