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osmo-sip-connector
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MNCC<->SIP bridge; attaches to OsmoMSC to interface with external SIP VoIP telephony https://osmocom.org/projects/osmo-sip-conector
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Alexander Couzens f678caa819 Revert "sip: Specify invite contact tag" 
This reverts commit 52b2afce2c.
The contact header is generated by the original sofia-sip library.
By adding the contact header explicit as user header it violates the
SIP RFC because sofia will add the Contact header to the BYE message as
well.

Let's fix the bugs in the freeswitch sofia-sip and make it compatible
(not bug compatible) with the original sofia-sip.

Change-Id: I712f17fecbc372d1e486e80673a548e281b37800
 		2020-09-05 23:13:55 +02:00
contrib contrib: integrate RPM spec 2020-05-19 15:49:42 +02:00
debian debian/control: change maintainer to the Osmocom team / mailing list 2020-08-13 14:59:51 +00:00
doc debian: create -doc subpackage with pdf manuals 2019-05-29 12:14:22 +02:00
src Revert "sip: Specify invite contact tag" 2020-09-05 23:13:55 +02:00
tests distcheck/tests: Add the referenced osmoappdesc.py for testing 2016-04-24 22:28:35 +02:00
.gitignore contrib: integrate RPM spec 2020-05-19 15:49:42 +02:00
.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 Makefile.am: EXTRA_DIST: debian, contrib/*.spec.in 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
configure.ac contrib: integrate RPM spec 2020-05-19 15:49:42 +02:00
git-version-gen Fix git-version-gen 2017-10-28 18:20:00 +02:00
osmoappdesc.py osmoappdesc.py: switch to python 3 2019-12-11 09:43:11 +01:00

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.