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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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Holger Hans Peter Freyther 211ad859de sip/call/mncc: Move source/dest into the call structure 
In preparation of a better show calls VTY command it is of interest
to know which number has been dialed by whom. For that store the
source/dest in there.

MNCC: Change the talloc root context to the call and don't try to
free the strings after calling the routing code

SIP: Use talloc_strdup to duplicate them.

Call: Add null check because the talloc_strdup of the SIP layer
could have failed.
 		2016-04-04 19:52:41 +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 sip/call/mncc: Move source/dest into the call structure 2016-04-04 19:52:41 +02:00
.gitignore Initial commit for a MNCC to SIP gateway (and maybe auth GW too) 2016-03-21 09:54:37 +01:00
COPYING Initial commit for a MNCC to SIP gateway (and maybe auth GW too) 2016-03-21 09:54:37 +01:00
Makefile.am Initial commit for a MNCC to SIP gateway (and maybe auth GW too) 2016-03-21 09:54:37 +01:00
README.asciidoc Write down some of the limitations of the current setup 2016-03-26 16:31:00 +01:00
configure.ac Initial commit for a MNCC to SIP gateway (and maybe auth GW too) 2016-03-21 09:54:37 +01:00
git-version-gen Initial commit for a MNCC to SIP gateway (and maybe auth GW too) 2016-03-21 09:54:37 +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.