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 5319d4d979 coverity: Address issue found by coverity
Add NULL checks on the return value of call_leg_other() in
update_rtp()

If the remote side has requested media change and we cannot
find the other leg, then release call. This should not
happen.

Also, Add an assert to show that we cannot be here
without call type of SIP or MNCC (not related to coverity)

Fixes: CID#202863
Change-Id: I6f1f26533a25c93f243090bc02f1bc83b9108d42
2019-08-09 14:12:32 +02:00
contrib contrib/jenkins.sh: run "make maintainer-clean" 2019-07-10 13:25:00 +02:00
debian Bump version: 1.2.0.25-ff8a → 1.3.0 2019-08-08 17:54:49 +02:00
doc debian: create -doc subpackage with pdf manuals 2019-05-29 12:14:22 +02:00
src coverity: Address issue found by coverity 2019-08-09 14:12:32 +02:00
tests distcheck/tests: Add the referenced osmoappdesc.py for testing 2016-04-24 22:28:35 +02:00
.gitignore build manuals moved here from osmo-gsm-manuals.git 2018-11-27 18:34:56 +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
Makefile.am Fix DISTCHECK_CONFIGURE_FLAGS override 2018-12-04 15:32:11 +01:00
README.asciidoc Write down some of the limitations of the current setup 2016-03-26 16:31:00 +01:00
configure.ac Require newer libosmocore 1.0.0 2019-08-08 17:54:11 +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.