MNCC<->SIP bridge; attaches to OsmoMSC to interface with external SIP VoIP telephony
https://osmocom.org/projects/osmo-sip-conector
Keith Whyte
ab5b4311b0
Do not send an MNCC_RTP_CONNECT as a result of a SIP re-INVITE, unless the media connection information has changed. Change-Id: I7c48300092a309e50a8fe091b30e395e7c72de9d |
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| contrib | ||
| debian | ||
| doc | ||
| src | ||
| tests | ||
| .gitignore | ||
| .gitreview | ||
| COPYING | ||
| Makefile.am | ||
| README.asciidoc | ||
| configure.ac | ||
| git-version-gen | ||
| osmoappdesc.py | ||
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.