MNCC<->SIP bridge; attaches to OsmoMSC to interface with external SIP VoIP telephony
https://osmocom.org/projects/osmo-sip-conector
Keith Whyte
5f73c2033b
SIP end points can send periodic re-INVITES. Previous to this commit, the osmo-sip-connector would send a new call SETUP to the MSC for each re-INVITE. Add a function to find if we already handle this call based on the nua handle. Use this function to detect and respond with an ACK to re-INVITES. Add a function to extract the media mode from the SDP. In the case the re-INVITE has a=sendonly (HOLD) respond with a=recvonly In the case that the re-INVITE changes the media connection ip/port, forward this to the MNCC side with an MNCC_RTP_CONNECT Change-Id: I4083ed50d0cf1b302b80354fe0c2b73fc6e14fed |
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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.