doc: Expand the virtually empty user manual with some basics

Change-Id: Id42904a183b045eefac15a94139221a3bc65ecdd
Harald Welte 2023-03-21 20:00:04 +01:00
parent a935cbec7c
commit 95312c53c2
1 changed files with 21 additions and 0 deletions

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@ -16,6 +16,27 @@ has the following interfaces:
- SIP towards the PBX
- The Osmocom typical telnet VTY interface.
The SIP implemented by osmo-sip-connector can be characterized as follows:
Only a SIP trunk is supported; it will appear to the remote SIP server (PBX) like
another PBX (or a public network) interfaced via a trunk. Specifically, this means
there is no SIP REGISTER or any form of authentication supported. You
will need to configure the SIP peer to implicitly authorize the trunk by
its IP address / port.
osmo-sip-connector handles only the signaling translation between GSM CC
and SIP, but does not handle RTP. The RTP user plane is passed
transparently from the MSC-colocated osmo-mgw to the SIP side. This also
means that no transcoding is performed. The RTP streams contain whatever
cellular specific codec you have configured your network to use for this
call (FR, EFR, HR, AMR). Hence, **the SIP peer must support the
codec[s] you have configured on your MSC/BSC**
As the osmo-sip-connector attaches to the external MNCC socket of
OsmoMSC, running osmo-sip-connector will disable the internal call
routing of OsmoMSC, see the related OsmoMSC documentation. All mobile
originated calls originating in GSM will be passed to the SIP connector.
Find the OsmoSIPConnector issue tracker and wiki online at