From 95312c53c2b96285c2ea7fbd9b72ad932b41a00d Mon Sep 17 00:00:00 2001 From: Harald Welte Date: Tue, 21 Mar 2023 20:00:04 +0100 Subject: [PATCH] doc: Expand the virtually empty user manual with some basics Change-Id: Id42904a183b045eefac15a94139221a3bc65ecdd --- doc/manuals/chapters/overview.adoc | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/doc/manuals/chapters/overview.adoc b/doc/manuals/chapters/overview.adoc index c65e5ad..a06b410 100644 --- a/doc/manuals/chapters/overview.adoc +++ b/doc/manuals/chapters/overview.adoc @@ -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 - https://osmocom.org/projects/osmo-sip-connector