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Re: oSIP status
Re: oSIP status
Wed, 6 Mar 2002 17:54:28 +0100 (CET)
This message is about a new GNU package: the GNU oSIP library.
You'll find the latest release (0.8.1) in
oSIP is a one year old project that implements a SIP stack in C.
SIP is the Session Initiation Protocol. It is a signalling
protocol for initiating, modifying and terminating mutlimedia
sessions (mainly for IP based sessions). SIP is a light and
simple replacement for H323. SIP not only provides basic
telephony services; it can be used to build services provided
through signalling. Presence and instant messaging are part
of the extra features you'll get on any next generation phones.
There are many features to talk about, but the most known are name
mapping and user mobility:
Wherever you are on earth, whatever phone you are using, you'll
be reached with the following "phone number":
As you may notice, a SIP-URL is very similar to a mailto url. This
makes it easy to use and implement. Like in the mail, you'll get
a "To" header showing the callee, a "From" header showing the caller
and a "Subject" header... SIP uses proxy to route calls through
networks. RTP will be used for most sessions.
SIP is being adopted by the majority of vendors nowdays. H323 is heavy
and has many interoperability issues.
SIP is not currently deployed, but it's coming fast and there is a need
of free software based on SIP. Currently, deployement costs a lot and
this must change: Have you ever bought a MAIL reader? :-)
You can test very basic functionality with a GTK sip phone built on
oSIP. It is not yet fully interoperable but currently works fine
for linphone to linphone calls. If you don't want to exchange
IP addresses, you'll need to configure linphone to use a registrar.
"osrd" (at very early stage) provides this. Both "linphone" and "osrd"
are available through freshmeat.net.
Hope you'll enjoy the SIP protocol and the oSIP stack.
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