This is a module for handling SIP, the IETF standard for VOIP
It is written completely in perl.

With the help of this module you can write SIP endpoints (e.g
phones, answer machines), SIP proxies and registrars.
It contains no GUI and no real code for working with video or
audio, but has some support for RTP (no RTCP) and working
with PCMU/8000 data, enough for sending PCMU/8000 encoded
audio to a SIP peer and for receiving and saving PCMU/8000
audio data.

The module is designed to be completely asynchronous, e.g. you
either integrate it in your own event handling or you can use
the simple event handling which is included.

It was tested on Linux (Ubuntu 6.10,7.04,7.10), MacOSX 10.3+10.4,
OpenBSD3.9+4.1 with various perl versions starting with
perl5.8.7, including 5.10

Sample Code was tested with Snom 300 Phones, Asterisk 1.2,
Fritz!Box and KPhone.

See TODO for a list what still need to be done and BUGS for
known bugs.

See THANKS for contributors, bug reporters and sponsors.

See samples/ for small examples.
See bin/ for usable applications.