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Table of Contents
Frequently Asked Questions
General
Kamailio is a SIP server, implementing the specifications from RFC3261. Its root functionality is routing any kind of SIP packets. On top of that, many conceptual features are implemented, see more at:
No, Kamailio is a flexible SIP proxy. Many people integrate Asterisk, FreeSWITCH, SEMS, or other products with Kamailio for a B2BUA.
At network layer, Kamailio supports both IPv4 and IPv6.
At transport layer, Kamailio support UDP, TCP, TLS and SCTP. Transmission can be done in asynchronous mode (configuration option), inclusive for TCP and TLS.
Kamailio can be used to bridge between any combinations of these protocols, e.g., receiving SIP packets on UDP over IPv4 and sending out on TLS over IPv6.
No, you must restart after you update the configuration file.
But note that many global parameters can be changed via RPC/MI commands without restart (e.g., TCP connecting timeout, debug level). Applying changes related to loaded modules or routing block require always a restart.
Kamailio is an open source application licensed under GNU Public License version 2 (aka GPLv2). It can be used for free “as in beer”.
Starting with end of 2008, contributions to core and several modules are done under BSD license. That means parts of it can be extracted and used under BSD license terms. But over all, when used as one application, the use must follow the terms of GPLv2, because GPLv2 is viral.
Modules
A module is an extension that compiles in a separate object file and can be loaded at Kamailio startup. Usually a module exports new functions that can be used in configuration file routing blocks.
Media Streams
No, however Kamailio can be configured to proxy media if needed.
Kamailio supports all codecs (even codecs that haven't been created yet).
Since Kamailio is a SIP proxy, it does not handle the media streams. Codecs are negotiated between the two endpoints.
Yes, Kamailio can be used for video calls. The two SIP phones must support video codecs/calling.
