2002 October

October 2002


In order for IP-telephony to become the primary voice line for households, certain requirements need to be fulfilled. These are called ETS, an authorized emergency telecommunication service. A new IETF draft outlines these requirements for IP-telephony
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A new IETF draft
outlines header compression for voice/audio over IP.

VoIP typically uses the encapsulation voice/RTP/UDP/IP, wherein the packet header is at least 40 bytes, while the voice payload is typically no more than 30 bytes. VoIP header compression can significantly reduce the VoIP overhead through various compression mechanisms. This is important on access links where bandwidth is scarce, and can be important on backbone facilities, especially where costs are high (e.g., some global cross-sections). In this draft we propose to re-use the methods in cRTP to determine the header compression context and to use the cRTP session context ID to route a compressed packet between the ingress and egress routers.

*asterisk.org* :: Let There Be Telephony
Asterisk is a complete PBX in software. It runs on Linux and provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in three protocols, and can interoperate with almost all standards-based telephony equipment using comparitively inexpensive hardware.

Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queing. It has support for three-way calling, caller ID services, ADSI, SIP and H.323 (as both client and gateway). Check the \’Features section for a more complete list.

The GNU oSIP library is written in C and get no dependencies except the standard C library. oSIP is thread safe and will generally be used in a multi-threaded application. Nevertheless, this is optional.

oSIP is little in size and code and thus could be use to implement IP soft-phone as well as embedded SIP software. oSIP is not limited to endpoint agents, and can also be used to implement \”SIP proxy\”.

oSIP does not intend to provide a high layer API for controlling \”SIP Session\” at this step. Instead, it currently provides an API for the SIP message parser, SDP message parser, and library to handle \”SIP transactions\” as defined by the SIP document.

Another Open Source SIP Software:

BerliOS Developer: Project Info – SIP swiss army knife

SIP swiss army knife – Summary Summary Homepage Bug Tracking CVS Code Repository Downloads

sipsak is a command line tool which can send simple requests to a SIP server. It can run additional tests on a SIP server which are usefull for admins and developers of SIP enviroments.