Name

sipconf - SIP Audio Conference Server

Synopsis

sipconf

Availability

The code runs on Solaris 5.8, with other Unix platforms available upon request.

Description

sipconf is an SIP based audio conference bridge.

New Features

1. Add REDIRECT service in the conference servers.
2. Choose the least loaded server for the new "ad hoc" conference created via INVITE.
3. Choose the server has the maximum left capacity for the conference created via web interface.

Options

-p portnumber
Port number listening for SIP request, default is 5060.
-h
Print usage information and exit.
-v
Print version and exit.
-D sql://user:password@host:port/database
URL for the SQL database that stores user information. The port sipecification is optional. Th -D parameter is optional. If it is not present, sipconf will look fo ra file named cinema_db.conf in its directory. Additionally, on UNIX system it will try reading the file /etc/cinema_db.conf. This file should contain a database url of the form described before. Normally, these files are automatically created during installation by CINEMA setup programs and hence, they need not be modified. On Windows system, sipconf will also try reading for database uri under the registry key HKLM\Columbia University\CINEMA_DB.
-X
Run as a console application. If this parameter is not specified, sipconf will run as a deamon.
-i
Run in interactive mode. In this mode, sipconf can accept user commands through console input.
-m packetizatin_time
Specify a packetization_time in milliseconds between 20 and 2000 . The default is 20 ms.
-n
Use numeric IP addresses in Via headers instead of hostname.
-a
Accept all conferences. This is typically used for testing. If this option is enabled. sipconf doesn't require the conference information to be present in the database.
-b
Send back audio to participants. This is typically used for testing. sipconf will loop-back the audio to the sender in addition to distributing it to other participants.
-d
Make the server printing out debug information.

See Also

SIP and RTP

Author

Huwei Zhang at Columbia University, Department of Computer Science

Copyright

Copyright 2000 by Columbia University; all rights reserved

Permission to use, copy, modify, and distribute this software and its documentation for not-for-profit research and educational purposes and without fee is hereby granted, provided that the above copyright notice appear in all copies and that both that the copyright notice and warranty disclaimer appear in supporting documentation, and that the names of the copyright holders or any of their entities not be used in advertising or publicity pertaining to distribution of the software without specific, written prior permission.

The copyright holders disclaim all warranties with regard to this software, including all implied warranties of merchantability and fitness. In no event shall the copyright holders be liable for any special, indirect or consequential damages or any damages whatsoever resulting from loss of use, data or profits, whether in an action of contract, negligence or other tortuous action, arising out of or in connection with the use or performance of this software.