ip-test.1.0.0.source-code.peers.xml Maven / Gradle / Ivy
<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<!--
This file is part of Peers.
This program is free software: you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation, either version 3 of the License, or
any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program. If not, see <http://www.gnu.org/licenses/>.
Copyright 2007-2013 Yohann Martineau
-->
<peers xmlns="http://peers.sourceforge.net"
xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance"
xsi:schemaLocation="http://peers.sourceforge.net peers.xsd">
<!-- a specific address can be specified here, may you have several
network interfaces, or several addresses on a specific interface,
you can specify an address to bind on here. -->
<!-- Example: 192.168.1.20 -->
<ipAddress></ipAddress>
<!-- username (corresponding to the user part of your sip uri) -->
<!-- Example: alice -->
<userPart></userPart>
<!-- domain (corresponding to the domain part of your sip uri) -->
<!-- Example: atlanta.com -->
<domain></domain>
<!-- if password is empty, no REGISTER message is sent -->
<!-- Example: 1234 -->
<password></password>
<!-- you can specify an outbound proxy for registration and calls -->
<!-- Example: sip:192.168.1.20;lr -->
<outboundProxy></outboundProxy>
<!-- you can specify the sip listening port you want, 0 can be used to
choose a random free port -->
<!-- Example: 5060 -->
<sipPort>0</sipPort>
<!-- you can specify an even rtp port to use for incoming and outgoing rtp
traffic, 0 can be used to choose a random even free port -->
<rtpPort>0</rtpPort>
<!-- if you need to specify an authorization username different than
the userPart, if empty will default to what is specified in userPart -->
<authorizationUsername></authorizationUsername>
<!-- mediaMode corresponds to the way media is managed. Three values are
possible for this parameter:
- captureAndPlayback: capture sound from microphone, send
corresponding rtp packets, receive rtp packets and play those
packets on speakers.
- echo: receive rtp packets, do not play them on speakers and send
those packets to remote party
- none: no media is capture, played, send nor received
- file: stream audio from audio file provided in mediaFile -->
<mediaMode>captureAndPlayback</mediaMode>
<!-- mediaDebug is a boolean parameter. If set to true, files will be
created in a media directory in peers.home directory. Those files will
contain raw data at input and output of microphone, encoder, rtp
sender, rtp receiver, and speaker. -->
<mediaDebug>false</mediaDebug>
<!-- mediaFile file read and sent during call. This file must be a raw audio
file with the following format: linear PCM 8kHz, 16 bits signed,
mono-channel, little endian. -->
<!-- Example: media/message.raw -->
<mediaFile></mediaFile>
</peers>