l'applet Jain Sip est ce que ca marche pour vous ?

Bonjour,
j'essai de compiler l'applet jain sip pour l'utiliser dans mon projet mais je ne sais pas comment la configurer et la doc et cruelement insufisante.
est ce que quelqu'un la déja compiler et utiliser avec succes ??!!
sinon si quelqu'un peut m'aider à configurer sur tous qu'il a des variables comme public interface et private interface que j'en sais rien donner comme valeur !!!!

configuration_gateway.xml

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<?xml version='1.0' encoding='us-ascii'?>
<CONFIGURATION> 
<!-- pstn_gateway is where the cisco dial peer is located -->
<SIP_STACKS pstn_gateway="129.6.55.81" >     
 
    <SIP_STACK 
        stack_name="public interface"
<<<<<<< configuration_gateway.xml
        stack_IP_address="129.6.100.252" 
=======
        stack_IP_address="127.0.0.1" 
>>>>>>> 1.8
    >       
        <LISTENING_POINT port="4000" transport="tcp" />
    </SIP_STACK>
 
    <!--
    <SIP_STACK 
        stack_name="private interface"
        stack_IP_address="129.6.230.77" 
    >       
        <LISTENING_POINT port="4000" transport="udp" />
    </SIP_STACK>
    -->
</SIP_STACKS>
 
<!-- Push data between two networks -->
<RTP_GATEWAY
    enable="false"
    min_rtp_port="9000"
    max_rtp_port="9990"
    max_packet_size="5000"
/>
 
<LOGGING
enable_debug="false"
server_log="./logs/server_gateway_log.txt"
bad_message_log="./logs/bad_message_gateway_log.txt"
debug_log="./logs/debug_gateway_log.txt"
/>
 
<!-- send mail when phone is busy  or the other party is unreachable -->
<MAIL_TASK 
	enable = "false"
	waitTime = "15"
/>
 
 
 
</CONFIGURATION>

J'ai changer le fichier xml comme ceci :

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<?xml version='1.0' encoding='us-ascii'?>
<CONFIGURATION> 
<!-- pstn_gateway is where the cisco dial peer is located -->
<SIP_STACKS pstn_gateway="192.168.1.40" >     
 
    <SIP_STACK 
        stack_name="public interface" stack_IP_address="127.0.0.1">
        <LISTENING_POINT port="5060" transport="tcp" />
    </SIP_STACK>
 
 
    <SIP_STACK 
        stack_name="private interface"
        stack_IP_address="127.0.0.1" 
    >       
        <LISTENING_POINT port="4000" transport="udp" />
    </SIP_STACK>
 
</SIP_STACKS>
 
<!-- Push data between two networks -->
<RTP_GATEWAY
    enable="true"
    min_rtp_port="9000"
    max_rtp_port="9990"
    max_packet_size="5000"
/>
 
<LOGGING
enable_debug="false"
server_log="./logs/server_gateway_log.txt"
bad_message_log="./logs/bad_message_gateway_log.txt"
debug_log="./logs/debug_gateway_log.txt"
/>
 
<!-- send mail when phone is busy  or the other party is unreachable -->
<MAIL_TASK 
	enable = "false"
	waitTime = "15"
/>
 
 
 
</CONFIGURATION>

Capture d'ecran : http://rousseau.john.free.fr/sip/erreur/