[asterisk-users] Sending live audio in Asterisk

[Kutman.DK at forces.gc.ca]

Hello,

I am trying to create a Java GUI that will interact with an Asterisk Server. This Java GUI will essentially be a custom made SIP softphone.  I will most likely use the Asterisk-Java Live API to create the connection to the Asterisk server and to open a new call.  Then, I plan to use the JAIN SIP API to initialize the session and the JMF to send the audio streams via RTP when the two users are connected in a call.  I had two questions about this type of system:

1.	I believe I have a good idea of the overall process of opening a SIP session and streaming live audio(phone conversations) via RTP, but is there any Asterisk-	specific sources or examples that start a session via SIP and then transmit the audio via RTP all done through the Asterisk server?

2.	I know there is an rtp.conf file which outlines the ports available for rtp transfer.  How is the actual RTP transfer between users completed through the Asterisk 	server?  I am looking to transmit live audio between the users through the Asterisk server once the call is connected.

Thanks in advance,

Denis Kutman