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 _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
