tabrezmemon
Programmer
Hi there
I am trying to develop a voice conference server based on jmf. For this i began withtaking two incoming audio RTP streams and merged them into one using the Manager.createMergingDataSource. I then created a player for this merged datasource. This works fine. But on checking the bitrates of the two outgoing stream, which are in g723/rtp format, they are around 9kbps each, while on merging the bitrate of the merged stream being played is around 18kbps which means the merged datasources bitrate is the sum of the two input streams. but doesn't merging mean that the bitrate should remain constant, the bandwidth requirement shouldn't change on merging right? I have used the BitRateControl Interfaces getBitRate method to get the bitrates. please help with with this concept.. and could u please suggest some firm documentaion about implemeniting a conference server using JMF.
thanks
Tabrez
I am trying to develop a voice conference server based on jmf. For this i began withtaking two incoming audio RTP streams and merged them into one using the Manager.createMergingDataSource. I then created a player for this merged datasource. This works fine. But on checking the bitrates of the two outgoing stream, which are in g723/rtp format, they are around 9kbps each, while on merging the bitrate of the merged stream being played is around 18kbps which means the merged datasources bitrate is the sum of the two input streams. but doesn't merging mean that the bitrate should remain constant, the bandwidth requirement shouldn't change on merging right? I have used the BitRateControl Interfaces getBitRate method to get the bitrates. please help with with this concept.. and could u please suggest some firm documentaion about implemeniting a conference server using JMF.
thanks
Tabrez