Handling Audio streams accross a network

Hi,
I have written a Computer Telephony Interface package for a SIP based PBX system and I'm currently exploring the possibility of turning it into a fully functional SIP softphone. The program has a basic SIP stack which I have extended to include the call setup messages, I just need some advice in how I can handle RTP properly.
The PBX uses G.711 A-law audio codec and I have noticed that the JMF doesn't support type 8 RTP for A-law.
Here is the scenario, a call comes into the program over the network using a standard Datagram socket, when my program answers the call I can see a load of bytes fired into my system every 20ms (160 bytes audio), I need a solution to convert those bytes into an audio stream and play them through my computer speakers.
Can anyone give me any guidance on this?
Thanks
Brett

I'm currently doing something similar, and I think this explains a lot
http://java.sun.com/developer/JDCTechTips/2002/tt0319.html#tip1
atleast how to play those data bytes as audio and how to do some
encodings.
If you're using datagram socket, the packets come in in a random order,
so doesn't your audio sound a bit weird? Unless you have some mechanism
to put the packets in correct order.
kari-matti

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