I suspect that the problem is that you are not (in your "FramedSource" subclass) setting "fDurationInMicroseconds" before delivering each chunk of PCM data (to your downstream "uLawFromPCMAudioSource" filter). Before completing the delivery (i.e., before calling "FramedSource::afterGetting()"), you should set "fDurationInMicroseconds" based on the number of samples in the chunk of data, and the sampling frequency.
(If you don't set "fDurationInMicroseconds", then it will get set to the default value of 0, which will cause the downstream "RTPSink" object to request new data immediately after sending each RTP packet. Because you're streaming from a buffer, that's probably not what you want.) Ross Finlayson Live Networks, Inc. http://www.live555.com/
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