How to handle clock skew in audio streaming

Problem : 1. Its live audio-video streaming over Wiffi + udp network. 2. Stream : Mpeg2Ts 3. Player Framework : gstreamer. 4. Pipeline Appsrc ----> tsdemuxer -----> audio-queue---->faad decoder--->alsasink - ----->video-queue->vpudecoder ---->videosink 5. Audio device is configured for processing 48000 samples per second. 6. Senders clock is faster than receiver clock and i get this info by tracking the pcr value coming in the stream and receiver system clock. After 1 hour there is 8 second difference between sender and receiver clocks. 7. So problem is sender is sending more samples in one seconds with respect to receiver clock due to this latency between sender and receiver is keep increasing with the time.