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RTP over UDP implementation for A/V multicast-streaming

Objetive

  • Analyze the impact on the network by the A/V streaming. Calculate delay, jitter and losses.
  • Implementation of RTP Protocol. Spec of RTP
  • Chat-multicast over UDP
  • Webcam USB multicast-streaming (jpeg compression)
  • Microphone multicast-streaming (encoded with a-law[g711])

Disclaimer: The sound only reproduce if the videochat-client form is focus.

Wireshark RTP decoding

If we capture the traffic of videochat-server with wireshark, we can decode the UDP packets with the RTP spec. In case of JPEG frames, we find a RTP Stream and in other case, the audio streaming, we find too a diferent RTP Stream

Usage of wireshark for RTP statistics

Pics of a example traffic capture! 6 7 8

Analyze

On the client, we implement the delay, jitter and losses for every RTP received. We calculate this values for each 1sg

Pics of client! 5

Pics of server! 2

Spec of RTP. Appendix for jitter calc

Analysis of RTP Packet Delay and Loss

File explanation

  • ALaw. Teacher files to encode and decode audio with ALaw
  • RTPStream. My implementation of RTP spec. Create the RTP header, create the packet and send it over the network.
  • videochat-client. Client that connect to the net, receiving the incoming RTP A/V packets and show that to the user (reproduce and display).
  • videochat-server. Server to wrap the audio and the video, encapsulate and send over the net.

External references

  • WebCamLib + WebCamWrapper => C++ Lib that teacher give to wrap USB cameras
  • DirectX DirectSound => To capture the sound
  • RTPStream => My implementation of RTP Spec
  • ALaw => To encode/decode PCM.

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