We propose a simplified model of real-time joint source-channel coding, which can be used to adaptively determine the quality-optimal code rate of forward error correction (FEC) coding. The objective is to obtain the maximum video quality in the receiver, while taking time-varying packet loss into consideration. To this end, we propose a simplified model of the threshold set of the residual video packet loss rate (RVPLR). The RVPLR is the rate of residual loss of video packets after channel decoding. The threshold set is defined as a set of discrete RVPLRs in which the FEC code rate must be changed in order to maintain minimum distortion during increases or decreases of channel packet loss. Because the closed form of the proposed model is very simple and has one scene-dependent model parameter, a video sender can be easily implemented with the model. To train the scene-dependent model parameters in real-time, we propose a test-run method. This method accelerates the test-run while remaining sufficiently accurate for training the scene-dependent model parameters. By using the proposed model and test-run, the video sender can always find the optimal code rate on the fly whenever there is a change in the packet loss status in the channel. An experiment shows that the proposed model and test-run can efficiently determine the near-optimal code rate in joint source-channel coding.

The transmission of duplicate packets provides a loss-resilience without undue time-delay in the video transmission over packet loss networks. However, this method generally deteriorates the problem of traffic congestion because of the increased bit-rate for duplicate packet transmission. In this paper, we propose a set of techniques for an efficient packetization and transmission of duplicate video packets. The proposed method transmits the duplicate packet containing high priority data that is quite small in volume but very important for the reconstruction of the video. This method significantly reduces the required bit-rate for duplicate transmission. An efficient packetization method is also proposed to reduce additional packet overhead which is required for transmitting the duplicate data. It is shown by simulations that the proposed method remarkably improves the packet loss-resilience for video transmission only with small increase of redundant duplicated data for each slice.

Joint source-channel coding (JSCC) is a method to jointly allocate the given total transmission bitrate to the source coding and channel coding to maximize the video quality at the receiving end. In this paper, we propose a practical model for efficiently determining a near-optimal code rate for JSCC in real-time video communications. The conventional code rate decision schemes using analytical source coding distortion model and channel-induced distortion model are usually complex, and typically employ the process of model parameter training which involves potentially high computational complexity and implementation cost. To avoid the complex modeling procedure, we introduce a very simple video quality model based on the playable bitrate which is defined as the total bit amount per unit time that is not affected by the channel loss during transmission including correctly recovered bits by the channel decoder. Because the video quality at the receiving end is clearly commensurate with the playable bitrate, we can easily determine the quality-oriented near-optimal code rate by finding the code rate that maximizes the playable bitrate at the sender side. The proposed playable bitrate model is very simple because it does not require the complex training procedure for obtaining model parameters, which is usually required in the conventional code rate decision method. It is shown by simulations that the proposed code rate decision scheme based on the playable bitrate model can efficiently determine the near-optimal code rate for JSCC in terms of high accuracy on the optimal code rate.