Video Conferences and Internet Traffic

As the bandwidth available on networks and the speed of computers increases, real-time transmission of video between general purpose work stations becomes a more and more realistic application. However, even with a high speed network, video has to be compressed before transmission. 

This leads to highly variable transmission rates because the amount of information to code between two images greatly varies, ranging from very low for still scenes to very high for sequences with many scene changes. Packet switched networks such as the Internet is very well suited for transmitting such variable bit rate traffic.

H.261 codecs designed for operation over ISDN circuits produce a bit stream composed of several levels of encoding specified by H.261 and companion recommendations. The bits resulting from the Huffman encoding are arranged in 512-bit frames, containing two bits of synchronization, 492 bits of data and 18 bits of error correcting code.

Errors in a video stream require a different form of correction than errors in a normal data stream. Tests transmitting video stream over a standard TCP connection allowed us to transmit data over the Internet without concern of lost or out of sequence packets because of TCP reliability.

On the Internet, most packet losses are due to congestion rather than transmission errors. Alternatively, packets can be delayed or received out of order. This could happen as a result of the routing and flow control in the network. Due to real-time requirements, delayed video packets are considered as lost packets if delay exceeds a maximum delay value. Using UDP, no mechanism is available at the sender to know if a packet has been successfully received. It is up to the application (i.e., coder and decoder) to handle packet loss and re-sequencing of out of order packets.

Each RTP packet includes a header in which a sequence number field and timestamp are stored. The sequence number is incremented by one for each packet sent, whereas, the timestamp reflects the time when the frame was grabbed. Packet losses can be detected using the RTP sequence number.

Video Conference on the Internet could well be a killer application; the network administrator’s nightmares are probably already populated by thousands of hosts all sending megabits of videos over the net and swamping the T3 based backbones.

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