Real-time media networks
Oct 19, 2007 1:39 PM
Network issues
In theory, a GigE network can carry 100Mb/s compressed video, but in real-world installations, three major factors can degrade network performance to unacceptable data rates — packet jitter, latency and loss.
Packet jitter is the variation in the time between packets arriving at a device. This can be caused by network congestion, timing drift or route changes.
Latency is the amount of time it takes a packet of data to move across a network connection. For TCP (Transmission Control Protocol) there is latent time when the computer that sent the packet waits for confirmation that the packet has been received. For UDP (User Datagram Protocol), no handshake is required and latency is reduced.
Best-effort UDP is frequently used for real-time media transfers. When a routing network becomes saturated with more traffic than a device can handle in real time, packets are dropped. This has dire consequences on media quality.
Even when all packets are received, however, latency and jitter can cause problems for real-time media transfers. Jitter and latency can vary for each packet. Referring to the numbers in Table 4, hundreds, or even thousands, of data packets will arrive asynchronously at transcoding devices.
If latency can range from 100ms to seconds while packet jitter can be greater than 40ms, audio and video transfers will take more than one video frame time duration. Table 5 shows how many packets for a given video data rate will be in transit on a network for a given frame rate. The simultaneous traveling on the network of packets for many video programs at any given instant multiplies congestion. As a result, jitter and latency times increase.
| 60fps | 30fps | 24fps | |
| 40Mb/s | 55/9 | 111/18 | 138/ |
| 100Mb/s | 138/23 | 277/46 | 347/57 |
| 270Mb/s | 375/62 | 750/125 | 937/156 |
Table 5. Video data rate and Ethernet/jumbo Ethernet packets vs. frame rate
The indeterminate nature of an IT network is fundamentally at odds with its use in media applications. It could be argued that Ethernet and IP are the worst possible network technologies for real-time media systems.
A modern broadcast technologist must think in terms of both the broadcast engineering and IT domains when designing a networked infrastructure that must support real-time transcoding. The impact of the loss of data is a factor of video data rate, compression ratio, network bandwidth and packet size with respect to packet jitter, latency and loss.
Tomorrow’s networks
Connection-oriented routing devices and tagging for Quality of Service (QoS) are traffic engineering techniques intended to facilitate the particular needs of transferring video over IT networks. ATM, with its deterministic TDM structure, in many ways is ideally suited for media network applications. RTP (Real Time Protocol) and RTSP (Real Time Streaming Protocol) also attain performance that can support adequate real-time data delivery.
These protocols, however, were never intended to carry digital video, particularly HD resolutions. Now, there is a need to develop new real-time network protocols specifically engineered to support HD video.
Bursty network performance has an impact on real-time format conversion equipment. The next Transition to Digital will delve into how this problem is solved and address transcoding and format conversion for multichannel distribution.
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