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Marc Todd
Marc.todd@ineoquest.com Jim Welch
jim.welch@ineoquest.com IneoQuest Technologies, Inc
www.ineoquest.com
Summary
Video-on-Demand systems are being implemented using IP over Ethernet switched transport systems. This paper reviews a systematic approach to assure consistent high video quality through proper design and monitoring of parameters such as video source quality and network packet jitter, loss, utilization, and other key parameters. Understanding how these parameters combine to affect video distribution quality is essential to successfully scale such systems to large production environments.
Introduction
Video-on-Demand (VOD) implementations today combine packet switched Ethernet networks, cable television hybrid fiber and coax (HFC) distribution networks, system resource management servers, video servers, and various system support resources to deliver a growing number of video stream options to a user at his command. Assuring that a user receives high quality video regardless of other system activities or other user requests requires both careful system design and continuous monitoring of a very dynamic system. Quality of the user end point video is a function of the MPEG stream source quality, transport network characteristics, and server capacity. Impairments can be either static or dynamic and can originate in the video sources (servers or feeds) or the network infrastructure. Streaming video with its inherent sensitivity to delivery time distortions and packet loss is especially demanding on traditional packet switched networks that, to date, have had little provision for assuring Quality of Service (QOS). This paper will explore some critical network and video source parameters and some approaches that can be followed to assure high quality video delivery in a VOD environment.
Types of Video Impairment
Figure 1 shows a simplified view of a VOD network. A customer can originate a video request via a set top box (STB), which is transported to a resource manager for accounting, authenticating, billing, and ultimately, initiating video playback at the VOD server. This 2 to 20 Mb/s video stream is packetized for transport over the network infrastructure and delivery to the STB which decodes the MPEG stream for display.
Figure 1 – Simplified Block Diagram of a VoD Network
To assure that the customer receives high quality video, clearly one must start with assuring that the video source be high quality. Equally as important, the network infrastructure must deliver the video stream with a minimum of distortion. Impairments of each of these two components can be thought of as either static or dynamic. A static fault such as a video server hardware failure that causes corruption or loss of the video source stream, or a communications link failure that causes loss of the source stream, or any other path failure of a permanent nature must be detected as soon as possible and corrective action taken. A dynamic fault such as a particular program corruption, a transient occurrence of network congestion, or congestion at one or more of the servers may clear up, given a period of time, with no intervention but still results in poor video quality or no video at all to the customer and negatively affects customer satisfaction. These dynamic issues must also be detected as soon as possible so that corrective action can be taken. Ideally, instrumentation should indicate the instantaneous operating margin and be able to log the same such that personnel with minimum training can be alerted to impending problems rather than waiting until an issue causes a fault that can be detected by the customer.
A wide range of video source quality issues can be addressed by monitoring the MPEG stream. The critical elements within an MPEG transport stream are the building blocks on which the MPEG stream is built. If any one of these elements fails, decoding will typically cease, resulting in a total loss of sound and picture. These critical building blocks are defined as First Priority elements in TR 101 290. These guidelines are primarily designed to check the integrity of an MPEG-2 transport stream in an operational environment. The aim of these tests is to provide a “health check” of the most important – 1st priority – elements. It is worth noting that full quality of service cannot be realistically achieved unless TR 101 290 2nd and 3rd priority functions are also correct.
Evaluations such as those recommended by ETSI TR 101 290 for decodability, quality, and reliability can be performed in real time with instruments such as the Tektronix MTM400 MPEG Transport Stream Monitor, for example, to assure high quality video sources whether from a Video server or other program feeds.
Addressing impairments in the network infrastructure that can lead to a customer visible fault is equally important. MPEG video carried in Transport Streams via conventional Ethernet packet switched networks can arrive at its destination node with a time distortion – that is, the time between packets at the destination is different from that at the source. If packets are delayed by the network such as in Figure 2, some packets arrive in bursts with interpacket delays shorter than when they were transmitted such as those in in the 1 Second group while others are delayed, as in the 3 Second group, such that they arrive with greater delay between packets than when they were transmitted from the source. This is defined as packet jitter and is the time difference between when a packet actually arrives and the expected arrival time. A receiver (decoder) displaying the video at its nominal rate must accommodate the varying input stream arrival times by buffering the data arriving early and assuring that there is enough already stored data when packets arrive late by prefetching data before beginning to display the video. Thus there is a design tradeoff made in all systems between having sufficient buffer memory to smooth all network delays and encountering excessive delay due to a large buffer memory.
Figure 2
For a given size de-jitter buffer then, there is a maximum amount of network jitter that can be accommodated without buffer overflow or underflow -- either of which would result in a poor video display.
Similarly, the network switch/router infrastructure uses buffers at each node to accommodate instances where multiple input streams arrive simultaneously and all are destined for a single output port. These buffers (queues) must also be sized appropriately to handle network congestion that might be due to the way traffic is routed through the infrastructure or perhaps due to differing link speeds in the infrastructure. If the switches implement methods for delivering Quality of Service (QOS) using packet metering algorithms, they may intentionally hold back packets to meet a QOS transmission specification further using buffer memory. Should such a buffer overflow, packets would likely be lost. Environmental electrical noise might create corrupted packets also leading to packet loss. Even small packet loss rates result in a poor video display.
Packet delay variation and packet loss have been shown to be the key characteristics in determining whether a network can transport good quality video. These characteristics can be used to assess how well a network can transport video and to troubleshoot problems in networks whose performance has deteriorated due to reconfiguration or changing traffic loads and, finally, as an indication of margin, or headroom, in the network.
Assuring Customer Satisfaction
Viewing a video decoder’s output picture is the traditional, basic method to assure that acceptable video is being delivered to a customer’s location but is lacking in several important capabilities:
· Audit records. If a video impairment occurs, it may be missed since viewing a picture does
not occur 100% of the time. In either case, there is no automated way to record the event
with an accurate timestamp to possibly correlate its occurrence with other network events
to aid in troubleshooting.
· Subjective result. The severity of a fault is subject to the viewer’s evaluation. Verification
results may vary with different viewers.
· Timliness. If an impairment is noted, by definition, it’s too late to prevent delivery of the
fault.
· Troubleshooting help. If an impairment is noted, there are few clues about what went
wrong or how to repair it.
· Margin Indication. If the video is not impaired, there is no indication of the system’s
operating margin or the probability of a future impairment.
A better alternative to assure acceptable video delivery is to measure the video source quality against a set of video industry quality standards and to measure the known, critical network parameters associated with streaming content; namely, packet delivery delay variation and packet loss as described above. It is straightforward to detect lost packets in a network infrastructure by monitoring the links between network devices, but it is important to note that even if the infrastructure does not drop packets, it may affect delivery times to a video decoder such that the decoder buffer overflows or underflows resulting in packet loss within the decoder. Therefore it is equally important to monitor, log, and trigger alarm conditions based on packet jitter values. By tracking jitter values as network utilization grows, the margin for error due to buffer overflows or underflows will indicate how close the dynamic behavior of the network is to the limit at which visible errors occur. Taking action to address excessive network jitter will help prevent cases where network behavior causes conditions visible to the customer.
To troubleshoot concerns identified with the network and to aid in system configuration and monitoring, other measurement parameters may also be desired.
· Network Utilization. Tracking the instantaneous, minimum, and maximum overall network
utilization is needed to verify that sufficient raw bandwidth is available for a stream on a
network. High utilization level is also an indicator that localized congestion is likely due to
queue behavior in network components. Jitter measurements thus provide a measure of
the results of congestion on a given stream.
· Video stream statistics such as:
o Instantaneous Flow Rate (IFR) and Instantaneous Flow Rate Deviation (IFRD). The
measured IFR and IFRD confirm a stream’s nominal rate and, if not constant over time,
gives insight into how a stream is being corrupted.
o Average Rate in Mb/s. This measure indicates whether a stream’s rate being analyzed
conforms to its specified rate over a measurement time. This is the longer term
measurement of IFR.
o Stream Utilization in percent of network bandwidth. This measure indicates how much of
the available network bandwidth is being consumed by the stream being analyzed.
In summary, to assure the highest quality and most reliable customer video delivery, a system must be designed and maintained to guarantee excellent video source quality, low network jitter margin and low packet loss, an then continuously monitored with reliable instrumentation to both quickly detect static failures and dynamic behavior that may signal an impending failure so that immediate preventative action may be taken.
Conclusions and observations
Streaming Video transported by packet switched networks is an emerging technology and requires a systematic approach to network performance measurement to assure quality. Assuring high quality at the end-points of such networks requires measuring and monitoring of the payload quality, packet loss, packet jitter and its relation to buffer limits in the system components as well as other network parameters such as instantaneous flow rate and network utilization. As higher numbers of streams are served, it is critical is that system parameter bounds are maintained. Measurement and monitoring at several points in the system may be required. Instrumentation from Tektronix (www.tektronix.com) and IneoQuest Technologies (www.ineoquest.com) can simplify tracking the instantaneous health of VOD systems.
This system approach to measuring end-point video quality of streaming video is focused on video delivery requirements and sets demands on the network transport. The faster the MPEG rate the more constricting on the boundaries of the network. For example, using the same buffer at the end-point for DVD quality MPEG-2 vs. High Definition quality MPEG-2, the requirements on the network are quite different, even though the increase in data rate is well inside the Ethernet network boundaries.
DVD quality MPEG-2 has a data rate around 3.75 Mb/s, High Definition (HD) quality MPEG-2 is about 19.3 Mb/s. The rate is approximately 5X on the network and means a buffer can handle 1/5th the network jitter or interpacket time drift. So a jitter/delay of 30ms may be fine for the DVD case as the drain rate from the buffer is less then the HD case. While a jitter/burst of 5 packets may be handled by the HD and not the DVD because HD drains the buffer faster than DVD.
The drain rate of the buffer imposes real limits and behavior profiles on the network transport. Likewise the buffer size has the same effect; in fact, the drain rate and the buffer are directly related with respect to the network transport. For example if the buffer is increased by the same factor as the increase in the speed between the rates of HD MPEG-2 over the DVD MPEG-2 in the last example, the behavior differences would be minimized. Note too that other system parameters may be affected by faster or slower streams at the server and/or in the network. In conclusion, as this technology matures more video streams will be transported over packet switched networks but measurements and monitoring of the system must be considered to insure a quality video product for the consumer.
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