Telecommunication networks are carrying increasing amounts of digital multimedia content (e.g. video and/or audio data). Standards have been developed for transporting digital multimedia content across the networks. Part 1 (Systems) of the Moving Pictures Expert Group (MPEG-2) standard defines a Transport Stream (TS) for carrying multimedia content across somewhat unreliable mediums such as broadcast channels, internet protocol networks, etc. The transport stream consists of packets that carry video or audio data in their payload. The data may be compressed to maximize usage of the bandwidth available, although data compression of the payload is not a requirement of the Transport Stream standard.
The digital multimedia content must be transported across the network in a reliable and timely manner to the end user. Streaming video services (e.g. Internet Protocol television (IPTV), video conferencing, video-on-demand, etc.) are especially sensitive to delay, jitter, or data loss, which can all negatively impact the quality of the end user's experience.
Determining the performance of a network that carries digital multimedia content is an important element to the successful design and operation of such a network. Some network parameters that are commonly determined include latency, jitter, and packet loss. Another measure of a network's performance is known as the Media Delivery Index (MDI). The MDI has two components: the delay factor (DF) and the media loss rate (MLR). The MDI is expressed as two numbers separated by a colon: DF:MLR.
The DF component of the MDI is the maximum difference, observed at the end of each network packet, between the arrival of media data and the drain of media data. To calculate DF, consider a virtual buffer VB used to buffer received packets of a stream. When a packet Pi arrives during a measurement interval, compute two VB values, VB(i,pre) and VB(i,post) as follows:
                              VB          ⁡                      (                          i              ,              pre                        )                          =                                            ∑                              j                =                1                                            i                -                1                                      ⁢                          S              j                                -                      MR            *                          T              i                                                          (        1        )                                          VB          ⁡                      (                          i              ,              post                        )                          =                              VB            ⁡                          (                              i                ,                pre                            )                                +                      S            i                                              (        2        )            where Sj is the media payload size of the jth packet, Ti is the time, relative to the previous packet, at which packet i arrives in the interval, and MR is the nominal media rate in bytes per second. VB(i,pre) is the virtual buffer size just before the arrival of Pi, and VB(i,post) is the virtual buffer size just after the arrival of Pi. This calculation is subject to the initial condition of VB(0,post)=VB(0,pre)=0 at the beginning of each measurement interval. A measurement interval is defined from just after the time of arrival of the last packet during a nominal period (typically 1 second) to the time just after the arrival of the last packet of the next nominal period.
The DF is calculated as follows:
                    DF        =                                            max              ⁡                              (                                  VB                  ⁡                                      (                                          i                      ,                      post                                        )                                                  )                                      -                          min              ⁡                              (                                  VB                  ⁡                                      (                                          i                      ,                      pre                                        )                                                  )                                              MR                                    (        3        )            
The MLR is simply defined as the number of lost or out-of-order packets per second.
For more information regarding the MDI, DF, and MLR measurements, please refer to the following publications: “A Proposed Media Delivery Index (MDI)”, by J. Welch and J. Clark, published in April 2006 by the Internet Engineering Task Force as IETF RFC 4445 and available at the following URL: http://www.rfc-editor.org/rfc/rfc4445.txt; and “IPTV QoE: Understanding and Interpreting MDI Values”, a white paper published by Agilent Technologies on Aug. 30, 2006 and available at the following URL: http://cp.literature.agilent.com/litweb/pdf/5989-5088EN.pdf
Prior art methods for measuring delay factor are unsatisfactory for several reasons. For one thing, the prior art methods are not very accurate. For example, the typical value used for the media rate MR in equations (1) and (3) is the advertised bit rate at which the data is to be consumed at its destination. However, the advertised bit rate is only an approximation because it only includes the video stream, not the audio stream or other meta data (e.g. program guide information that might be embedded in the TS packet stream) While it is possible to restrict the measurement of the DF to video packets only, thus addressing the rate approximation issue, this results in more complicated and inferior measurements for network performance. Furthermore, the actual bit rate will vary depending on the accuracy of the encoder for the data bitstream. The encoder will do its best to achieve the advertised bit rate, but in reality can only approximate it.
Some prior solutions also required that the payload in the packets be filled with instrumentation data, which rendered the packets incapable of containing valid multimedia data. However, it is desirable to keep the multimedia data within the transport stream packet intact, so as to recreate actual network conditions as closely as possible.
Therefore, there remains a need for improved performance testing of a network carrying multimedia traffic, in particular streaming video content.