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Video On Demand and IPTV : Technologies QoS (English)

Video-on-Demand and IPTV: Two Distinct Technologies (Par EXFO)

  • Florin Hodis, Member of Technical Staff, Access Business Unit

Video-on-demand (VoD) services are being used by telcos as a way to leverage their digital subscriber line (DSL) market base to provide a more interactive level of video entertainment. This article focuses on access networks delivering VoD using xDSL on the physical layer.

In a typical telco network environment, the access network transports not only VoD, but also voice, high-speed Internet and Internet protocol TV (IPTV). Note that the physical-layer choices for access networks are ADSL2+ or VDSL2.

Both VoD and IPTV services deliver video content to subscribers, yet they use slightly different technologies. VoD is based on unicast technology, whereas IPTV is based on multicast technology. The most common IP address is a unicast address, which either has a single sender or a single receiver.  This address can be used for both sending and receiving; however, sending the same data to more than one unicast address requires the sender to send the data many times over, i.e., once for each recipient.

A multicast address is associated with a group of interested receivers (according to RFC 3171, addresses to are designated as multicast addresses). The sender sends a single datagram to the multicast address, and the routers take care of making copies and sending them to all the receivers that have registered their interest in data from that sender.

IPTV et VoD transmission Multicast et unicast

Figure 1. Unicast vs. multicast

In order to be able to communicate with the VoD or IPTV network infrastructure, subscribers need to support different protocols: real-time streaming protocol (RTSP) for VoD, and Internet group management protocol (IGMP) for IPTV.

RTSP establishes and controls either a single or several time-synchronized streams of continuous media, such as audio and video. However, RTSP does not typically deliver the continuous streams itself; rather, it acts as a network remote control for multimedia servers. The streams controlled by RTSP may use real-time protocol (RTP), but the operation of RTSP does not depend on the transport mechanism used to carry continuous media.

IGMP is a communications protocol designed to allow the management of the IP multicast groups memberships. IGMP is used by IP hosts and adjacent multicast routers to establish multicast group memberships. IGMP is used for IPTV multicast, which allows for joining and leaving multicast groups in a more efficient way—with minimal use of network resources.

Services IPTV

Figure 2a. IPTV services

Services Vidéo On Demand

Figure 2b. VoD transport services

For both VoD and IPTV, the compression algorithm can be based on MPEG-2, H.264 (MPEG -4, Part 10) or VC-1—both can deliver HD or SD programming and, in both cases, the compressed and encrypted content is carried by MPEG2-TS.

Factors Affecting VoD Quality of Service

  • Network Bandwidth

Ultimately, the total amount of video-stream data that can be sent is limited by the bandwidth provisioned for the access network. Any increase in bandwidth demand that goes beyond the maximum capacity of the link will result in video packets being lost, which causes impairments on the screen. Since VoD is based on unicast, every subscriber must request his own video stream and there is no sharing.

  • Impulse Noise

The copper loop plant is susceptible to short impulses caused by external sources. These impulses cause large bursts of errors, which could have a significant impact on the quality of the video picture.

  • Packet Loss

IP packet loss can represent a single unnoticeable missing point of the video sequence or a large period of degraded, pixilated or unavailable image.

  • Jitter

Jitter affects MPEG-2 or H.264 video streams, resulting in poor video-image quality.  The transport stream carrying a program clock reference may be affected by jitter as well; this condition has a direct impact on the decoding process performed by the set-top box (STB).

VoD Metrics

  • Packet Loss

Packet loss occurs when one or more packets of data traveling across the access network fail to reach their destination. If the missing packets were related to the reconstruction of the I Frames, there is a good chance of losing the video signal for a short period of time. If the missing packets are related to B or P Frames, the impact is less severe, but image-quality issues could still be experienced.

  • Packet Delay

In a packet-based network, it is quite common that the route for transporting the packets is not always the same and that the packets may arrive at different times and out of order. The RTP protocol enables the out-of-order arrival of packets. Since every RTP packet has a sequence number, as long as the delay does not exceed the size of the receiving decoder buffer, the packet can be processed and placed in the right position for decoding. However, if the delay exceeds the buffer, the packet is dropped and considered lost.

  • Packet Interarrival Jitter

Packet interarrival jitter is an estimate of the statistical variance of the RTP data-packet interarrival time, which is measured based on the RTP time stamp. The interarrival jitter is defined as “the mean deviation of the difference in packet spacing at the receiver compared to the sender for a pair of packets”.

  • PCR Jitter

Transport stream (TS) is a format specified in MPEG-2, Part 1 (ISO/IEC standard 13818-1), which contains seven packets of 188 bytes each (184 bytes of payload and 4 bytes of packet header). TS can also be used to transport H.264 encoded video.

Format de trame de transport

Figure 3. Representation of the transport stream format

Included in this transport stream are clock-synchronizing parameters that are sent at regular time intervals. These clock-synchronizing fields, referred to as program clock reference (PCR), are the instantaneous value or a sample of the 27 MHz system time clock (STC) located at the MPEG video encoder. The PCR in the transport stream enables the MPEG decoder to recreate the encoder’s system time clock; this recreated clock guarantees that the decoded video output operates at the same rate as the video signal input to the MPEG encoder.

MPEG transport stream is transmitted over any real network being exposed to certain effects caused by network components, which are not ideally transparent. One of the pre-dominant effects is the acquisition of jitter in relation to the PCR values and their position in the TS. If the PCRs do not arrive with sufficient regularity, then this clock may jitter or drift. Recovery of the PCR enables the decoder (the STB) to synchronize its clock to the same rate as the original encoder clock. High PCR jitter levels may cause the receiver/decoder to go out of lock, which will affect the video quality displayed on the TV.

  • PID

TS packets have a fixed length of 188 bytes with a minimum 4-byte header and a maximum 184-byte payload. Within the 4-byte header, there are multiple data fields—some of which are used for management and control purposes.

Structure de trame de transport

Figure 4. Transport stream structure

Source: iec.org

The PID is part of that 4-byte header and represents a unique address identifier for the type of packet or payload carried by the TS. Video or audio packets in the stream need to have a unique PID; this enables the decoder or STB to process the packets accordingly.

It is necessary to ensure that the PID assignment is done correctly and that there is consistency between the payload structure identifier (PSI) tables content (set of tables required for demultiplexing of the MPEG and sorting out which PIDs belong to which programs) and the associated video and audio streams in order for an STB to reconstruct a media stream from all its video, audio and table components. This is one of the main components of MPEG monitoring and testing.

Finding and correctly decoding a specific PSI table determines whether or not the STB will subsequently be able to identify and decode the video and audio information for the specific stream.

For service providers, the ability to see the PID configuration and consistency, as well as the ability to monitor packet loss and rates for specific video packets, audio packets, or PSI tables, allows them to isolate and troubleshoot problems affecting VoD service.

  • Media Delivery Index

Media delivery index (MDI)—RFC 4445—is a scoring mechanism that combines jitter and packet loss in order to determine the quality of the network to transport good quality video; it does not take into consideration the encoding method. MDI measurements can be used as a diagnostic tool or as a quality indicator for monitoring a network intended to deliver applications such as streaming media, VoD, IPTV, VoIP or other information sensitive to arrival time and packet loss.

Delay Factor
The delay factor (DF) is the maximum difference observed at the end of each media stream packet between the arrival of the media data and the drain of media data. This assumes that the drain rate is the nominal constant traffic rate for constant bit rate streams or the piece-wise computed traffic rate of variable rate media stream packet data. The DF is the maximum observed value of the flow-rate imbalance over a calculation interval. This buffered media data in bytes is expressed in terms of how long, in milliseconds, it would take to drain (or fill) this data at the nominal traffic rate to obtain the DF.

The DF gives a hint of the minimum size of the buffer required at the next downstream node. Greater DF values also indicate that more network latency is necessary to deliver a stream—due to the need to pre-fill a received buffer before beginning the drain to guarantee no underflow. For complete details on the DF calculation algorithm, refer to RFC 4445.

Media Loss Rate
The media loss rate (MLR) is the count of lost or out-of-order flow packets over a selected time interval. The flow packets are packets that carry streaming application information.

MDI is the combination of the delay factor and media loss rate test results (displayed in DF: MLR format).


In order to detect network malfunctions that could affect the quality of VoD service, it is important to perform accurate network testing to locate the source of the problems. Metrics—such packet loss, jitter, packet delay and MDI—have all been defined to help service providers achieve efficient VoD transmission, thus allowing them to retain and increase their customer base.

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