Quality of Service in IP Networks for Video Contribution and Distribution

Author:Helge Stephansen

Executive summaryThe paper describes the challenges that must be handled by network adaptors that receive MPEG-2 Transport Stream over IP. The most critical issues are jitter and packet loss. Packet loss is adequately handled by Pro-MPEG Forward Correction and is not considered in detail in this document. Please refer to T-VIPS White Paper for details on Forward Error Correction.

The effects of jitter are thoroughly discussed and adequate solution for jitter removal is presented.

In order to avoid these interruptions the long time frequency error of the IP network adaptor must be zero. The best way to achieve this stability is to lock the ASI rate to same frequency source that is used to generate the Transport stream. In practical implementations this means that the IP network adaptor must be locked to GPS source used or generation of SFN Mega-frames.

IntroductionWhy do broadcasters convert from ATM to IP for real time video transport?Transport of video over IP provides a range of benefits compared to traditional ATM or PDH based solutions. These are:

• IP technology can be used for all video transport. IP technology is ready to be used for all type of transport of video to, from, or between studios, venues, production sites, etc. The transport may be in real time using the Pro-MPEG Code-of-Practice no 3 for compressed content or the Code-of-Practice no 4 for uncompressed SDI. For non real time applications such as file transfer in MXF format or other formats, IP is already the best suited media. Thus it is not required to invest in and operate both an ATM network and an IP network, which would be the alternative.

• IP networks are less expensive than ATM networks. ATM networks are more costly than IP networks. This is partly due to a smaller market and smaller competition than in the IP technology. Gigabit Ethernet interfaces cost only a fraction per port compared to corresponding STM/SONET interfaces.

• IP technology is replacing STM/ATM for telco network. Most large telecom operators are constructing or planning to construct new IP network. The purpose is to replace the telephone network with packet switched networks. The traditional connection oriented voice will be replaced by packet routing. Using legacy technology for broadcast contribution and distribution, one will expect that maintenance and operational costs will gradually increase

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Quality of services with IPIP network performanceA benefit with ATM networks is low risk for loss of data due to the rigid rules for insertion of ATM cells in the SDH STM/Sonet frame. In addition is forward error correction included with AAL-1, ATM Adaptation Level 1. This is the most common way used for MPEG-2 transport over ATM. This result in high quality of service compared to traditional IP network where best effort routing is used. However it is possible to achieve the same high level of signal quality by taking the proper precautions in the design of the IP network and video gateways.

The challenges that arise with an “all IP solution” are:• Packet loss Packet loss may be a result of buffer overflow in IP routers.• Jitter Jitter is mainly caused by variable delay in router buffers but there are other sources, as for example rerouting and load balancing.

Figure 1. Transmission of real time video will introduce jitter, packet out or order and packet loss. These distortions can be removedby the receiver as long as the distortion is within certain limits. This document contains a set of test on the robustness of TVG420.

Some IP performance examples.IP network performance characteristics are not generally available. However, Sprint has made a large number of measurements that are available on www. The figure below is not to be taken as general performance but illustrates network behaviour that may be expected from a well managed IP network.

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IPNetwork

Jitter-free Packet Stream

1 2 3 4 5 78

Packet stream with jitter, reordering and loss

MPEG-2TransportStream#1-8

TVG420TX

TVG420RX

RegeneratedMPEG-2

TransportStream#1-8

9

1 2 3 4 5 6 7 8 9

IPNetwork

IPNetwork

Jitter-free Packet Stream

11 22 33 44 55 7788

Packet stream with jitter, reordering and loss

MPEG-2TransportStream#1-8

TVG420TX

TVG420RX

RegeneratedMPEG-2

TransportStream#1-8

99

1 2 3 4 5 6 7 8 91 2 3 4 5 6 7 8 9

Figure 1. Transmission of real time video will introduce jitter, packet out or order and packet loss.These distortions can be removed by the receiver as long as the distortion is within certain limits.This document contains a set of test on the robustness of TVG420.

Figure 2. IP jitter variations measured by Sprint. These figures are taken from http://ipmon.sprint.com/ and represent delay measurements for a 1 hop 622 Mbit/s connection between two sprint exchanges in San Jose over 6 hour period. The jitter for this connection is less than 300 μs.

Figure 3. The performance of IP network will vary greatly depending on a long list of items. These can be distance, type of network, number of routers, bandwidth, and traffic load, public type of network or closed network. These figures are taken from http://ipmon.sprint.com/ and represent delay measurements for a 6 hop 622 Mbit/s connection between San Jose and New York. Two interesting observations may be drawn from the diagrams. Traffic balancing causing packets to go through different paths results in delay difference of in the order of 10

QoS standards for IP NetworkITU-T Y.1541 Network Performance objectives for IP-based services define 6 different service classes for QoS for IP networks [1]. The ITU Y series is considering Global Information Infrastructure and Internet Protocol Aspects with regards to Quality of service and network performance. This recommendation defines performance objectives ranging from services with a low demand for QoS up to VoIP services. Live video services are falling outside the current scope of Y.1541.

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Network Nature of network QoS Classes performance performance objective parameter Class 0 Class 1

IPTD Upper bound on the mean IPTD 100 ms 400 ms

IPDV Upper bound on the 1 − 10–3 50 ms 50 ms quantile of IPTD minus the minimum IPTD

IPLR Upper bound on the packet loss 1 × 10–3 1 × 10–3

probability

IPER Upper bound 1 × 10–4

Table 1: Extract from ITU-T Rec. Y.1541 Network performance objectives for IP -Based services.Class = and 1 are the highest classes of services that are defined and are considering VoIP services. The fields are; IPTD: IP Packet Transfer Delay, IPDV: IP Packet Delay Variation, IPLR: IP Packet Loss Ratio, IPER: IP Packet Error Ratio.

The Pro-MPEG Code-of-Practice #3 gives some advice on what jitter tolerance should be expected. It is recommended that the receiver should handle +- 10-15 ms high frequency (short term) delay variations. In addition a receiver should also be able to handle long term (drift/wander/delay variations in the order of 30 – 40 ms over a 24 hour period. With regards to packet-out-of order it is considered that this will normally be within a range of 10 packets. Pro-MPEG CoP #3 considers 60 ms buffer appropriate to handle both short and long term packet delay jitter.

The Video Service Forum (VSF) which is an organisation for broadcaster, network operators and manufacturers have made a proposal to ITU to include a new class of service in Y.1541 covering live video transport [2]. This contains three applications covering contribution, primary distribution and access distribution. The requirements for latency and delay variation are identical to those of class 0. For compressed video the packet loss ratio is ranging from 3E-10 to 4E-7 after error correction.

This has been included in the new version of ITU G 1050G: “Network model for evaluating multimedia transmission performance over internet protocol” setting a performance range for modelling the IP network.

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Impairment Type Units Range (min to max)

One Way latency ms 20 to 100 (regional) 90 to 300 (intercontinental)

Jitter (peak to peak) ms 0 to 50

Sequential Packet Loss ms Random loss only (except when link failure occurs)

Rate of Sequential Loss sec-1 Random loss only (except when link failure occurs)

Random Packet Loss % 0 to 0.05

Reordered Packets % 0 to 0.001

Table 2. – Impairment Rangers for Well-Managed Network (Profile A). Extracted from ITU G. 1050

Comments to ITU Y1541 and G1050.LatencyY.1541 do es not differentiate between long term and short term delay variations. This is because normal IP sessions have limited duration. Video transport, on the contrary, may be running continuously at a constant bit rate for 24 hours per day, 7 days a week and it is important to have knowledge of both short term and long term delay variations for a connection. The reason for this is that while high frequency delay variations do not influence the average buffer filling but only create unsystematic variations in the buffer filling, long term latency variation on the other hand will lead to semi permanent changes in the buffer filling. This should be compensated for in order to bring the buffer filling back to normal operational level. A redundancy rerouting may for instance increase the latency with 15 ms. In this case the IP receiver will continue to output transport packets from the output buffer at the nominal rate, and the buffer filling will be abruptly reduced by a value corresponding to 15ms of data. Following this, the receiver will try to increase the amount of data in the buffer in order to restore the average latency (i.e. buffer size) to the user setting. This is done by continuously reducing the output rate in small steps over a long time, and midway through start increasing the output rate again so that both the nominal rate and the requested latency are reached again at the same time. During all this period the output rate and change of output rate will be within MPEG-2 and DVB specifications.

The Pro-MPEG specification has indicated tolerance limits for both short time jitter and long term latency variations.

QoS requirements for Transport StreamsThe requirements for signal quality for transport of MPEGvarious standards. The most important are:

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MPEG-2 System [4] • PCR Jitter tolerance ± 500 ns • PCR clock rate 27MHz ± 810 Hz (30ppm) • PCR rate of change of 75 x 10-3 Hz/s.

TV standards [6, 7, 8] • Subcarrier frequency stability 3 ppm (NTSC) 1 ppm (PAL B, G) 0.23 ppm (PAL I) • Subcarrier clock rate change 0.023 ppm/s (PAL I) • Horizontal sync jitter 2.5 ns (ITU 601)

SFN Network [10] • SFN Mega frame adapter and SFN modulators must be locked to same time and frequency source. The total delay between the SFN Mega frame adapter and modulator must be less than 1 second. Some modulators have limitations on variations of buffered data. For example one modulator does have a variable buffer of 4 kB. Such limits are implementation dependent “features”.

QoS improvements in the networkA modern IP network is far beyond a best effort network. IP networks deployed for videotraffic achieve broadcast quality of services by combining multiple technologies. • Multi Protocol Label Switching MPLS• Service prioritisation (Precedence bits or diff-serve)MPLS is an additional tag added to the IP packets at the input port and removed at the output port of the network. This tag is linked to a predefined routing setup and effectuates expedient forwarding of data with an allocated bandwidth. MPLS configuration is normally part of network traffic engineering. Service prioritisation secures that video payload packets are the last ones to be droppedin case of buffer overflow in a router. The T-VIPS TVG gateways can support both normal (TOS) Type of Service bit setting and diff serve bits setting. When TVG420 is used in VPN mode likewise it is possible to configure the V-LAN priority (CoS Bits).

QoS improvement in the TVG420 ASI to IP gatewayTVG420 have two means to remove the impairments caused by to IP network.• Forward Error Correction (FEC)• De-jitteringForward error correction is an additional measure in order to regenerate occasionallylost packets that may occur, and more importantly to regenerate burst packet losses that

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may occur occasionally. TVG gateways do support Pro-MPEG forward error correction. Variation in inter packet arrival time or jitter needs to removed by the IP to videogateway. A high quality de-jittering solution is included in TVG video gateways in order to deliver an output signal that is within the acceptable tolerances. This solution is built on several algorithms in order to lock very fast and later stabilise the ASI clock withminimal clock adjustments while at the same time aiming to have a constant delay in the receiver.

Clock recovery in TVG420IP transport overview.

Figure 4. IP over video network model.

A MPEG-2 source is forwarding a CBR Transport Stream to the TVG420 transmitter. The transmitter accumulates several TS packets and inserts them in RTP/UDP/IP packets and them out on the IP network. In addition, the transmitter does also compute Forward Error Correction (FEC) packets. These protects a number of IP packets and can be computed only when the last packets it is protected is sent. The TVG420 receives payload packets and FEC packets. The packets are buffered. Lost packets are regenerated by means of the FEC. The receiver measures the stream characteristics and applies the result in order to regenerate a TS stream nearly identical to the source stream.

Figure 5. IP over timing model.

The IP packets are sent from the TVG420 TX at the time marked as TX output. Some time later the packet enters the TVG420 RX input. The transport time (network delay = jitter) will vary as indicated by the packet delay distribution curve. The TVG420 receiver will remove the jitter introduced in the IP network.

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IP transport overview.

SourceMPEG-2TransportStream

RegeneratedMPEG-2TransportStream

Figure 4. IP over video network model.

Figure 5. IP network timing model.

Transmit ReceiveTVG420 TVG420TX RX

RX input

Packet delay distribution

TX output

Networkdelay

min average max time

IP transport overview.

SourceMPEG-2TransportStream

RegeneratedMPEG-2TransportStream

Figure 4. IP over video network model.

Figure 5. IP network timing model.

Transmit ReceiveTVG420 TVG420TX RX

RX input

Packet delay distribution

TX output

Networkdelay

min average max time

Figure 6. TVG420 IP to ASI receiver buffer management.

As explained in Figure 5, packets are arriving at the TVG420 exhibit variable delay and placed in the receiver buffer. After some time the packet is the removed from the buffer and delivered as ASI output. The goal is to deliver Transport Stream packets at a constant rate. Deviations from this goal will result in PCR jitter. At the same time is the receiver buffer adjusted continuously in order to keep a constant buffer filling. This will be obtained when the average values for input and output rates are equal. The total delay from the time the packets enters the TVG420 transmitter until it leaves the TVG420 receiver will be equal to the average network delay and the average latency. Note that long time delay changes as shown in Figure 3 will result in changes in the total delay while short time jitter will be removed by the TVG420.

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Figure 6. TVG420 IP to ASI receiver buffer management.

RX input

Packet delay distribution

Networkdelay

TX output

min average max time

Receiver latency ASI output

Jitter

margin FEC

Packets arriving too late

Total delay

10

Buffer regulation with IP rerouting.

Figure 7. Network Architecture

Figure 8. Buffer filling of TVG420

MPEG-2TransportStream

RegeneratedMPEG-2TransportStreamTVG420 TVG420Source Receive

TX RX

10msdelay

10msdelay

time

Bitrate at TVG420 receiver input

TVG420 receiver buffer filling

timeTVG420 rate of rate change

Buffer regulation with IP rerouting.

Figure 7. Network Architecture

Figure 8. Buffer filling of TVG420

MPEG-2TransportStream

RegeneratedMPEG-2TransportStreamTVG420 TVG420Source Receive

TX RX

10msdelay

10msdelay

time

Bitrate at TVG420 receiver input

TVG420 receiver buffer filling

timeTVG420 rate of rate change

Buffer regulation with IP rerouting.

Figure 7. Network Architecture

Consider a network as shown in the pervious figure. There are two paths between the transmitter and receiver, either a route with two hops or a route with three hops. The network delay for the three hop path in this case is 10ms. Normally the video stream will use the direct route but occasionally it will take the alternative route.

Figure 8. Buffer filling of TVG420

Initially packets are going the shortest path. At some time rerouting takes place. As this path is 10 ms longer no data will be received for this period as shown on the upper part of the figure. However TVG420 will continue to play out data at a constant rate. As the receiver buffer is now decreased with an amount corresponding to 10 ms of data, TVG420 will start to slowly reduce the rate in order to build up the buffer to the value entered by the operator. Some time later the main path is in operation again. During a 10 ms period IP packets will be arriving from both paths. The RTP header contains a packet sequence number and this is used to reorder packets in same order as they were sent. At this moment in time, however, the output rate is still lower

than the input rate and the buffer will continue to increase. As the buffer is now larger than the configured value, the bit rate will be increased. Since rate of rate change is kept small it will take some time before output rate is up to the same value as the input rate. At this point in time the buffer will reach the peak value. The bit rate must be increased further in order to reduce the buffer. The curves are made for explanation only and do not represent measured response. The major point with this figure is to show that changes in network delay will influence the receiver buffer.

AcronymsAAL-1 ATM Adaptation Layer.

ATM Asynchronous Transfer Mode. An ATM cell is 48 bytes

FEC Forward Error Correction

IP Internet Protocol

IPDV IP Packet Delay Variation

IPTD IP Packet Transfer Delay

ITU International Telecommunication Union. Organised by UN (United Nations)

PCR Program Clock Reference

Pro-MPEG Professional-MPEG Forum

References1. ITU-T Y.1541 Series Y: Global Information Infrastructure and Internet Protocol Aspects. Internet protocol aspects – Quality of service and network performance. Network performance objectives for IP-based services.2. ITU G.1050 Series G. Network model for evaluating multimedia transmission performance over internet protocol3. VSF forum: Proposed Draft of New Y.1541 Annex. SMPTE Motion Imaging Journal, July August 2005.4. Pro-MPEG Code of Practice 35. ISO/IEC 13818:2000-1 MPEG-2 Systems6. ITU-R BT.6017. ITU-R BT.6248. ETSI TR 101 2909. Tektronix. PCR Measurements Primer.10.ETSITS101191V1.4.1(2004-06)TechnicalSpecificationDigitalVideoBroadcasting (DVB); DVB mega-frame for Single Frequency Network (SFN) synchronization.

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About T-VIPSHeadquartered in Oslo, T-VIPS AS is a Norwegian technology company with new products and solutions for the growing professional Video over IP transport market. T-VIPS provides solutions for broadcast contribution, studio-to-studio media exchange, in-house signal distribution and routing, post-production, live event coverage and primary distribution. The company is funded through investments from the leading Scandinavian VC funds Northzone Ventures and Selvaag Venture Capital.

For further information, please visit: www.t-vips.com

www.t-vips.com • Tlf: +47 22 88 97 50 • Org: 981 885 236 • Nils Hansens vei 2, NO-0667 Oslo • Norway