high performance precision emulators · display the next frame in a sequence. mpeg compression...

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High PerformancePrecision Emulators

GEM Delay & Impairment Emulator

TRIPLE PLAY Streaming Video IPTV VoD Multicasting RTP RTCP MGCP MEGACO H.323

The Technology: VideoIP Video typically uses MPEG-2 encapsulated inReal Time Streaming Protocol (RTSP) and IP.Video is a set of individual frames played at a ratesufficient to be perceived as smooth motion, typi-cally 30-frames per second. However, transmit-ting 30 full frames per second requires morebandwidth than is practical on IP networks. TheMoving Pictures Experts Group (MPEG) devel-oped methods of reducing the required bandwidthby reducing the amount of information required todisplay the next frame in a sequence.

MPEG compression divides video into a set offrames called a group of pictures (GOP). A GOPcomprises three types of frames: I-frame (intra-frame), P-frame (predictive frame) and B-frame(bi-directional frame). An I-frame contains all theinformation required to display a frame.

A P-frame contains only changes from the pre-ceding I-frame – information on the parts of thepicture in motion, but no information on the partsof the picture that is static. The information con-tained in a P-frame is not sufficient to display afull image. It must be preceded by an I-frame.

A B-frame is similar to a P-frame, but containsinformation that has changed from the precedingframe or will change in the following frame.

In a GOP, I-frames are followed by P-frames andB-frames. The more I-frames, the better the videoquality. I-frames are two or three times larger.

A typical GOP will contain seven to ten frames inthe pattern IBBPBBPBBP, although the ordermay be adjusted before transmission. A GOP isencapsulated into one IP packet.

Overview

The delivery of voice, video and data over an Internet Protocol (IP) network, known as triple play, providesnew avenues for growth for service providers. The technology enables more intelligent services, such assimultaneous viewing/recording of multiple programs, targeted advertising, on-screen caller ID, and con-tent control. In addition providers have greater flexibility in bundling and marketing a range of services.

However, new opportunities bring new challenges. Delivering voice and video over IP requires time-sensitive content to be sent over a network that may introduce packet delay, loss and reordering with lim-ited bandwidth. A successful triple-play solution must provide quality of experience by compensating fornegative conditions that occur in IP networks. Anue Network Emulators allow test engineers to introducethose conditions in a controlled and repeatable fashion to predict quality of experience in actual networks.



The Technology: Voice

In addition to the Internet Protocol (IP), otherprotocols make VoIP possible. They can be di-vided into call setup and transport of the call it-self.

Transport Control Protocol (TCP) is the reliable-transport partner in the TCP/IP protocol suite. Asa result, it is used to encapsulate call-signalingprotocols to verify that the call is established.However, TCP has too much overhead to effi-ciently transport the actual call. User DatagramProtocol (UDP), a connectionless transport proto-col, is used to carry the voice packets.

During call setup, a TCP connection between thetwo endpoints is established. This can involve theuse of proxy, registration and redirect servers tolocate the IP address of the called party. Oncethe TCP session is in place, the ring (called party)and ring back (calling party) tones are triggeredand the parameters governing the call are negoti-ated. These can in-clude master/slavesettings for controllingthe session, choice ofcodec (coder-decoder)and compress ionscheme, the type ofcall (voice/video con-ference) and securitysettings, such as en-c r y p t i o n . C a l lsetup/signaling proto-c o l s i n c l u d eH.225/Q.913, H.245,H.323, SIP, SDP,MGCP and MEGACO.

Once the call is established, the conversation isencoded or digitized (converted from analog sig-nals to bits), compressed, and split into groups ofbytes to be encapsulated into payloads for trans

port across the network. It may also be en-crypted. Real-time Transport Protocol (RTP) isused to maintain packet sequence, since UDPdoesn’t have the sequencing support provided byTCP. Real-time Transport Control Protocol(RTCP) is the companion protocol that providesout-of-band control information for an RTP flow.RTCP can provide feedback on quality of service(QoS) for the flow to a VoIP application, whichcan adjust differentiated services (DiffServ) pa-rameters for the flow.



Testing Considerations

Triple play is a complex application requiring theseamless operation of many functions. Thetransport must deliver the required bandwidth atappropriate service levels. Voice conversationsmust be sampled, coded and decoded, com-pressed and decompressed, packetized and de-packetized, encrypted and decrypted. Signaling isrequired to establish voice connections and ne-gotiate settings. More signaling is required to se-lect video channels and Video on Demand pro-gramming. The system must be scalable to ac-commodate a sustainable customer base and theassociated stress of channel zapping during peaktimes. Most importantly, the system must be ableto deliver high voice and video quality over thedeployed infrastructure. And, all of this must bedelivered in the presence of data traffic.

In a Perfect World: The Performance Baseline.

The first step in testing a triple-play solution is toevaluate it in a test lab under optimum conditions.The infrastructure of the test lab should providean environment with no packet loss, no packetreorder and minimal delay. As all aspects of thesolution are tested, such as signaling capacity,voice capacity, voice quality (PESQ/PSQM MOSscores), channel zapping, video quality, QoSschemes and scalability, the maximum perform-ance limits of the system are determined. Theselimits establish the baseline, which will be usedfor comparison against the results of tests in amore realistic environment.

Real-World Testing: The Reality Check.

Triple-play services are not delivered on an idealnetwork, but with other traffic through multiple de-vices and across infrastructure that spans signifi-cant distances. The result is an environment thatpresents some level of delay, impairment andbandwidth limitation, the severity of which de-pends on how well the network is engineered.

To deliver triple play with confidence, test anddebugging iterations are performed under condi-tions in which the solution will actually be de-

ployed, and under worse conditions to account forequipment failure and disasters. Rather than teston a live network, which is expensive, impracticaland not repeatable, real-world testing is achievedby using network emulation in the test lab.

The tests performed during baselining are per-formed again with an Anue Network Emulatorcreating impairments such as packet loss, delay,bit errors, sequence errors or duplication in acontrolled fashion. The emulator can also modelthe effects of high traffic load by constrainingavailable bandwidth. The actual values used aredetermined by various methods.



Table 1: Determining Network Emulation Parameters

Educated guess

Trial and error

Values known to affect the service can be tweaked to determine quality-of-experience thresholds. For example, MDI and V-Factor use jitter, sequenceerrors and packet loss, among other metrics, to measure video quality. Valuesfor these impairments can be ramped up, both individually and jointly, to deter-mine the breaking point for video quality scores. Results can be used to char-acterize the limits of a device or service.

Pros: granular control over specific impairment metrics, goal-seeking of per-formance thresholds.

Cons: can be time-consuming if not automated; combinations tested may notreflect real-world conditions.

Network ManagementSystem or ProtocolAnalyzer

A network management system or protocol analyzer can be used to determinethe performance of the live network on which the solution will be deployed.Measured values for packet loss, delay, duplication and sequence errors can beused to configure the emulator during testing.

Pros: test conditions reflect the actual conditions measured on the productionnetwork.

Cons: conditions in IP networks are not static. They vary over time as eventssuch as traffic congestion, router flaps and topology changes occur. Testingwith a single value averaged over time will not produce an accurate reflection ofactual performance. Realistic testing requires impairments that change dynami-cally over time. Taking thousands of measurements over short intervals andautomating the changes in configuration of the network emulator can accom-plish this, but it can be complex and time consuming.

Standards-based model A standard network model for IP network impairment can be used. TIA-921 andITU-T G.1050 describe a Network Model for Evaluating Multimedia Transmis-sion Performance over Internet Protocol based on actual network data providedby anonymous IP service providers and IP network equipment manufacturers.An algorithm uses typical LAN, access and core network characteristics as in-put and produces as output dynamic values for end-to-end packet delay, lossand reorder that vary over time in the same manner as conditions in deployednetworks.

Pros: the model more accurately reflects the way actual IP networks behavethan other methods of testing.

Cons: since the impairments are changing dynamically, in the case of failure ofa test case, the impairment that caused the failure may not be immediatelyclear. Further troubleshooting may be required.



Before and After

Features that are susceptible to specific impair-ments are baselined and then tested in the pres-ence of the impairments known to affect them. Asissues are revealed, troubleshooting uncoversroot causes and facilitates the debugging of ap-plications or tuning of network parameters. Thisiterative process is required to assure robust so-lutions and minimize support costs.

Quality of Service. Real-time traffic is sensitiveto packet loss. While packet-loss concealment(PLC) algorithms can help mask the effect ofmissing packets, there is a threshold beyondwhich PLC is ineffective. For video, thedropping of a single MPEG I-frame isnoticeable. QoS is employed to guaranteedelivery of voice and video with minimumlatency. Different schemes may be useddepending on the implementation. Queuesfor real-time traffic must be able to deliverpackets with minimum loss, even in thepresence of other, lower priority traffic.Bandwidth oversubscription should be testedto verify that lower priority traffic is droppedto accommodate voice and video.

Impairment profiles can be configured based onwhere the QoS is implemented, such as in theexperimental bits in the MPLS header (core net-

work traffic engineering), the DSCP/ToS bits inthe IP header (aggregation-level QoS) or the802.1p-bits in the VLAN tag (DSLAM or PONQoS).

Voice quality. Voice quality can be affected bynetwork jitter, sequence errors, jitter discards(frames discarded due to arriving too late) andpacket loss. Implementations are tested againstmultiple loss profiles, from periodic and randomloss rates to burst loss events associated withtraffic congestion, route failure and route flapping.Acceptable quality thresholds are established foreach loss profile.

Video quality. Measures of quality of experience

for IP video can be categorized into those thatmeasure network statistics and those that meas-ure the content itself. Media Delivery Index (MDI),specified in IETF RFC 4445, predicts the qualityof packet video by measuring network statisticssuch as packet jitter, loss and sequence errors.V-Factor assesses the quality of packet video byinspecting the content of the MPEG frames them-selves. Both scores will be affected by emulationof network jitter, sequence errors, jitter discards(frames discarded due to arriving too late) andpacket loss. However, MDI will not detect corrup-tion of video content if that corruption doesn’t re-sult in a discarded packet. Using V-Factor with animpairment emulator that can modify payloadswith checksum correction or drop specific frames,such as I-Frames, can test for this kind of sce-nario.



Buffer size. Buffers are targeted specifically toaddressing issues of mis-ordered packets andjitter. Quality or performance scores from thebaseline tests should be compared to results asre-order and jitter are applied in a controlledfashion. As breaking points are established, ad-justments to buffering can be tested to knownvalues to discover optimum buffer design andconfiguration.

PLC Algorithms. PLC is targeted at packet loss.Implementations are tested against multiple lossprofiles, from periodic and random loss rates toburst loss events associated with traffic conges-tion, route failure and route flapping. Performancethresholds are established for each loss profile.

Channel changing. IP Video uses IGMPand multicast to efficiently transmit chan-nels to multiple destinations. IfJoin/Leave requests are corrupted ordropped, it will affect channel zapping asrequests are re-transmitted. The AnueGEM Network Emulator can be used totarget IGMP requests to measure theeffect of dropped requests.

Performance Thresholds. Whenbaselining performance, packet loss andsequence errors caused by the systemunder test may increase as thresholds are

reached. When testing with impairments, theamount of impairment injected should be com-pared to the impairment measured by the ana-lyzer to compare performance thresholds to thebaseline thresholds.

In addition, performance thresholds under realis-tic conditions are used to determine the servicelevel parameters required for a sustainable im-plementation. Testing with network emulation isused to right size the engineering of a productionnetwork, avoiding the needless expense of over-engineering, or the revenue-threatening conse-quences of under-engineering, the network.

Conclusion

A successful triple play solution requires testing under real-world conditions to guarantee that it can providequality of experience for the viewer by compensating for conditions that occur in IP networks. Using AnueNetwork Emulators enables engineers to predict application performance, thereby reducing risk, acceler-ating time to market, and allowing deployment with fewer problems and more confidence.

high performance precision emulators · display the next frame in a sequence. mpeg compression...

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