3IntroductionWithin a TCP/IP model, it supports best-effort delivery services.QoS for VoIP was affected by factors such as delay, codec, and echo.We adopt Differentiated Services to minimize the factor of delay.An efficient echo canceling scheme, which could be hardware-implemented in hosts or software-implemented in soft PBXs, has been used.Comparisons of different codecs was listed.
5Topics Comparisons of Different QoS Polices Introduction to Differentiated ServicesThe DS ArchitectureDS Configuration and OperationPer-Hop Behavior
6Comparisons of Different QoS Polices IPv6IPv6-enabled routersResource ReSerVation Protocol(RSVP)RSVP-enabled routersScalability for a large networkMultiProtocol Label Switching(MPLS)MPLS-enabled routersRequire significant changesDifferentiated Services(DS)The minimal changes in existing routers
7Introduction to Differentiated Services Offer different levels of QoS to different traffic flowsUse the existing IPv4 Type of Service or IPv6 Traffic Class octetA service level agreement is established between the service provider and the customerProvide a built-in aggregation mechanismDS is the most widely accepted QoS mechanisms
8The DS ArchitectureUse six bits of ToS field to encode the DS codepoint(DSCP)Packets are labeled for service handling by means of the DSCPTo avoid situations where every packet is marked highest priority, boundary node must ensure itThe amount of change required in the core of network is minimal
11Within a domain, the interpretation of DSCP is uniform The interior nodes handle packets based on their DSCP valuesThe boundary nodes include traffic conditioning and classificationThe services provided across a DS domain are defined in a service level agreement
12Traffic Conditioning Functions ClassifierSeparate submitted packets into different classesMeterMeasure submitted traffic for conformanceMarkerRe-mark packet with a different DSCP as neededShaper/DropperDelay or Drop packets so that the traffic doesn’t exceed profile
13Per-Hop BehaviorPHB is the treatment that a DS router applies to a packet with a given DSCP valueExpedited Forwarding (RFC 3246)low-loss, low-delay, low-jitter, assured bandwidth, end-to-end services through DS domainsConfigure nodes so that the traffic aggregate has a well-defined minimum departure rateArrival at any node is always less than that node’s configured minimum departure rate
14Assured Forwarding (RFC 2597) The concept was first introduced in Clark, D. and is referred to as explicit allocationEnsure high-priority packets are forwarded with a greater degree of reliabilityProvide four classes and three precedence levelsDSCP of the form xxx000 are reserved to provide backward compatibility with IPv4 precedence service
17OverviewThe new world of Internet telephony is facing one of the same challenges that early long-distance calling did.Traditionally, problems with echo have been experienced on long-distance or international calls, particularly those involving satellite connections.
18The Cause of Echo Electrical echo due to imperfect impedance matching Electrical signals of all types always are reflected at line terminations, except when the load at the line end exactly matches the impedance rating of the line itself.Acoustic echo due to microphone pickup of audio outputFor instance, on a hands-free cell-phone call, the echo characteristics change as the speaker moves around.
19What is Impedance matching? Impedance matching is the practice of attempting to make the output impedance of a source equal to the input impedance of the load to which it is ultimately connected, usually in order to maximize the power transfer and minimize reflections from the load.
20The Cause of Echo (cont.) The major difference is electrical echo is a property of the line connection and remains mostly constant throughout the call, while acoustic echo varies in strength and delay depending on the changing acoustic environment of the echo source.
21Echo becomes a problem Why is echo not a problem on every call? The echo heard at the same time as the caller is speaking is considered as part of side tone. It becomes noticeable only when there exists a delay over long distance.Troublesome echo in VoIPIt is packetization and processing delays inherent in VoIP that cause existing echo to become a problem.
22What to Do about EchoThe use of attenuation to eliminate echo was not a satisfactory solution, and this method was abandoned when digital echo cancellation became available.Digital echo cancellation is based on subtracting from the received signal a correction based on the response of the system to a short spike of sound, called the finite impulse response (FIR).
23Finite Impulse Response 780The FIR is simply the echo you would hear from a short ping.
24Finite Impulse Response (cont.) 128 digital sound taps taken at a rate of 8,000 times per second, covering 128/8 = 16 milliseconds.This echo starts at tap 7, or about 1ms after the impulse.The echo from the impulse has an effect that lasts about 10ms(80 taps).
25Finite Impulse Response (cont.) Therefore, Echo cancellation should take place close to the echo source.If the echo were cancelled at the caller, there would be many more leading idle taps, so the true echo would be shifted back, perhaps right out of the tap sample.
26Finite Impulse Response (cont.) The echo does not appear to be very strong. But because of the sensitivity of human ears, it is completely intolerable on a VoIP system.The higher the number of taps, the higher the computing load and memory requirement.
27Echo CancellationThe FIR is used to calculate a series of correction factors that represent the echo component of the received signal.For example, on a 128-tap echo canceller, the echo is:Echo = (128 values of FIR)˙(128 previous tap samples of transmission)By subtracting this "echo" from the signal as received, a substantially echo-free receive signal is obtained.
28Echo Cancellation (cont.) However, because of rounding errors and non-linearities, some of the echo remains. The nonlinear processor cuts out the remaining received signal if the signal is small enough.
29Echo Cancellation (cont.) Obtaining the FIR is an iterative training process based on measuring the residual signal after the calculated echo has been subtracted and changing the FIR estimate. This process requires silence on the other end of the line-there is no doubletalk.The iterative FIR optimization converges quite slowly.
31Echo Cancellation in Soft PBX Environments Echo cancellation is a hugely CPU-intensive process.Software echo cancellation is one of the major factors limiting the performance of soft PBX systems.
32Echo Cancellation in Soft PBX Environments (cont.) One compromise made in the interest of saving CPU cycles is that the "learning" algorithms that update the FIR estimate are not run every time a voice sample is processed, but much less frequently. So the system trains slowly.That is the reason why you often hear quite considerable echo well into the conversation until the echo canceller trains and the echo decreases.
33Optimization of Echo Cancellations Today, all long-distance calls over 600km routinely are echo-cancelled at each end.On most VoIP-PSTN geteways, a great deal of echo cancellation is unnecessary and, in fact, detrimental to voice quality.If the call has no echo, echo cancellation can be disabled.
34Echo MeasuringEcho cancellation isn't necessary for incoming calls that already are echo-cancelled. An echo detector can be used to switch off echo cancellation for these calls.In software echo cancellers, the considerable CPU load that can be freed by echo detection is always immediately available to other processes, which in turn can increase the quality and capacity of the system significantly.
37Types of Speech CodecsWaveform CodecsSource CodecsHybrid Codecs
38Waveform CodecsSample and code the incoming analog signal without any thought as to how the signal was generated in the first placeWhen the signal is reconstructed, at least to a very good approximationProduce a high-quality output and are not very complexConsume large amounts of bandwidthWhen they are used at lower bandwidths, the speech quality degrades significantly
39Source Codecs Also known as vocoders They attempt to match the incoming signal to a mathematical model of how speech is producedOperate at low bit rates but tend to produce speech that sounds syntheticUsing higher bit rates does not offer much improvementUsed in private communications system and military applications
40Hybrid Codecs Provide the best of both worlds They utilize knowledge of how people produce soundsProvide quite good quality at lower bit rates than waveform codecs
42G.711 Waveform Codec The most commonplace coding technique used today Used in circuit-switched telephone network all over the worldPulse Code Modulation (PCM)MOS of about 4.364 Kbps bandwidth requirement
43G.721 Waveform Codec Adaptive Differential PCM (ADPCM) 32 Kbps bandwidth requirementMOS of about 4.0G.721 has now superseded by ITU-T recommendation G.726, which is more advanced ADPCM codec16, 24, 32, 40 bandwidth requirement (G.726)G.726 has an MOS of about 4.0 (when running at 32 Kbps)
44G.728 Hybrid Codec Low-Delay Code-Excited Linear Predictive (LD-CELP) 16 Kbps transmit bit rate0.625ms coder delayMOS of about 3.9Digital Processors (DSPs) used for G.728 are quite expensive
45G.731.1 Hybrid Codec 30ms coder delay Algebraic Code-Excited Linear Prediction (ACELP) / Multi-pulse Maximum Likelihood Quantization (MP-MLQ)5.3/6.3 Kbps bandwidth requirementMOS of about 3.8Digital Processors (DSPs) used for G are quite expensive
46G.729 Hybrid Codec CS-ACELP 8 Kbps Bandwidth requirement MOS of about 4.015ms coder delay
48ConclusionDifferentiated Services are suitable for prioritizing different traffic classes with minimal modifications.Echo on a telephone call is an annoying phenomenon that has been mostly under control in the classic telephony system, but it is rearing its head again as VoIP proliferates.
49ConclusionIts effective control is vitally important for the eventual success of VoIP technologies in general, because of the effect of echo on perceived quality.Although G.711 Used in circuit-switched telephone network, but it’s not very suitable for VoIPG.721, G.726, G.728, G.729 are more suitable for VoIP