Motorola Canopy VoIP Whitepaper v4

8/3/2019 Motorola Canopy VoIP Whitepaper v4

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Motorola Canopy

Voice over IP over Canopy

September 27, 2004


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MotorolaVoice over IP over Canopy

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CIRCUIT SWITCHED VS.PACKET SWITCHED TELEPHONY ........................................................................... 6Circuit Switched: .................................................................................................................................... 6Packet-Switched Telephony:................................................................................................................... 6

CODER/DECODER (CODEC) ........................................................................................................................ 6INTEGRATION OF VOIP WITH THE PUBLIC SWITCHED TELEPHONE NETWORK............................................ 7THE ANATOMY OF A VOIPCALL................................................................................................................ 7

Encoding and Packetization ................................................................................................................... 7Transport................................................................................................................................................ 8Dejittering and Decoding ....................................................................................................................... 8

Mouth-to-ear delay and overall distortion .......... ........... .......... ........... .......... ........... ........... .......... ......... 8STANDARDS FOR MEASURING CALL QUALITY ........................................................................................... 9CONSIDERATIONS FOR NETWORK CAPACITY PLANNING ............................................................................ 9ERLANG TABLES ...................................................................................................................................... 10

Erlang to VoIP Bandwidth Calculation................................................................................................ 11

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CANOPY FRAME PACKETS ........................................................................................................................ 12Control slots ......................................................................................................................................... 12 Downlink and Uplink Acknowledgement slots...................................................................................... 12

HIGH PRIORITY/QUALITY OF SERVICE (QOS) ......................................................................................... 12

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ADVANCED CALL QUALITY MEASUREMENTS ............................................................................................ 13TESTS VOIP-ENABLED NETWORK EQUIPMENT .......................................................................................... 13EMULATES COMPLEX NETWORKS IN TEST LAB.......................................................................................... 13OPTIMIZES NETWORK DESIGN ................................................................................................................... 13

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CANOPY CONFIGURATION ........................................................................................................................ 14VONAGE CONFIGURATION........................................................................................................................ 14TEST SCENARIOS ...................................................................................................................................... 14TEST RESULTS .......................................................................................................................................... 15FINDINGS .................................................................................................................................................. 15RECOMMENDATIONS ................................................................................................................................ 15

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CANOPY CONFIGURATION ........................................................................................................................ 16IXIACHARIOT CONFIGURATION.............................................................................................................. 16TEST SCENARIOS ...................................................................................................................................... 17TEST RESULTS .......................................................................................................................................... 17FINDINGS .................................................................................................................................................. 17

Summary of Results .............................................................................................................................. 17Quality of Service ................................................................................................................................. 18Theoretical Modeling based on Actual Results .................................................................................... 19

RECOMMENDATIONS ................................................................................................................................ 21


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PHASE IITEST RESULTS ........................................................................................................................... 23Canopy set at 50% downlink / 50% uplink G.711a.......... ........... .......... ........... .......... ........... .......... .. 23

Canopy set at 50% downlink / 50% uplink G.711u.......... ........... .......... ........... .......... ........... .......... .. 23Canopy set at 50% downlink / 50% uplink G.726............................................................................. 24Canopy set at 50% downlink / 50% uplink G.729............................................................................. 24Canopy set at 50% downlink / 50% uplink G.723.1-ACELP............................................................. 24Canopy set at 50% downlink / 50% uplink G.723.1-MPMLQ........................................................... 25Canopy set at 75% downlink / 25% uplink G.711a.......... ........... .......... ........... .......... ........... .......... .. 25Canopy set at 75% downlink / 25% uplink G.711u.......... ........... .......... ........... .......... ........... .......... .. 26Canopy set at 75% downlink / 25% uplink G.726............................................................................. 26Canopy set at 75% downlink / 25% uplink G.729............................................................................. 27Canopy set at 75% downlink / 25% uplink G.723.1-ACELP and G.723.1-MPMLQ......................... 27

TEST RESULTS WITH CANOPY QOS .......................................................................................................... 28 High Priority = 1 Slot........................................................................................................................... 28High Priority = 2 Slots......................................................................................................................... 28

High Priority = 3 Slots......................................................................................................................... 29 ............................................................................................................ 31

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The Motorola Canopy solution is an excellent wireless broadband data transporter between acentral site access point (AP) and many end-point subscriber modules (SM). The solution isbeing well received world-wide by telecom providers, wireless internet service providers (WISPs),and enterprises where site-to-site wireless data connectivity is preferred over traditional wiredsolutions. It has also been observed that the Canopy product can be an effective transportmedium for real-time Voice over Internet Protocol (VoIP) technology. VoIP over Canopy is areality for early adopters because of the low latency, high bandwidth, reliability, and scalability ofCanopy. Based on this information, Motorola has been receiving an increased interest in usingCanopy as more than a general data mover for such applications as real-time VoIP solutions.Therefore, Motorola has retained an external third party consulting and testing organization, WestMonroe Partners, to validate the potential for VoIP over Canopy as well as to conduct testing tobenchmark performance and scalability.

To begin, the team focused on modeling and testing VoIP calls to confirm and benchmark theusability and performance over Canopy. A lab was built to simulate a test environment where anumber of elements were varied to determine optimal configuration and performance limitationsfor potential Motorola customers of the Canopy product.

The testing was conducted in two phases. The goal of the initial phase of VoIP testing overCanopy was to develop a theoretical model using the Vonage broadband product to simulate anactual VoIP phone call and document the results and performance. Vonage broadband phoneservice was used to test an actual VoIP call going through Canopy out through the Internet. Thecall was monitored to gather statistics on bandwidth usage so a theoretical model could becreated to determine how many VoIP calls a Canopy AP can support. The Canopy modules weresetup in a point to multipoint network configuration with one Access Point (AP) and two

Subscriber Modules (SM).

During the second phase the objective was to test scalability with more than just one VoIP calland determine where the theoretical breaking points of the Canopy network exist. To enablemultiple VoIP call testing without needing the interaction of actual people to test calls and judgecall quality, IXIAs Chariot product was used to perform traffic pattern analysis and load testing ofmultiple calls across a Canopy network. Canopy modules were setup in a point to multipointnetwork configuration with one Access Point (AP) and two Subscriber Modules (SM) in Phase II.Several settings and configurations were adjusted during the second phase to determine optimaldesign for performance and scalability. The types of settings that were adjusted during thesecond phase of testing to determine the Canopy network limitations and optimal configurationwere:

Bandwidth percent allocation between downlink and uplink connections

Number of calls per SMNumber of SMs per APCompression Algorithm (Codec)Quality of Service High Priority (on or off)Voice Traffic with and without data traffic

For the purposes of this paper Canopy 5.2 GHz radios were used and all SMs were set at adistance of within 2 miles of the AP. This testing was based on typical RF configurations withoutinterference.


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$%Taking into consideration the anticipated traffic patterns of the end users (predominately data orvoice or an equal mixture of both) the customer can configure the network optimally to maximizetheir return on investment of equipment and capital expense, while still providing a high quality ofservice and cost effective alternative solution to toll dialing to the end user.

Based on our findings, it has been calculated that in an all voice traffic network using the G.711codec (common VoIP codec), Canopy can support 28 - 33 simultaneous voice calls per AP (seePhase 2 Findings). The call volume can be distributed over a number of SMs, not to exceed 5calls per individual SM. If a wireless ISP is looking to support both voice and data traffic on thesame connection to the end user, and half of the bandwidth was allocated to data and half tovoice, then it is expected that approximately 14 to 16 simultaneous calls could be transmitted perAP. This would allow the voice calls to be transmitted at a high quality level while still allocatingsufficient bandwidth for data transmission.

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0,-.*Depending on a Canopy customers business plan for rate of oversubscription and acceptablelevel of blocks calls, the number of calls stated above will be adjusted up or down depending onthe expected traffic volume. To properly plan a Canopy network configuration with VoIP, Canopycustomers are strongly advised to use Erlang tables in creating a business plan to estimate theload on their network to be able to determine a specific oversubscription rate (acceptable amountof blocked calls). Please see the Voice Over IP Basics section of this paper for moreinformation on Erlang tables.


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Voice over IP (VoIP) permits the movement of voice traffic over Internet Protocol (IP)-basednetwork. IP is a standard for data transmission based on packet-switching technology. Voice isbroken into a series of packets at the transmitting end. The components are then reassembledand decoded at the receiving device.

Voice communications is both real time and mission-critical. Any delay can make a call prohibitiveand lead to an undesired poor quality of service. Packet loss can be caused by router congestionthat may lead to a loss of portions of words or sentences. Traffic can multiply as the number ofrouters is increased in the network leading to longer delays. Network jitter, where packets don'tarrive in sequence, can lead to unavoidable delays and poor quality of service.

Circuit Switched vs. Packet Switched Telephony

Circuit Switched:

Nearly all voice traffic is circuit switched and transmitted over a Public Switched TelephoneNetwork (PSTN). The speed with which voice is transmitted is an aggregate rate of 64 kbps. Adirect connection between two connection points provides a permanent 64 kbps link for theduration of the call. This link cannot be used for any other purpose during this time. PSTNprovides low latency (delay) and is bidirectional, allowing for a two-way or full-duplexconversation to take place. The main shortfalls of circuit switching are provided by the inflexibilityand inefficiency inherited in the network by requiring a dedicated connection each time.

Packet-Switched Telephony:

In a packet-switched network, data is broken down into packets, each with a destination address.When the packets are transmitted through the network, the addresses are read at each router, ornetwork switch, for forward routing. At the destination, the packets are reassembled and re-sequenced. Depending on congestion levels in the network, packets may take different routes ontheir way to the destination. Packet switching provides a virtual circuit connection and is generallyhalf-duplex. The main difference from the circuit-switched network is that there is no dedicatedconnection. This is a connectionless network, which allows network resources to be used veryefficiently as bandwidth can be shared between applications.

Coder/Decoder (Codec)

A voice coder is the device that converts an analog voice signal into a digital signal. The digitalsignal is also compressed to reduce bandwidth requirements. Using a hybrid coding techniquewith complex algorithms, the voice waveform is sampled and the speech parameters areextracted. Thus, in any predefined time period, the waveform is assembled by a synthesis

technique to closely assemble the original waveform. The best way to reduce latency is to changethe voice coding method; however, the trade-off is voice quality vs. bandwidth required. Whilethere is a delay in the voice compression methods used, there is little further delay withdecompression regardless of the algorithm used.


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Table 1

Compression Algorithms

Algorithm Description and Rates

G.711uPulse code modulation (PCM) specifies the initial analog-to-digital conversion ofspeech. Speech is transmitted at 64 kbps which is considered to be toll quality.ITU standard for H.323-compliant codecs and most frequently used in the USA.

G.711aSame as above, however it uses the A-law for companding, which is the mostfrequently used standard in Europe.

G.726A waveform coder that uses Adaptive Differential Pulse Code Modulation (ADPCM)at 32 kbps. ADPCM is a variation of PCM, which only sends the difference betweentwo adjacent samples, producing a lower bit rate.

G.729High-performing codec; offers compression with high quality. Algorithm runs at 8.4kbps with 10-ms delay and a compression ratio of 8-to-1.

G.723.1-MPMLQ

ITU algorithm that offers voice transmission with quality at a rate of 6.3 kbps with 30-ms delay. Uses the multi-pulse maximum likelihood quantization (MPMLQ)impression algorithm.

G.723.1-ACELP

ITU algorithm that offers voice transmission with quality at a rate of 5.3kbps with 30-ms delay. Uses the conjugate structure algebraic code excited linearpredictive compression (ACELP) algorithm.

Source: Gartner & IXIA

Integration of VoIP with the Public Switched Telephone Network

Despite some similarities, there are fundamental differences in the way signaling takes place in aPSTN and in packet networks based on IP. Signaling is essential to ensure call-related controlinformation necessary to establish, bill and terminate connections. To allow full convergence andseamless integration of functionality between PSTN and IP networks, signaling has to bedeveloped to avoid degradation of services. The signaling used in a PSTN is carried over adifferent physical network known as Signaling System 7 (SS7). SS7 messages are exchanged inthe form of data packets similar to the way IP networks transmit data, on a dedicated overlaynetwork used exclusively for signaling. By using this separate network for "out-of-band" signaling,this ensures that lines are clear and are free prior to setting up calls.

IP networks do not use the same type of out-of-band signaling. Instead, they use the protocolssuch as H.323 or SIPs that are not compatible with the PSTN. A new signaling architecture hasbeen required to integrate both networks seamlessly; however, this does not yet exist. A varietyof protocols have been proposed to meet PSTN-IP integration, and these include H.323, SIPIPS7 and Megaco to migrate interoperability and signaling issues. Nevertheless, these standardsare being improved to protect against packet loss and to allow for the transmission of toll-qualityvoice services. Until these are resolved, it is likely that Internet telephony traffic will be carried

over dedicated IP networks.

The Anatomy of a VoIP Call

Encoding and Packetization

In the first stage, the digitized voice signal (for example, in G.711 format) is encoded andpacketized. This operation can be performed either in the user terminal or in a gateway (forexample, between a PSTN and a packet-based network). In the latter case, it is assumed that thecircuit-switched transport of the voice signal from the user terminal to the gateway introduces only


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a negligible amount of delay and distortion. The packetization delay is defined as the timeneeded to collect all voice samples that end up in one packet. A new voice packet is produced atthe end of every interval of duration equal to the packetization delay. The choice of thepacketization delay involves a trade-off between effective bit rate and delay, because the payloadsize also scales linearly with the packetization delay

Transport

In the second stage, packet transport, the resulting flow of voice packets is transported over apacket-based network consisting of several (access and/or core) nodes. The network delay, i.e.,the delay incurred by transporting a voice packet over the network, can be split into two parts: adeterministic part referred to as the minimal network delay; and a stochastic part referred to asthe total queuing delay. The minimal network delay consists mainly of the propagation delay (5s/km), the sum of all serialization delays, and the route look-up delay. It is assumed that routeupdates are so infrequent that the probability of one occurring during a phone call is negligible.Hence, the minimal network delay is constant. The total queuing delay is the sum of the queuingdelays in all the nodes that are crossed. The queuing delay in one network node is due to thecompetition of several flows for the available resources in the queue of that node. The totalqueuing delay is responsible for the jitter introduced in the voice flow. That is, a flow of voicepackets that entered the network with constant inter-arrival times does not leave the network inthe same way, because some voice packets are delayed more than others. The jitter (alsoreferred to as packet delay variation) is defined as the delay difference between the fastestpacket and the slowest packet (though one still considered as on time). Note that duringtransport over the network, a fraction of the packets may get lost, either due to overflowingqueues or due to the erroneous transport over unreliable links.

Dejittering and Decoding

Dejittering is absolutely necessary. Since the decoder needs the packets at a constant rate, thejittered packet flow is dejittered and decoded in the third stage. Dejittering a voice flow consistsof retaining the fast packets in the dejittering buffer to allow the slow ones to catch up. The fastpackets are the ones that do not have to queue in any of the nodes. As voice can tolerate somepacket loss, it is not mandatory that the dejittering mechanism wait for the slowest packet. A(small) fraction of the packets may be considered as arriving too late. This fraction of overduepackets results in what is called a dejittering loss. If the dejittering buffer were to know if the firstpacket that arrived was a slow or a fast one, it could compensate precisely for its queuing delay.Since this knowledge is generally not available, the dejittering mechanism can either assume theworst case (i.e., assume that the first packet to arrive was the fastest possible) or try to graduallylearn how the delay of the first packets relates to the delay of consecutive packets.

Mouth-to-ear delay and overall distortion

The individual contributions of the above stages (Encoding and Packetization, Transport,Dejittering and Decoding) combine to give the one-way mouth-to-ear delay and the overall packetloss. The mouth-to-ear delay of a packetized phone call (in one direction) is made up of fourcomponents:

1. The packetization delay.2. The DSP delay, which includes the delay due to encoding, decoding, look-ahead, and

echo control. The lower limit of this delay is the sum of all look-aheads: even iftechnology continues to evolve to culminate in DSPs with dazzling processing power, thelook-aheads remain unaffected.

3. The total minimal network delay, which is the delay of the fastest possible packet. Thetotal propagation delay is the lower limit of the minimal network delay.


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4. The dejittering delay, which is the delay introduced by the adaptive dejittering mechanismon the fastest possible packet to compensate for the total queuing delay encountered bythe slowest packet.

Overall distortion stems from encoding the voice signal and packet loss in the network or in thedejittering buffer.

Standards for Measuring Call Quality

Call quality measurement has traditionally been subjective: picking up a telephone and listeningto the quality of the voice. The leading subjective measurement of voice quality is the MOS (meanopinion score) as described in the ITU (International Telecommunications Union)recommendation.

In voice communications, particularly Internet telephony, the mean opinion score (MOS) providesa numerical measure of the quality of human speech at the destination end of the circuit. Thescheme uses subjective tests (opinionated scores) that are mathematically averaged to obtain a

quantitative indicator of the system performance.

Compressor/decompressor (codec) systems and digital signal processing (DSP) are commonlyused in voice communications because they conserve bandwidth. But they also degrade voicefidelity. The best codecs provide the most bandwidth conservation while producing the leastdegradation of the signal. Bandwidth can be measured using laboratory instruments, but voicequality requires human interpretation.

To determine MOS, a number of listeners rate the quality of test sentences read aloud over thecommunications circuit by male and female speakers. A listener gives each sentence a rating asfollows: (1) bad; (2) poor; (3) fair; (4) good; (5) excellent. The MOS is the arithmetic mean of allthe individual scores, and can range from 1 (worst) to 5 (best).

Table 2

Mean Opinion Score

(lower limit)User Satisfaction

4.34 Very satisfied

4.03 Satisfied

3.60 Some users dissatisfied

3.10 Many users dissatisfied

2.58 Nearly all users dissatisfied

The E-model is a complex formula; the output of an E-model calculation is a single score, calledan R factor, derived from delays and equipment impairment factors. Once an R factor is

obtained, it can be mapped to an estimated MOS. R factor values range from 100 (excellent)down to 0 (poor). An estimated MOS can be directly calculated from the E models R factor.

Considerations for Network Capacity Planning

When a Canopy provider (WISP) is building a business plan they must take into account,oversubscription rate, number of acceptable blocked calls, and quality of service to customers.All of these factors are important and must be considered before providing or advertising VoIPservice. The WISP will have to make a choice on whether or not they will provide VoIP servicethrough call manager equipment (i.e. Cisco, Nortel, etc.) or if customers will acquire service on


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their own (i.e. Vonage, AT&T CallVantage, etc.) and leverage Canopy for broadband accesswhere the third-party CPE (customer premise equipment) device does the VoIP encapsulation.

As in any other bandwidth based network there is going to come a point when the network is atcapacity. When this happens, there are a few options on how to handle it depending on thechoice of service. In the case of VoIP service being provided through a call manager, the WISPhas a couple of options. The WISP can either continue to allow calls to be added to the networkand allocate less and less bandwidth per call or block the last call that pushes the network overits capacity. If the number of calls on the network is allowed to increase without limit, this willdegrade the call quality of all calls on the network because less bandwidth will be available percall. This is not the recommended approach because this will cause an overall low quality ofservice resulting in jittering phone connections to many customers. However, if the last call is justblocked and given a network is busy signal, then this user can just try redialing in a few secondswhen some capacity will most likely have become available and all other calls will not be affected.Managing the bandwidth in this fashion will allow a higher level of service to be provided andmanaged across the network.

If the WISP is going to allow its customers to purchase a Vonage type service on their own andutilize Canopy for broadband bandwidth, this is a little more difficult to manage. In this scenario,the WISP will not be able to differentiate between regular data packets and voice packets that areencapsulated as data. They will only have the option to manage overall bandwidth consumption.A WISP should take this into consideration when forming their business plan in this scenario.Since the WISP will have less control over voice call quality on an individual call level, they maywant to be more conservative in the service guarantees that they state for their customers aroundvoice quality.

To assist WISPs in this type of capacity planning it is recommended that they take intoconsideration Erlang tables which are discussed in the next section.

Erlang Tables

An Erlangis a unit of telecommunications traffic measurement. Strictly speaking, an Erlangrepresents the continuous use of one voice path. In practice, it is used to describe the total trafficvolume of one hour. For example, if a group of users made 30 calls in one hour, and each callhad an average call duration of 5 minutes, then the number of Erlangs this represents is workedout as follows:

Minutes of traffic in the hour = number of calls x duration = 30 x 5 = 150

Hours of traffic in the hour = 150 / 60

Hours of traffic in the hour = 2.5

Traffic figure = 2.5 Erlangs

Erlang traffic measurements are made in order to help telecommunications network designersunderstand traffic patterns within their voice networks. This is essential if they are to successfullydesign their network topology. Erlang traffic measurements or estimates can be used to work outhow many lines are required between a telephone system and a central office, or in the case ofCanopy, given a level of available bandwidth, determine the acceptable amount of blocked callsbetween a SM and an AP. Several traffic models exist which share their name with the Erlangunit of traffic. They are formulas which can be used to estimate the number of lines required in anetwork.


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The main Erlang traffic models are listed below:

Erlang B

This is the most commonly used traffic model and is used to work out how many linesare required if the traffic figure (in Erlangs) during the busiest hour is known. Themodel assumes that all blocked calls are immediately cleared.

Extended Erlang BThis model is similar to Erlang B, but takes into account that a percentage of calls areimmediately represented to the system if they encounter blocking (a busy signal).The retry percentage can be specified.

Erlang CThis model assumes that all blocked calls stay in the system until they can behandled. This model can be applied to the design of call center staffingarrangements where, if calls cannot be immediately answered, they enter a queue.

Erlang to VoIP Bandwidth Calculation

As explained above, the concepts of Erlang tables can be applied in a number of different ways toa voice telecommunications network. In the context of a Canopy network, the users will beapplying these concepts to VoIP applications. There are a number of calculators that areavailable to assist Canopy users in developing an appropriate business plan for their network.Below is an example calculation of Erlangs capacity for a VoIP network using a Erlang to VoIPBandwidth Calculator

7:

Given That:

G.711 Codec (64 kbps) is being used with a 20 ms packet durationThe Acceptable amount of blocked calls is 1 out of 100 (.01)

Available bandwidth is 3.1 Mbps (3250.59 kbps)

Then the network can handle a level of 28.1 Erlangs. This means that in its busiest hour thenetwork can handle 28.1 hours worth of total traffic volume. This volume can be distributedamong a number of different users, which should be taken into consideration in developing anappropriate business plan.


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Canopy Frame Packets

There are 33 data slots in a basic canopy frame and each slot has a 64 byte payload. However,this number will vary based on the number of control slots selected. Each frame has three controlslots by default and can be increased at the cost of data slots. Every two control slots allocateddecreases available data slots by one. There are also three downlink and three uplinkacknowledgement slots.

Control slots

The control slots are contention slots for the SMs to request to transmit data to the AP. Frameduration is 2.5ms (400 frames/second). The SMs can only transmit data when they have beengranted a control slot by the AP.

Downlink and Uplink Acknowledgement slots

There are three downlink and three uplink acknowledgement slots in a canopy frame. Thepurpose of the acknowledgement slots is to confirm a successful receipt of an upload ordownload to the AP. The amount of acknowledgement slots can be altered in the APconfiguration, but is not suggested unless turning on QoS.

High Priority / Quality of Service (QoS)

The purpose of the high priority channel is to handle traffic with a small tolerance for latency, suchas voice. If the Type of Service (TOS) bit is set for high priority the AP will prioritize this traffic inthe queue and hold back any data that is not designated as such. The high priority designation isa static allocation, meaning that when a number of slots are reserved for high priority they canonly be used for this purpose. If no high priority traffic is being passed the designated high

priority bandwidth will remain idle and unavailable for other traffic.


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To determine values such as MOS and R-factor, it is not feasible to have human listeners tomake these subjective judgments at all times. For the purposes of these tests and this paper,IXIAs Chariot software product was used to determine these values and compile the datanecessary. Chariot has the capability to provide a tremendous amount of data in a testingenvironment. The following is an example of the types of information that can be gathered. Forthe purposes of this whitepaper, the focus was to use the advanced call quality measurements todetermine how VoIP traffic performs on the Canopy network.

Advanced call quality measurements

Predicts call quality by calculating a MOS based on the industry standard E-model specified in

the ITU recommendation G.107. Improving on the base standard, the VoIP Test Module takesinto account additional network factors, such as jitter and consecutive lost datagrams, which canseverely impact overall call quality

Tests VoIP-enabled network equipment

Examines the effectiveness and performance of VoIP-enabled network equipment. The VoIP TestModule enables the user to verify that prioritization techniques work as planned with a mixture oftraffic and measure the performance impact of other network elements, such as VPNs, on delay-sensitive VoIP traffic. Enables the user to test the limits of the network by generating up to10,000 VoIP sessions. By identifying the point where call quality begins to suffer, the VoIP TestModule empowers the user to make informed decisions about the implementation and expansionof VoIP in the network.

Emulates complex networks in test labAllows the user to emulate complex networks with a mixture of both VoIP and non-VoIP traffic byusing Chariot and its VoIP Test Module. By using Chariot in the lab environment, the user canstress test network equipment, test network changes before deployment or replicate end-userenvironments and reported problems. Chariot evaluates the effectiveness of QoS. The user canensure that voice traffic is receiving necessary resources at the proper time without starving otherbusiness-critical applications.

Optimizes network design

Supplies on-demand testing for tuning network to minimize delay, jitter and lost data.


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In the initial phase of VoIP testing over Canopy, a Vonage broadband phone was used to test anactual VoIP call going through Canopy out through the Internet. The call was monitored to gatherstatistics on bandwidth usage so a theoretical model could be created to determine how manyVoIP calls a Canopy AP can support.

Canopy Configuration

In this phase, the Canopy modules were setup in a point to multipoint network configuration withone Access Point (AP) and two Subscriber Modules (SM). All Canopy modules were 5.2 GHzwith uplink and downlink percentages at 50%. During this test, Quality of Service (QoS) was notset as it would not have any affect on one VoIP call with no data traffic in the throughput of thecall.

The AP was connected to the Internet via a DMZ and router. We used a Vonage VT1000 (madeby Motorola) to conduct actual VoIP calls via the Internet. While conducting the VoIP calls,Compuwares Application Vantage tool was used to monitor the calls and gather call statistics(throughput, frames, etc.).

Hub

Internet

CompuwareTool

AP RouterDMZSM

Vonage/Motorola

Analog Phone

SM

Vonage Configuration

The Vonage VT1000 can be used with three different codecs:

G.711 (listed at 90 kbps by Vonage)G.726 (listed at 50 kbps by Vonage)G.729 (listed at 30 kbps by Vonage)

All three codecs were used during the tests to determine theoretical call capacity results for the

AP depending on the codec being tested.

Test Scenarios

Three calls were completed from the Vonage VT1000 for each codec. Each call was recordedand monitored with the Compuwares Application Vantage tool.


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Test Results

The results below were gathered using Compuwares Application Vantage tool. The Theoreticalmaximum VoIP calls per AP is based on Motorolas Canopy documentation that the maximum

uplink throughput using more than one SM is 3.1 Mbps (3250.59 kbps). Since a VoIP call is full-duplex and approximately the same amount of throughput will go upstream as well asdownstream, the amount of VoIP calls Canopy can support is limited by the lower throughputlimitation of the AP. When the AP is configured with two or more SMs on the network, theCanopy AP is documented as having up to 3.1 Mbps (3250.59 kbps) upstream, so 3.1 Mbps(3250.59 kbps) was used as the theoretical maximum.

Table 3

Vonage call(kbps)

Duration(seconds)

Ave.Bandwidth

(kbps)

AP Max*(kbps)

Theoretical maxVoIP calls per AP

FrameSize

(bytes)

Framesper

second

G.711 (90) 94.63 162 3250.59 20.1 214 96

G.726 (50) 105.04 99 3250.59 32.8 134 93

G.729 (30) 76.96 58 3250.59 56 74 99* Upstream maximum of AP is approximately 3.1 Mbps (3250.59 kbps)

Findings

The Access Point should be able to handle anywhere from 20 to 56 calls depending on the codecused and if all the traffic is VoIP (no data). The Access Point can handle more VoIP calls whenusing a codec that uses lower compression.

The Canopy system was fairly easy to setup and assemble and integrates well with the Vonageproduct. There were no special configurations needed to setup the Vonage VoIP product to worksuccessfully on the Canopy network.

Recommendations

The following recommendations are based on the output and statistics determined during theabove testing:

When configuring a Canopy network that is primarily for VoIP traffic, the AP shouldbe set at 50% for the ratio of uplink to downlink traffic. Since the uplink throughput isthe constraint for VoIP traffic on a Canopy network, the user needs to have the uplinkthroughput as close to the 3.1 Mbps (3250.59 kbps) maximum.In a primarily data network, it is likely that the downlink to uplink percentage will beset at approximately 75% to accommodate for most subscribers doing moredownload-type traffic. If VoIP is added to a saturated data network, customers couldexperience choppy and jittery VoIP calls. A WISP will need to carefully plan and testbefore adding VoIP as the throughput to an SM has many variables (distancebetween AP and SM, uplink/downlink percentage, number of SMs on the network,

etc.).


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,#%%)%+$%+,(%

In the second phase, the objective was to test scalability with more than just one VoIP call. Toenable multiple VoIP call testing without needing the interaction of actual people to test calls andjudge the call quality, IXIAs Chariot product was used to perform traffic pattern analysis and loadtesting of multiple calls across a Canopy network.

Canopy Configuration

In this phase, Canopy modules were setup in a point to multipoint network configuration with oneAccess Point (AP) and two Subscriber Modules (SM). Tests were performed using both one andtwo SMs, but two SMs were always registered with the AP. All Canopy modules were 5.2 GHzwith the following variables:

Uplink / Downlink percentageo 50% / 50%o 75% / 25%

Quality of Service (QoS)o Tests were run with QoS on and offo High priority percentage was set at 25%.o Acknowledgement and Control slots:

High priority uplink acknowledgement slots, high priority downlinkacknowledgement slots, and high priority control slots were set at 3.

Total uplink acknowledgement slots, total downlinkacknowledgement slots, and total control slots were set at 6.

Switch

AP Switch

SM

IXIA Endpoint

Hub

IXIA EndPoint

SM

IXIA EndPoint

IXIA Chariot Monitor

IXIA Chariot ConfigurationThe Chariot tool consists of two programs: the IXIA Chariot Console and the IXIA Endpoints.The console station is used to setup and initiate all tests. The tests can consist of VoIP traffic,multiple types of data traffic, or both VoIP and data traffic. The Endpoints are used to accept andreceive the data defined in the test scenario. The tool allows VoIP or data traffic to be generatedand sent to different Endpoints to simulate real-world traffic over a network. In this scenario, anEndpoint was put on the end of each SM and the AP with the Chariot console connected via ahub at an SM to be able to initiate and monitor all tests.


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Test Scenarios

The testing environment was a combination of one or two SMs and one AP. Additionally, thereare a number of variables that can influence the overall performance of the voice/data network.

The variables that were adjusted during testing included:

Bandwidth percent allocation between downlink and uplink connectionso The downlink and uplink percentage allocation was adjusted to determine

optimal performance and limitations of the equipment. The two scenariosthat were tested were 50% downlink and 50% uplink as well as 75%downlink and 25% Uplink.

Number of calls per SMo The call volume was adjusted from one call up to as many as 25 calls per

SM. The purpose of adjusting this variable was to determine the theoreticalmaximum capacity of each SM and AP.

Number of SMs per APo This was adjusted to test the limitations and performance of the AP under

different conditions.Compression Algorithm (Codec)

o As a result of altering the compression, the codec affects the possible callvolume and call quality.

Quality of Service High Priority (on or off)o The high priority bit in the Canopy product is simply a Type of Service (TOS)

bit that the user can select to either have on or off. In an all data world thehigh priority bit designation is not as critical given that a slight delay on thetransmission of data traffic is not nearly as meaningful to networkperformance as a delay on a voice packet. A delay to a voice packet canresult in an audible flaw in quality as jitter and lost information. The highpriority bit was applied in different test scenarios to demonstrate the impact itwould have on transmitting voice traffic more cleanly and managing datatraffic accordingly.

Voice traffic with and without data traffico To properly test the impact of having a high priority bit designated this

scenario was tested in conjunction with data traffic passing at the same time.

Test Results

For a summary of the test results from Phase II, please see the Findings section below. Fordetailed test results broken down codec and variable scenario please see Appendix A.

Findings

Summary of Results

After collectively looking at the testing results for the different codecs with different variables it ismore apparent which performed the best with the most robust call quality. The summary below islooking at the tests with two SMs. The detailed information on the testing results with one SM isincluded in Appendix A., however, for the purposes of analysis and theoretical modeling theresults from two SMs were used as they gave more real world results. One of the other maindistinctions to note in the data is the difference in performance of 50/50 downlink/uplinkpercentage versus 75/25. The uplink connection is the limiting factor in the call quality andbandwidth throughput because the SMs have less bandwidth capacity on the uplink. Taking thisinto consideration along with the amount of percentage that is allocated for downlink/uplink traffic,


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some conclusions can be drawn about which scenarios would work best for each networkconfiguration. For the most robust codecs five calls per SM appeared to be the breaking pointafter which point the call quality dropped off considerably.

Given that the Canopy product is currently used primarily as a data mover, one would want totake into consideration the performance of the Canopy under these different configurations.Noting that the downlink/uplink allocation is a static setting, the user wants to make sure that theyare not leaving more bandwidth than necessary in an idle position. When determining how toallocate the data it is worth while to take a look at the differences in performance below.

In a 50/50 configuration one can clearly push more calls through with a higher rate of quality(higher MOS). If the anticipation is that the network will have predominately voice traffic, the userwill want to get as close to the 50/50 configuration as possible to raise the amount of possibleVoIP calls that Canopy can support. In a 75/25 configuration where users will still be doing somedownload of data traffic, the user can still expect acceptable call quality with a significant volumeof calls being pushed through the Canopy network.

Actual Results with two SMs

Table 4

CodecDownlink/Uplink

PercentageNumber of Callsfrom Each SM

Total # ofCalls

MOS

G.711a 50/50 5 10 4.36

G.711a 75/25 4 8 4.13G.711u 50/50 5 10 4.37

G.711u 75/25 4 8 4.32G.726 50/50 5 10 4.16

G.726 75/25 5 10 4.02G.729 50/50 5 10 3.93

G.729 75/25 5 10 4.01G.723.1 ACELP 50/50 8 16 3.60G.723.1 MPMLQ 50/50 8 16 3.80

Quality of Service

The high priority setting on the Canopy product is strictly a bit designation indicating that thetraffic with that bit set has high priority and goes before any other data. Several differentconfigurations were tested to determine the robustness of the QoS setting. One, two and threehigh priority slots were allocated during testing. The results showed that with the 75/25downlink/uplink configuration, the best performance was with two high priority slots set with twoSMs. The high priority slots are set in the Canopy configuration table by indicating whichpercentage of available slots the user would like to set as high priority. To achieve two slots, thehigh priority percentage was set to 25% (to achieve two slots, a customer may need to trymultiple percentage configurations as the slots allocated will be dependent on the specificCanopy setup). Tests in this configuration yielded a very good MOS value of 4.33.

As part of the QoS testing, different size data packets were sent with the voice traffic to determineif there would be any negative impact on the quality of the voice call. These tests weresuccessful with varying data size files being passed. Starting with a file size of 128 kb andgradually moving up to a file size of 1.2 Mb added to the network during voice testing, calls werestill transmitted with a MOS value of approximately 4.3. This demonstrates that the Canopy QoS


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performs as desired as the latency and delay sensitive voice traffic is given priority and is able toperform with acceptable quality no matter how much data is being passed at the same time.

Theoretical Modeling based on Actual Results

After looking at the actual results from testing and knowing the maximum throughput of the AP, itis possible to extrapolate what the system performance would be like under expanded conditions.

The capacity of the AP for an uplink connection with multiple SMs is 3.1 Mbps (3250.59 kbps) ofdata. The theoretical bandwidth compression of each codec is also known and listed in the tablebelow. The actual bandwidth compression can be calculated by taking the rate at which the datawas sent for a particular call and multiplying that by the frame size.. The rate is calculated bytaking the total amount of data sent over one call test and dividing that by the call duration. Forexample, the G.711 codec sent 7,500 packets of data during a 150 second call. The rate is 50packets/sec. The rate for G.726 and G.729 was identical to G.711, 50 packets/sec. G.723.1MPMLQ and ACELP had a rate of 33 packets/sec.

It is also known that a Canopy frame is 64 bytes. From the testing done with the Ixia Chariotsoftware, the frame size used for all of the codecs is also known. Theframe size for each codec,including overhead, is listed in the table below. By knowing the total frame size and the rate oftransmission, the actual compression rate can be calculated. By continuing the example of theG.711 codec, 50 packets/sec (rate of transmission) times 238 bytes (frame size) equals 11,900Bps. This now should be converted from Bytes/sec to Bits/sec by multiplying the value by 8,which equals 95 kbps.

When the size of the frame packet of the codec is larger than 64 bytes, 4 bytes for overhead mustbe added for routing purposes to each packet before the packet is fragmented into 64 bytepackets. Using this information as a benchmark for a typical VoIP application, the efficiency ofdata transfer on the network can be calculated. By taking into account all of this information amore accurate theoretical model can be constructed. Given all of this information, the theoreticaltotal number of simultaneous calls that the AP can support can be calculated and is listed in the

below table.

Looking at the G.711 codec as an example, the flow of the calculation would be as follows:

Given that there is approximately 3.1 Mbps (3250.59 kbps) of total bandwidthavailable and the actual data compression rate is 95 kbps, the theoretical calculationwould simply be 3250.59 Kbps / 95kbps = 34 simultaneous calls.Since the total frame size is 238 bytes and the Canopy equipment transmits in 64byte packets, 4 Canopy frames (256 bytes) would be required to handle the totalbandwidth from the G.711 codec. To account for overhead, 4 bytes must be addedto each fragmented frame. In this case 4 frames are required, so 16 bytes total mustbe added to the 238 byte frame. This produces an average transmission efficiencyrate of 99% overall (254 bytes/256 bytes). In a transmission where each 64 byte

Canopy packet was being completely utilized (the application produced packets in 64bytes frames), the network would be operating under perfect conditions.The theoretical 34 calls can now be used in conjunction with the efficiency rate tocalculate the adjusted theoretical number of calls the network can handle. Thisresults in 34 calls times 99% efficiency, which is equal to 33 calls total.This same calculation can be made for each codec using the corresponding numberswith each different compression algorithm.


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Table 5

CodecListed

Compression

ActualCompressionThroughput

TotalFrameSize

(bytes)

TheoreticalSimultaneousCalls an APCan Support

TransmissionEfficiency

Rate

TheoreticalSimultaneousCalls an APCan Support

FactoringEfficiency

Rate

G.711a&

G.711u64 kbps 95 kbps 238 34 99% 33

G.726 32 kbps 55 kbps 138 59 78% 46

G.729 8.4 kbps 39 kbps 98 83 83% 68

G.723.1-

MPMLQ6.3 kbps 26 kbps 98 125 83% 103

G.723.1-

ACELP5.3 kbps 27 kbps 102 120 86% 103

*Upstream maximum of AP is approximately 3.1 Mbps (3250.59 kbps)

The test scenarios that were conducted with one and two SMs demonstrated a clearunderstanding of the capabilities of individual SMs and the call volume limitations. These resultsare documented by codec in Appendix A. These tests demonstrated that by utilizing differentcodecs the SM will have a varying breaking point of when call quality becomes unacceptable.For one SM, this point was approximately five calls depending on the codec.

The AP also has a breaking point at the number of simultaneous calls it can support withacceptable call quality with the uplink being the limitation. The initial thought would be that theAP has a total bandwidth limitation and the theoretical maximum should be approximately thesame no matter whether the calls are coming from one SM or multiple SMs. However, it is knownthat the performance of the AP does scale with an increased number of SMs. An example of thisis shown by looking at the performance of the G.711 codec when comparing one SM to two SMs.With the G.711 codec and one SM, the total number of calls that the AP could handle at anacceptable quality level was five calls. Similarly looking at the performance with two SMs it isshown that the AP can now handle a total of ten calls between the two SMs. These results showthat the AP does scale as SMs are added and suggests that as more SMs are added, the AP willbe able to support the theoretical numbers listed in the table above.

It is important to note that these results are in a voice only environment. As the amount of data

transferred on the upstream increases, the amount of bandwidth allocated to voice traffic willdecrease. The way to guarantee that the same bandwidth is consistently allocated to voice trafficis through the high priority classification under the quality of service option.

Taking the actual data from the test results and comparing it against the theoretical calculationsof what an AP can handle, the maximum number of calls to AP can be estimated and evaluatedper codec. Knowing that the G.711a, G.711u, G.726 and G.729 codecs have produced the mostreliable results during testing, those codecs are evaluated below. The test results (low MOSscores) from the G.723.1-ACELP and G.723.1-MPMLQ codecs suggest that these codecs do not


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transfer well over a VoIP network due to the low data compression (kbps). It is recommended tonot use these codecs for any VoIP applications independent of the network being used.

G.711a & G.711uBy looking at the column Theoretical Simultaneous Calls an AP Can Support Factoring EfficiencyRate from the above table one can see that when using the G.711a and G.711u codecs the APcan theoretically handle approximately 33 calls at 95 kbps per call. This call volume can begenerated using as few as 7 SMs or as many as 33 SMs depending on the call volume per SM.

Since it has been shown that the APs robustness scales with increased traffic it is expected thatit should be able to actually handle the theoretical value of 33 calls. Given the high MOS scorevalue with five calls from each SM (4.37), it is expected that the AP will still be able maintain aquality level that is considered satisfactory as it scales to 33 calls.

G.726By looking at the column Theoretical Simultaneous Calls an AP Can Support Factoring EfficiencyRate from the above table one can see that when using the G.726 codec the AP cantheoretically handle approximately 46 calls at 55 kbps per call. This call volume can begenerated using as few as 10 SMs or as many as 46 SMs, depending on the call volume per SM.

Although theoretically the G.726 codec appears like it can handle a large volume of simultaneouscalls, by looking at the MOS score at a low call volume it is the expectation that as the networkscales any additional degradation in the MOS score would make the call quality unacceptable.Using two SMs and five calls from each SM, the MOS score was 4.16, which is considered agood value. However, when the network scales to as many as 46 simultaneous calls per AP it isexpected that the MOS score will continue to degrade to an unacceptable level and may not beable to reach the theoretical maximum.

G.729By looking at the column Theoretical Simultaneous Calls an AP Can Support Factoring EfficiencyRate from the above table one can see that when using the G.729 codec the AP cantheoretically handle approximately 68 calls at 39 kbps per call. This call volume can begenerated using as few as 13 SMs or as many as 68 SMs depending on the call volume per SM.

Similarly to the G.726 codec, theoretically the G.729 codec appears like it can handle a large callvolume of simultaneous calls. However, looking at the MOS score at a low call volume it is theexpectation that as the network scales any additional degradation in the MOS score would makethe call quality unacceptable. Using two SMs and five calls from each SM, the MOS score was3.93, which is already on the borderline of when call quality begins to falter. In conclusion, eventhough 68 calls is the theoretical maximum, the call quality will lower to an unacceptable level and

will most likely not get very close to 68 calls with an acceptable call quality.

Recommendations

Based on the testing done in Phase II the following Conclusions & Recommendations are beingmade:

The G.711a & G.711u codecs performed the best in terms of highest MOS value witha high number of successful simultaneous calls completed. It is recommended touse these codecs for all VoIP applications. Although other codecs may allow more


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calls to pass through, it is at the cost of degraded call quality.

As the number of SMs is increased the performance of the AP also scales, which willaid in the Canopy network reaching the theoretical call maximum.

The number of calls an AP can handle will vary based on the codec. The codecswith higher bandwidth compression, such as G.729, will allow more calls to becompleted per AP as they take up less bandwidth per call. However, with thesecodecs the call quality is considerably less and as the call volume increases the MOSvalue will degrade and lead to unacceptable call quality as the theoretical maximumis approached.

The codecs with the lower bandwidth compression, such as G.711, allowed fewercalls to pass through at a much higher quality level. It would be expected thatdegradation in MOS value here would not have as much of an impact because it isstarting at a much higher value.

Bandwidth allocation through Canopy is static and should be taken into considerationwhen determining how much bandwidth is needed for a customers network.

o For a network that will be running primarily voice traffic, a WISP would want toconsider setting the downlink/uplink percentage at 50/50. This will give sufficientbandwidth to the uplink on the AP and will allow the SM to support more calls ata higher call quality rate.

o In a customer environment where non-voice data is still the primary type of databeing transferred, it may be that the WISP will require a significant amount ofbandwidth for downstream data transfer. The downstream bandwidth allocationrequirement could force the WISP to put the downlink/uplink closer to a 75%/25%configuration. If the expectation is that a customer will not be doing more than 1- 2 calls simultaneously from one SM, the 75%/25% configuration can transmit

the calls with very good quality (MOS value greater than 4 and low packet loss).

A high performance setting for Canopy QoS with the downlink/uplink set at 75%/25%is two high priority slots active. This will allow sufficient bandwidth for 1 or 2 calls tobe completed with good call quality, which will be typical in most Canopy applicationsin a residential setting.

QoS was shown to successfully filter data traffic independent of the size of the datafile. While increasing the data file size incrementally up to 1.2 Mb the voice callquality still passed at a MOS value well above the acceptable level of 4.


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**.%"

Phase II Test Results

The following tables depict the statistical results for each test using the IXIA Chariot software.Within each of the results below, the test setup, explanation and all variables are defined.

Canopy set at 50% downlink / 50% uplink G.711a

Multiple tests were conducted with VoIP traffic only for the G.711a (64 kbps) codec over one andtwo SMs. From the test results below, five VoIP calls were run successfully over one SM, but theaddition of a sixth call begins to show call degradation. As the upstream VoIP traffic begins to geta large one-way delay and amount of lost packets, the MOS score lowers which indicates callquality would become jittery and would not be acceptable.

Tests were then conducted using traffic from two SMs. As SMs are added, the AP is able toscale better as 5 calls from each SM were successfully completed (10 total). As a sixth call wasadded to each SM, a sharp increase in lost bytes and one-way delay resulted which contributedto unacceptable MOS scores. In a two SM environment, 10 11 calls can be supportedsuccessfully, but any additional calls will cause degradation in performance.

Table 6

CodecDownlink /

UplinkPercentage

Numberof SMs

Number ofcalls fromeach SM

One waydelay

avg (ms)Bytes Sent

MOSavg.

R-ValueAvg.

Lost Bytes

G.711a 50/50 1 5 20 12,000,000 4.37 91.32 0

G.711a 50/50 1 6 56 13,942,400 3.62 71.21 221,594

G.711a 50/50 2 5 32 11,999,840 4.36 90.85 160

G.711a 50/50 2 6 93 13,932,160 2.87 51.10 467,680

Canopy set at 50% downlink / 50% uplink G.711u

The tests below were conducted for the G.711u codec (64 kbps). This codec is very similar toG.711a and received similar results. From the data below, five VoIP calls were run successfullyover one SM, but the addition of a sixth call begins to show call degradation. When the tests arerun over two SMs, very similar results to the G.711a codec are produced. Five calls from eachSM were successful, but the addition of a sixth (11

thand 12

thcalls) increased lost bytes and one-

way delay which contributed to unacceptable MOS scores.

Table 7

Codec Downlink /UplinkPercentage

Numberof SMs


One waydelayavg (ms)

Bytes Sent MOSavg.

R-ValueAvg.

Lost Bytes

G.711u 50/50 1 5 19 11,979,040 4.34 90.42 20,960

G.711u 50/50 1 10 66 18,892,320 2.69 45.82 4,843,840

G.711u 50/50 2 5 13 11,999,840 4.37 91.35 160

G.711u 50/50 2 6 100 13,928,960 2.86 51.24 456,160


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Canopy set at 50% downlink / 50% uplink G.726

Multiple tests were conducted with VoIP traffic only for the G.726 codec (32 kbps) over one andtwo SMs. From the test results below, six VoIP calls were run successfully over one SM, but the

addition of a seventh call begins to show call degradation. As the upstream VoIP traffic begins toget a large one-way delay and amount of lost packets, the MOS score lowers which indicates callquality would become jittery and choppy and would not be acceptable.

Tests were then conducted using traffic from two SMs. As SMs are added, the AP is able toscale better and is able to successfully complete 5 calls from each SM (10 total). As additionalcalls were added to each SM a sharp increase in lost bytes and one-way delay resulted whichcontributed to unacceptable MOS scores. In a two SM environment, 12 14 calls can besupported successfully, but any additional calls will cause degradation in performance.

Table 8

CodecDownlink /

UplinkPercentage

Numberof SMs

Numberof calls

from

each SM

One waydelay

avg (ms)Bytes Sent

MOSavg.

R-ValueAvg.

Lost Bytes

G.726 50/50 1 6 23 7,199,920 4.18 84.44 80

G.726 50/50 1 7 68 8,397,120 2.60 45.75 1,390,320

G.726 50/50 2 5 22 11,998,320 4.16 83.96 1,680

G.726 50/50 2 8 66 16,963,960 2.52 40.57 2,236,000


Multiple tests were conducted with VoIP traffic only for the G.729 (8 kbps) codec over one andtwo SMs. From the test results below, six VoIP calls were run successfully over one SM, but theaddition of a seventh call begins to show call degradation. As the upstream VoIP traffic begins toget a large one-way delay and a large amount of lost packets, the MOS score lowers whichindicates call quality would become jittery and would not be acceptable.

Tests were then conducted using traffic from two SMs. As SMs are added, the AP is able toscale better as five calls from each SM (10 total) were completed at the acceptable limit. Asadditional calls were added to each SM, a sharp increase in lost bytes and one-way delayresulted in unacceptable MOS scores. In a two SM environment, 10 calls can be supportedsuccessfully, but any additional calls will cause degradation in performance.

Table 9

CodecDownlink /

UplinkPercentage

Numberof SMs


One waydelay

avg (ms)Bytes Sent

MOSavg.

R-ValueAvg.

LostBytes

G.729 50/50 1 6 21 1,800,000 4.02 79.87 20

G.729 50/50 1 7 57 2,096,860 3.41 67.18 98,260

G.729 50/50 2 5 21 2,991,500 3.93 77.72 8,500G.729 50/50 2 8 54 4,799,760 2.90 56.66 486,640

Canopy set at 50% downlink / 50% uplink G.723.1-ACELP

Multiple tests were conducted with VoIP traffic only for the G.723.1-ACELP (5.3 kbps) codec overone and two SMs. From the test results below, this codec received a low MOS score eventhough there was no lost data and a low one-way delay. Test results showed that anywhere fromone to five calls had approximately the same results. Also, performing a test with a low amount ofcalls between two SMs resulted in a low MOS score.


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The test results suggest that this codec does not transfer well over a VoIP network due to the lowkbps. It is recommended to not use this codec for any VoIP applications independent of thenetwork being used.

Table 10

CodecDownlink /

UplinkPercentage

Numberof SMs


One waydelay

avg (ms)Bytes Sent

MOSavg.

R-ValueAvg.

LostBytes

G.7231 -ACELP

50/50 1 1 - 5 18 990,000 3.63 70.65 0

G.7231 -ACELP

50/50 1 10 45 1,977,540 3.24 62.61 38,000

G.7231 -ACELP

50/50 2 8 30 3,168,000 3.60 70.09 320

G.7231 -ACELP

50/50 2 10 56 3,959,780 3.19 61.56 76,180

Canopy set at 50% downlink / 50% uplink G.723.1-MPMLQMultiple tests were conducted with VoIP traffic only for the G.723.1-ACELP (6.3 kbps) codec overone and two SMs. From the test results below, this codec received very similar scores as theACELP codec as it displayed a low MOS score even though there was no lost data and a lowone-way delay. Test results showed that anywhere from one to five calls had approximately thesame results. Adding a low amount of calls between two SMs resulted in a similarly low MOSscore.

The test results suggest that this codec does not transfer well over a VoIP network due to the lowkbps. It is recommended to not use this codec for any VoIP applications independent of thenetwork being used.

Table 11

CodecDownlink /

UplinkPercentage

Numberof SMs

Numberof calls

fromeach SM

One waydelay

avg (ms)Bytes Sent

MOSavg.

R-ValueAvg.

Lost Bytes

G.7231 -MPMLQ

50/50 1 1 - 5 19 1,176,000 3.80 74.63 0

G.7231 -MPMLQ

50/50 1 10 58 2,346,672 3.44 67.06 35,184

G.7231 -MPMLQ

50/50 2 8 24 3,763,200 3.80 74.39 312

G.7231 -MPMLQ

50/50 2 10 91 4,646,688 2.46 42.11 1,198,272

Canopy set at 75% downlink / 25% uplink G.711a

Multiple tests were conducted with VoIP traffic only for the G.711a (64 kbps) codec over one andtwo SMs. From the test results below, four VoIP calls were run successfully over one SM, but theaddition of a fifth call begins to show call degradation. As the upstream VoIP traffic begins to geta large one-way delay and a large amount of lost packets, the MOS score lowers which indicatescall quality would become jittery and would not be acceptable.

Tests were then conducted using traffic from two SMs. As SMs are added, the AP is able toscale better as four calls from each SM (8 total) were completed successfully. As a fifth call wasadded to each SM, a sharp increase in lost bytes and one-way delay contributed to unacceptable


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MOS scores. In a two SM environment, 8 9 calls can be supported successfully, but anyadditional calls will cause degradation in performance.

Table 12

CodecDownlink /

UplinkPercentage

Numberof SMs

Numberof calls

fromeach SM

One waydelay avg

(ms)Bytes Sent

MOSavg.

R-ValueAvg.

Lost Bytes

G.711a 75/25 1 4 22 9,512,000 4.22 87.07 16,960

G.711a 75/25 1 5 68 11,906,720 3.54 69.87 218,880

G.711a 75/25 2 4 24 19,047,040 4.13 85.02 52,320

G.711a 75/25 2 5 71 16,842,240 3.41 67.24 349,280

Canopy set at 75% downlink / 25% uplink G.711u

Multiple tests were conducted with VoIP traffic only for the G.711u (64 kbps) codec over one andtwo SMs. From the test results below, four VoIP calls were run successfully over one SM, but theaddition of a fifth call begins to show call degradation. As the upstream VoIP traffic begins to geta large one-way delay and a large amount of lost packets, the MOS score lowers which indicatescall quality would become jittery and choppy and would not be acceptable.

Tests were then conducted using traffic from two SMs. As SMs are added, the AP is able toscale better as four calls from each SM (8 total) we completed successfully. As a fifth call to eachSM was added, a sharp increase in lost bytes and one-way delay contributed to unacceptableMOS scores. In a two SM environment, 8 9 calls can be supported successfully, but anyadditional calls will cause degradation in performance.

Table 13

Codec

Downlink /

UplinkPercentage

Numberof SMs

Number of

calls fromeach SM

One way

delayavg (ms)

Bytes SentMOSavg.

R-ValueAvg.

LostBytes

G.711u 75/25 1 4 23 9,600,000 4.33 90.13 4,960

G.711u 75/25 1 5 69 12,000,000 3.78 76.11 121,600

G.711u 75/25 2 4 28 19,200,000 4.32 89.67 11,840

G.711u 75/25 2 5 70 23,999,360 3.78 76.29 240,960


Multiple tests were conducted with VoIP traffic only for the G.726 (32 kbps) codec over two SMs.From the test results below, 5 calls from each SM (10 total) were completed successfully. As asixth call was added to each SM, a sharp increase in lost bytes and one-way delay contributed tounacceptable MOS scores. In a two SM environment, 10 11 calls can be supportedsuccessfully, but any additional calls will cause degradation in performance.

Table 14

CodecDownlink /

UplinkPercentage

Numberof SMs


One waydelay

avg (ms)Bytes Sent

MOSavg.

R-ValueAvg.

LostBytes

G.726 75/25 2 5 40 11,875,520 4.02 80.70 22,720

G.726 75/25 2 6 75 14,399,360 2.83 50.64 948,720


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Multiple tests were conducted with VoIP traffic only for the G.729 (8 kbps) codec over two SMs.From the test results below, 5 calls from each SM (10 total) were completed successfully. As a

sixth call to each SM was added, a sharp increase in lost bytes and one-way delay contributed tounacceptable MOS scores. In a two SM environment, 10 11 calls can be supportedsuccessfully, but any additional calls will cause degradation in performance.

Table 15

CodecDownlink /

UplinkPercentage

Numberof SMs


One waydelay

avg (ms)Bytes Sent

MOSavg.

R-ValueAvg.

LostBytes

G.729 75/25 2 5 24 2,961,440 4.01 79.73 1,040

G.729 75/25 2 6 61 3,546,880 3.57 70.44 96,880

Canopy set at 75% downlink / 25% uplink G.723.1-ACELP and G.723.1-MPMLQ

Due to the poor performance of these codecs during the tests where Canopy was set at 50%downlink and 50% uplink, no tests were run in the 75% / 25% configuration. Since the uplink isthe known constraint, lowering it would only lower the performance of the codec. The test resultssuggest that this codec does not transfer well over a VoIP network due to the low kbps. It isrecommended to not use these codecs for any VoIP applications independent of the networkbeing used.

Test Scenarios with Canopy Quality of Service (QoS)To test the robustness of the quality of service functionality a data file was passed through at thesame time as the VoIP. The purpose was to see how effectively the AP could regulate the trafficand if it was at all affected by the amount of data that was being simultaneously passed with theVoIP traffic. The size of the data files were incrementally increased starting at 128 Kb up to 1.2Mb. As shown in the table below, the quality of the call did not fluctuate noticeably with increasedamount of data being passed. This is shown by noting the MOS values being at 4.36 when 128kbps file was passed and 4.29 when a 1.2 Mbps file was passed. These results demonstrate thatregardless of the amount of data, the AP will provide a level of guaranteed bandwidth for VoIPtraffic.

Table 16

CodecDownlink/

UplinkPercentage

TestingConfig.

(# of SMs)

Number ofCalls from

SM

Data FileBandwidth

SizeBytes Sent

MOSAvg.

R-ValueAvg.

LostBytes

G.711u 75/25 1 1 128 Kbps 8,800,000 4.36 90.95 480

G.711u 75/25 1 1 1.2 Mbps 62,400,000 4.29 88.89 2,400

Now that it has been shown that the high priority bit will serve the function of regulating the datatraffic while voice traffic is present, the limitations needed to be tested. In the following testscenarios the number of high priority slots allocated for voice traffic was adjusted from 1 slot up to3 slots. This was done to demonstrate whether there is a positive, negative or indifferent impacton overall performance and throughput as a result of adding more slots for high priority.

Another variable that was adjusted was the bandwidth allocation on the downlink and uplink from75/25 to 50/50. The number of Acknowledgement slots for downlink and uplink were not varied.


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The different codecs were tested with one or two SMs as well as different call volumes todetermine the breaking point in quality.

Test Results with Canopy QoSTesting on QoS was done using both the 75/25 and 50/50 downlink/uplink settings. In regards toQoS, the most important setting is how many high priority slots are allocated. This is a function ofthe downlink/uplink percentage and the high priority percentage that is set in the configuration ofthe AP. Different percentage configurations will yield a varied number of high priority slotsdepending on your Canopy environment and configurations. When applying these settings it isencouraged to adjust them accordingly such that the desired number of high priority slots isproduced. Some example scenarios and their results from the testing are shown below.

High Priority = 1 Slot

This test was done with 1 high priority slot set. To achieve one high priority slot, thedownlink/uplink percentage was set to 75/25, the high priority percentage was set to 25%, andthe G.711u codec was tested. The tests were run with one and two SMs. With one SM two calls

were successfully completed, which is shown in the table below with a MOS score of 4.34. Withtwo SMs, one call from each SM failed to complete at an acceptable quality level, which is shownin the table below with a MOS score of 2.69.

Downlink Data Percentage 75% UpLink Data Percentage 25%

Downlink Slots 21 Uplink Slots 7

High Priority Slots 1

Total Num UAck Slots 6 UAcks Reserved High 3

Num DAck Slots 6 DAcks Reserved High 3

Num Ctl Slots 6 Num Ctl Slots Reserved High 3

Table 17

CodecDownlink/

UplinkPercentage

TestingConfig.

(# of SMs)

Numberof Callsfrom SM

Data FileBandwidth

SizeBytes Sent

MOSAvg.

R-ValueAvg.

Lost Bytes

G.711u 75/25 1 2 22 4,800,000 4.34 90.43 2,080

G.711u 75/25 1 3 231 7,195,680 2.69 45.81 1,766,880

G.711u 75/25 2 1 355 4,800,000 2.69 45.81 132,800

High Priority = 2 Slots

This test was done with two high priority slots set. To achieve two high priority slots, thedownlink/uplink percentage was set to 75/25, the high priority percentage was set to 25% and theG.711u codec was tested. The tests were run with one and two SMs. With one SM two callswere successfully completed, which is shown in the below table with a MOS score of 4.30. Withtwo SMs, two calls total (one from each SM) were completed at an acceptable quality level, whichis shown in the table below with a MOS score of 4.33. Once the call volume was increased totwo calls per SM, the high priority channel failed to handle it at an acceptable quality level.


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Table 18

CodecDownlink/

UplinkPercentage

TestingConfig.

(# of SMs)


One WayDelay Avg.

(ms)Bytes Sent

MOSAvg.

R-ValueAvg.

Lost Bytes

G.711u 75/25 1 2 30 4,800,000 4.30 89.32 2,880

G.711u 75/25 1 3 128 7,200,000 2.73 49.60 529,440

G.711u 75/25 2 1 21 4,800,000 4.33 90.14 2,400

G.711u 75/25 2 2 68 9,600,000 2.73 47.41 282,240

The G.711 codec has shown to be the most consistent and highest in quality in terms of overallperformance. The above tests showed that the 75/25 bandwidth allocation did not allow a largeamount of calls through the network while using the G.711 codec and some instances didnt alloweven one call through. It was determined that further testing on the other codecs would notproduce more favorable results, so the testing was limited to just this codec.

High Priority = 3 Slots

This test was with 3 high priority slots set. To achieve three high priority slots, the downlink/uplinkpercentage was set to 50/50, the high priority percentage was set to 25% and the G.711, G.726and G.729 codecs were tested. The tests were run with one and two SMs. By allocating morebandwidth by increasing the uplink percentage, an extra high priority slot was made available.This allowed more calls to be successfully completed utilizing the high priority channels.However, keep in mind that these slot allocations are static and one data slot needs to be tradedfor every additional slot that is allocated for high priority VoIP traffic. To make sure that there isnot too much idle bandwidth in the network, it is in the best interest of the customer to closelycalculate how much bandwidth is necessary for high priority.








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G.711G.711a and G.711u performed almost identically, so the results of just G.711u have been listedbelow as a representation of both tests. With one SM, three calls were successfully completed

with an average MOS score of 4.29. When another SM was added the total number ofsuccessful calls increased to four, (2 calls from each SM), with a MOS score of 4.29.

Table 19

CodecDownlink/

UplinkPercentage

TestingConfig.

(# of SMs)


One WayDelay Avg.

(ms)Bytes Sent

MOSAvg.

R-ValueAvg.

Lost Bytes

G.711 50/50 1 3 31 7,200,000 4.29 89.06 6,560

G.711 50/50 1 4 108 9,599,680 2.68 45.76 967,040

G.711 50/50 2 2 25 9,600,000 4.29 89.05 8,320

G.711 50/50 2 3 124 14,369,120 3.08 59.59 455,040

G.726With one SM, three calls were successfully completed with an average MOS score of 4.12.When an additional SM was added, the total number of successful calls increased to six (3 callsfrom each SM) with a MOS score of 4.07.

Table 20

CodecDownlink/

UplinkPercentage

TestingConfig.

(# of SMs)


One WayDelay Avg.

(ms)Bytes Sent

MOSAvg.

R-ValueAvg.

Lost Bytes

G.726 50/50 1 3 25 3,600,000 4.12 82.82 2,640

G.726 50/50 1 4 91 4,800,000 2.84 55.82 142,560

G.726 50/50 2 3 25 7,200,000 4.07 81.77 8,880

G.726 50/50 2 4 93 9,600,000 2.87 56.44 290,640

G.729With one SM, four calls were successfully completed with an average MOS score of 4.02. Whenanother SM was added the total number of successful calls increased to eight (4 calls from eachSM) with an average MOS score of 4.02.

Table 21

CodecDownlink/

UplinkPercentage

TestingConfig.

(# of SMs)


One WayDelay Avg.

(ms)Bytes Sent

MOSAvg.

R-ValueAvg.

Lost Bytes

G.729 50/50 1 4 22 1,200,000 4.02 79.90 80

G.729 50/50 1 5 77 1,500,000 3.39 66.52 63,900

G.729 50/50 2 4 22 2,400,000 4.02 79.82 640

G.729 50/50 2 5 78 2,999,920 3.38 66.21 128,460


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()#()'

Acronym Meaning

AP Access Point

Codec Compressor/Decompressor

CPE Customer Premise Equipment

DSP Digital Signal Processing

ITU International Telecommunications Union

kbps 1000 bits per second

Kbps 1024 bits per second

Mbps 1024 Kbps = 1,048,576 bitsMOS Mean Opinion Score - To determine MOS, a number of listeners rate the

quality of test sentences read aloud over the communications circuit bymale and female speakers. A listener gives each sentence a rating asfollows: (1) bad; (2) poor; (3) fair; (4) good; (5) excellent. The MOS is thearithmetic mean of all the individual scores, and can range from 1 (worst)to 5 (best)

PSTN Public Switched Telephone Network

QoS Quality of Service

R-Value/Factor The E-model is a complex formula; the output of an E-model calculation isa single score, called an R factor, derived from delays and equipmentimpairment factors. R factor values range from 100 (excellent) down to 0(poor).

SM Subscriber ModuleTOS Type of Service

VoIP Voice Over IP

VPN Virtual Private Network

WISP Wireless Internet Service Provider


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1. Canopy Network Advanced Technical Training, Canopy Advanced Technical Training

Course, Motorola.

2. D. De Vleeschauwer and J.Janseen, Voice performance over packet based networks,Alcatel Networks Technology White Paper, 2002.

3. J. Walker, A handbook for Successful VoIP Deployment: Network Testing, QoS and More,NetIQ Corporation, 2002.

4. K. Adams, K. Bhalla, An Introduction to Internet Telephony (or Voice over IP) , OperationalManagement Report, Gartner Group, November 25, 2003.

5. VoIP Test Module for Chariot, NetIQ Corporation, 2002.

6. Westbay Engineers Limited What is an Erlang, http://erlang.com/whatis.html.

7. Westbay Engineers Limited Erlang to VoIP Bandwidth Calculator,http://erlang.com/calculator/eipb/.

Disclaimer:

This whitepaper merely provides a starting point for planning and sizing hardware requirementsfor customers to deploy VoIP over Canopy. Because these tests were run in constrainedenvironments, such as an isolated lab, they do not necessarily translate directly to deployablescenarios. Therefore, it is important to understand that while this whitepaper is meant to help

customers prepare for a VoIP over Canopy roll out and capacity-planning effort, any datagenerated contained in this whitepaper is only meant for general sizing, benchmarking, ordeployment recommendations. Results may not be representative and may vary. Accordingly,neither Motorola nor West Monroe Partners can guarantee actual results in a real worlddeployment. In addition to these benchmarking results and recommendations, customers shouldalso consider, but not limit, evaluation to point-to-point mileage, line of sight, network capacity,and expected peak call volume time (Erlang tables) when planning a VoIP overCanopy deployment.

Documents

Motorola Canopy VoIP Whitepaper v4