designing and deploying a voip network when itu meets ietf thomas(at)kernen.net
TRANSCRIPT
Designing and Designing and deploying a VoIP deploying a VoIP
networknetworkWhen ITU meets IETFWhen ITU meets IETF
Thomas(at)Kernen.NetThomas(at)Kernen.Net
A quick VoIP recapA quick VoIP recap Directory Gatekeeper (DGK): Performs call routing Directory Gatekeeper (DGK): Performs call routing
search at highest level (ex: country code distributes). search at highest level (ex: country code distributes). Country codes among other DGKs Forward LRQ Country codes among other DGKs Forward LRQ (location request) to a partner DGK if call doesn't (location request) to a partner DGK if call doesn't terminate in local SP DGKterminate in local SP DGK
Gatekeeper (GK): Performs call routing search at Gatekeeper (GK): Performs call routing search at intermediate level (ex: NPA-NXX). Distributes NPA intermediate level (ex: NPA-NXX). Distributes NPA among other GKs. Provides GW resource management among other GKs. Provides GW resource management (Ressource Availabilty Indicator, gw-priority, ....)(Ressource Availabilty Indicator, gw-priority, ....)
Gateway (GW): Acts as interface between the PSTN Gateway (GW): Acts as interface between the PSTN and IP. Normalizes numbers from PSTN before and IP. Normalizes numbers from PSTN before entering IP. Normalizes numbers from the IP before entering IP. Normalizes numbers from the IP before entering the PSTN. Contains the dial-peer entering the PSTN. Contains the dial-peer configuration. Registers with the GK.configuration. Registers with the GK.
Gatekeeper A Gatekeeper B
RRQ/RCF
ARQ
RRQ/RCF
LRQ
IP Network
Phone A
Gateway A Gateway B
H.225 (Q.931) Setup
H.225 (Q.931) Alert and Connect
H.245
RTP
ACF
LCF
VV
Basic H.323 CallBasic H.323 Call
VV
ARQ
ACF
Phone B
Various Codec Various Codec Bandwidth Bandwidth
ConsumptionsConsumptionsEncoding/Compression
ResultBit Rate
G.711 PCMA-Law/u-Law
64 kbps (DS0)
G.726 ADPCM 16, 24, 32, 40 kbps
G.727 E-ADPCM
G.729 CS-ACELP 8 kbps
G.728 LD-CELP 16 kbps
G.723.1 CELP 6.3/5.3 kbpsVariable
16, 24, 32, 40 kbps
StandardTransmissionRate for Voice
Cisco Encoding Cisco Encoding ImplementationImplementation
= Sample
8 kHz (8,000 Samples/Sec)
= 0010110101
IP QoS WAN
Encode Decode
20 Byte packet every 20ms (50pps)8kbps Data Rate
Note - This 8bkps for “Voice Payload” only!!
Add on 40 bytes of IP/UDP/RTP and you now have 24kbps!RTP Header Compression will take this down to 11.2kbps
Voice Quality of Service Voice Quality of Service (QoS) Requirements(QoS) Requirements
Loss
Delay
Delay Variation (Jitter)
Avoiding The 3 Main QoS Challenges
Loss and Delay SourcesLoss and Delay Sources
• Input queuing
• Jitter buffer
• CODEC (Decode)
• Access (up) link transmission
• Backbone network transmission
• Access (down) link transmission
• CODEC (Encode)
• Packetization
• Output queuingVoice Path
Loss+
Delay+
DelayVariation
Delay—How Much Is Too Delay—How Much Is Too Much? Much?
Cumulative Transmission Path Delay
Time (msec)
0 100 200 300 400
CB ZoneCB Zone
Satellite QualitySatellite Quality
Fax Relay, BroadcastFax Relay, BroadcastHigh QualityHigh Quality
Delay Target
500 600 700 800
ITU’s G.114 Recommendation = 0 – 150msec 1-way delay
Fixed Delay ComponentsFixed Delay Components
Propagation—six microseconds per kilometerPropagation—six microseconds per kilometer Serialization Serialization ProcessingProcessing
Coding/compression/decompression/decodingCoding/compression/decompression/decodingPacketizationPacketization
Processing Delay
Propagation Delay
Serialization Delay—Buffer to Serial Link
Variable Delay Variable Delay Components Components
Queuing delayQueuing delay Dejitter buffersDejitter buffers Variable packet sizesVariable packet sizes
DejitterBuffer
Queuing Delay
Queuing Delay
Queuing Delay
56kb WAN
Large Packets “Freeze Large Packets “Freeze Out” VoiceOut” Voice
Large packets can cause playback buffer Large packets can cause playback buffer underrun, resulting in slight voice degradationunderrun, resulting in slight voice degradation
Jitter or playback buffer can accommodate Jitter or playback buffer can accommodate some delay/delay variationsome delay/delay variation
~214ms Serialization Delay
10mbps Ethernet 10mbps Ethernet
Voice Packet60 bytes
Every 20ms
Voice 1500 bytes of Data Voice
Voice Packet60 bytes
Every >>214ms
Voice Packet60 bytes
Every >214ms>214ms
Voice 1500 bytes of Data Voice
Voice 1500 bytes of Data Voice
RTP Controlling Dejitter RTP Controlling Dejitter BufferBuffer
RTP Timestamp From Router AInterframe gap of 20ms
CC
Sender Receiver
IPNetwork
VV VV
BB AA
RouterA RouterB
10 30 50
20ms 20ms
RTP Timestamp From Router AVariable Interframe Gap (Jitter)
CC BB AA10 30 50
20ms 80ms
RTP Timestamp From Router ADelitter Buffer removes Variation
CC BB AA10 30 50
20ms 20ms
Calculate Delay Budget - Calculate Delay Budget - Worst CaseWorst Case
PropagationDelay—8 ms
Coder Delay25 ms
Serialization Delay2 ms
Dejitter Buffer50 ms
QueuingDelay4 ms
Site A Site B
(128kbps Frame Relay)
Total 89 msecDejitter Buffer 50 msec
Min 8 msecNetwork Delay (e.g.,Public Frame Relay Svc)
Serialization Delay 128 kbps Trunk 2 msec
4 msecQueuing Delay 128 kbps Trunk
5 msec
Packetization Delay—Included in Coder Delay
Coder Delay G.729 (5 msec look ahead)
Propagation Delay (Private Lines)
Fixed Delay
Variable Delay
Coder Delay G.729 (10 msec per frame) 20 msec
Fragmentation and Fragmentation and InterleavingInterleaving
Serialization delay for 64Kbps link with an Serialization delay for 64Kbps link with an MTU of 1500 bytesMTU of 1500 bytes
(1500 bytes x 8bits/byte) / (64000 bits/sec) (1500 bytes x 8bits/byte) / (64000 bits/sec) = 187.5ms= 187.5ms
Fragmentation size: design for 10ms Fragmentation size: design for 10ms fragmentsfragments
(0.01 sec x 64000 bps) / (8 bits/byte) = 80 (0.01 sec x 64000 bps) / (8 bits/byte) = 80 bytesbytes
It takes 10 ms to send an 80 byte packet or It takes 10 ms to send an 80 byte packet or fragment over a 64kbps link.fragment over a 64kbps link.
Fixed Frame Serialization Fixed Frame Serialization Delay MatrixDelay Matrix
Frame Size
LinkSpeed
56kbps
64kbps
128kbps
256kbps
512kbps
768kbps
1536kbs
1Byte
143us
125us
62.5us
31us
15.5us
10us
5us
64Bytes
9ms
8ms
4ms
2ms
1ms
640us
320us
18ms
128Bytes
16ms
8ms
4ms
2ms
1.28ms
640us
36ms
256Bytes
32ms
16ms
8ms
4ms
2.56ms
1.28ms
72ms
512Bytes
64ms
32ms
16ms
8ms
5.12ms
2.56ms
144ms
1024Bytes
128ms
64ms
32ms
16ms
10.24ms
5.12ms
1500Bytes
46ms
214ms
187ms
93ms
23ms
15mss
7.5ms
Multilink PPP with Multilink PPP with Fragmentation and Fragmentation and
InterleaveInterleave
Elastic Traffic MTUReal-Time MTU
64 kbps Line
Elastic MTU Real-Time MTUElastic MTU Elastic MTU
Addendum to PPP Specification
187ms Serialization Delayfor 1500 byte Frame at 64 kbps
64 kbps Line
Media Link Layer Media Link Layer OverheadOverhead
Layer 2 Media Layer 2 Header Size
Ethernet 14 bytes
PPP/MLPPP 6 bytes
Frame Relay
ATM (AAL5) 5 bytes + waste5 bytes + waste
MLPPP over FR 14 bytes
MLPPP over ATMMLPPP over ATM 5 bytes for every ATM cell5 bytes for every ATM cell+ 20 bytes for MLPPP/AAL5+ 20 bytes for MLPPP/AAL5
6 bytes6 bytes
RTP Header RTP Header CompressionCompression
20ms@8kb/s yields 20 20ms@8kb/s yields 20 byte payloadbyte payload
IP header 20; UDP header IP header 20; UDP header 8; RTP header 128; RTP header 12
2X payload!!!!!!!!2X payload!!!!!!!! Header compression Header compression
40Bytes to 2-4 much of the 40Bytes to 2-4 much of the timetime
Hop-by-HopHop-by-Hop on on slow links <512kbpsslow links <512kbps
CRTP—Compressed Real-CRTP—Compressed Real-time protocoltime protocol
Overhead
Version IHL Type of Service Total Length
Identification Flags Fragment Offset
Header ChecksumProtocolTime to Live
Source Address
Destination Address
PaddingOptions
Source Port Destination Port
ChecksumLength
PTPTMMCCCCXXPPV=2V=2 Sequence NumberSequence Number
TimestampTimestamp
Synchronization Source (SSRC) IdentifierSynchronization Source (SSRC) Identifier
RTP Header compression RTP Header compression detailsdetails
Can save a lot of bandwidth (>50%) per Can save a lot of bandwidth (>50%) per flow.flow.
Works on serial links between 2 routersWorks on serial links between 2 routers CPU intensive, might overkill the routersCPU intensive, might overkill the routers Limited to 256 sessions (128 calls) over FRLimited to 256 sessions (128 calls) over FR Limited to 1000 sessions (500 calls) over Limited to 1000 sessions (500 calls) over
HDLC (checked in 12.2(8)T)HDLC (checked in 12.2(8)T) Not recommend on links with data rates Not recommend on links with data rates
above E1above E1
Silence suppressionSilence suppression
VAD (Voice Activity Detection) (Cisco)VAD (Voice Activity Detection) (Cisco) Codec built-in silence suppression Codec built-in silence suppression
(G.729a/G.723.1b)(G.729a/G.723.1b) Should not be taken into account for Should not be taken into account for
circuits carrying less than 24/30 calls circuits carrying less than 24/30 calls since based on aggregate volume, not since based on aggregate volume, not individual calls.individual calls.
Should not be taken into account when Should not be taken into account when engineering the network.engineering the network.
IP Precedence/DSCPIP Precedence/DSCP
DSCP - Differentiated Services Code DSCP - Differentiated Services Code Point (RFC 2474-2475)Point (RFC 2474-2475)
Set IP Precedence/DSCP higher for Set IP Precedence/DSCP higher for VoIP. Usually set to 5/101000VoIP. Usually set to 5/101000
Set at source (gateway) if possible Set at source (gateway) if possible for less hassle.for less hassle.
Queuing mechanisms Queuing mechanisms (in Cisco’s world)(in Cisco’s world)
FIFO, First In First OutFIFO, First In First Out Packets arrive and leave the queue in exactly the Packets arrive and leave the queue in exactly the
same ordersame order Simple configuration and fast operationSimple configuration and fast operation No Priority servicing or bandwidth guarantees No Priority servicing or bandwidth guarantees
possiblepossible
WFQ, Weighted Fair QueuingWFQ, Weighted Fair Queuing A hashing algorithm, places flows into separate A hashing algorithm, places flows into separate
queues where weights are used to determine how queues where weights are used to determine how many packets are serviced at a time. You define many packets are serviced at a time. You define weights by setting IP Precedence and DSCP values.weights by setting IP Precedence and DSCP values.
Simple configuration.Simple configuration. No priority servicing or bandwidth guarantees No priority servicing or bandwidth guarantees
possible.possible.
Queuing mechanisms (2)Queuing mechanisms (2) CQ, Custom QueuingCQ, Custom Queuing Traffic is classified into multiple queues with configurable Traffic is classified into multiple queues with configurable
queue limits.queue limits. Has been available for a few years and allows approximate Has been available for a few years and allows approximate
bandwidth allocation for different queues.bandwidth allocation for different queues. No priority servicing possible. Bandwidth guarantees are No priority servicing possible. Bandwidth guarantees are
approximate and there are a limited number of queues. approximate and there are a limited number of queues. Configuration is relatively difficult.Configuration is relatively difficult.
PQ, Priority QueuingPQ, Priority Queuing Traffic is classified into high, medium, normal and low Traffic is classified into high, medium, normal and low
priority traffic is serviced first, then medium priority traffic, priority traffic is serviced first, then medium priority traffic, followed by normal and low priority traffic.followed by normal and low priority traffic.
Has been available for a few years and provides priority Has been available for a few years and provides priority servicing.servicing.
Higher priority traffic can starve lower priority queues of Higher priority traffic can starve lower priority queues of bandwidth. No bandwidth guarantees possible.bandwidth. No bandwidth guarantees possible.
Queuing mechanisms (3)Queuing mechanisms (3) CBWFQ, Class Based Weighted Fair QueuingCBWFQ, Class Based Weighted Fair Queuing MQC is used to classify traffic. Classified traffic is placed into MQC is used to classify traffic. Classified traffic is placed into
reserved bandwidth queues or a default unreserved queue.reserved bandwidth queues or a default unreserved queue. Similar to LLQ except there is no priority queue. Simple Similar to LLQ except there is no priority queue. Simple
configuration and ability to provide bandwidth guarantees. No configuration and ability to provide bandwidth guarantees. No priority servicing possible.priority servicing possible.
PQ-WFQ, Priority queue-Weighted Fair Queuing (IP RTP PQ-WFQ, Priority queue-Weighted Fair Queuing (IP RTP Priority)Priority)
Single interface command is used to provide priority servicing Single interface command is used to provide priority servicing to all UDP packets destined to even port numbers within a to all UDP packets destined to even port numbers within a specific range.specific range.
Simple, one command config. Provides priority servicing to RTP Simple, one command config. Provides priority servicing to RTP packets.packets.
All other traffic is treated with WFQ. RTCP traffic is not All other traffic is treated with WFQ. RTCP traffic is not prioritized. No guaranteed bandwidth capability.prioritized. No guaranteed bandwidth capability.
Note: MQC = Modular QoS CLINote: MQC = Modular QoS CLI
Queuing mechanisms (4)Queuing mechanisms (4) Low Latency Queueing (LLQ) = Priority Queue (PQ)Low Latency Queueing (LLQ) = Priority Queue (PQ)
+ Class Based-Weighted Fair Queue (CB-WFQ).+ Class Based-Weighted Fair Queue (CB-WFQ). Allows a strict Priority Queue to handle a defined Allows a strict Priority Queue to handle a defined
class of packet to be prioritized over all other traffic.class of packet to be prioritized over all other traffic. Simple config, ability to provide priority to multiple Simple config, ability to provide priority to multiple
classes of traffic and give upper bounds on priority classes of traffic and give upper bounds on priority bandwidth utilization. Can also config bandwidth bandwidth utilization. Can also config bandwidth guaranteed classes and a default class.guaranteed classes and a default class.
All priority traffic is sent throught the same priority All priority traffic is sent throught the same priority queue which can introduce jitter.queue which can introduce jitter.
Note: Cisco appears to be working on improving LLQ Note: Cisco appears to be working on improving LLQ and this is currently the #1 queuing mechanism and this is currently the #1 queuing mechanism according to SEs, TAC and updated documentation.according to SEs, TAC and updated documentation.
TrafficTraffic Engineering Engineering Busy Hour (BH) = Number of lines required to support Busy Hour (BH) = Number of lines required to support
the worst hour of the daythe worst hour of the day Grade of service (GOS) = Percentage of lines that will Grade of service (GOS) = Percentage of lines that will
experience a busy tone on the 1st attempt during the experience a busy tone on the 1st attempt during the BHBH
A GOS of 0.05 means 5 out of 100 callers might get a A GOS of 0.05 means 5 out of 100 callers might get a busy tonebusy tone
Erlang B, most widely used traffic model to estimate Erlang B, most widely used traffic model to estimate the number of lines required for a specific GOS and BH the number of lines required for a specific GOS and BH of traffic.of traffic.
Based on various traffic assumptions such as call Based on various traffic assumptions such as call queueing, arrival rate, etc...queueing, arrival rate, etc...
1 trunk in use for 1 hour = 1 Erlang = 36 CCS of traffic1 trunk in use for 1 hour = 1 Erlang = 36 CCS of traffic 1 Centrum Call Seconds (CCS) = 100 call seconds1 Centrum Call Seconds (CCS) = 100 call seconds 1 hour = 3600 seconds or 36 CCS = 1 Erlang 1 hour = 3600 seconds or 36 CCS = 1 Erlang
Traffic Traffic Engineering (2)Engineering (2)
Step1: Obtain voice traffic dataStep1: Obtain voice traffic data Sources of traffic information: CDRs (Call Sources of traffic information: CDRs (Call
Detail Record) or carrier bills, carrier studies, Detail Record) or carrier bills, carrier studies, traffic reportstraffic reports
Data needs to be adjusted for call processing Data needs to be adjusted for call processing since a trunk in use = Dialing + Call setup + since a trunk in use = Dialing + Call setup + Ringing + Talking + ReleasingRinging + Talking + Releasing
Other sources: Ring No Answer, Busy Signal, Other sources: Ring No Answer, Busy Signal, etcetc
Add 10% to 16% to all call lengths/total time Add 10% to 16% to all call lengths/total time estimates.estimates.
TrafficTraffic Engineering (3) Engineering (3) Step 2: Convert to ErlangsStep 2: Convert to Erlangs Adjusted total hours a month / business Adjusted total hours a month / business
days * % of traffic in busy hourdays * % of traffic in busy hour Step 3: Calculate the number of voice linesStep 3: Calculate the number of voice lines Based on statistical model for the # of Based on statistical model for the # of
lines vs the grade of service desiredlines vs the grade of service desired Step 4: Calculate the data network Step 4: Calculate the data network
bandwidthbandwidth (Codec + protocol overhead) * number of (Codec + protocol overhead) * number of
voice lines = required bandwidthvoice lines = required bandwidth
POP POP SizingSizing Calculate the number of gateways (GW) Calculate the number of gateways (GW)
required to handle anticipated call volumerequired to handle anticipated call volume Use Busy Hour Call Attempts (BHCA) metricUse Busy Hour Call Attempts (BHCA) metric Calculate the number of (Directory) Calculate the number of (Directory)
Gatekeepers required to process the GW Gatekeepers required to process the GW signalingsignaling
GWs = max E1s per GW, BHCA, CPS (Calls GWs = max E1s per GW, BHCA, CPS (Calls per Second)per Second)
GKs = max CPS (check with vendor, not an GKs = max CPS (check with vendor, not an obvious figure to get, varies with each obvious figure to get, varies with each chassis/configuration/software release/DSP chassis/configuration/software release/DSP rev)rev)
Tips & tricksTips & tricks
Build GK redundancy by making sure Build GK redundancy by making sure all GWs have multiple GKs to reach. all GWs have multiple GKs to reach. HSRP can be very useful in conjunction HSRP can be very useful in conjunction with multiple GW->GK destinations.with multiple GW->GK destinations.
Make sure the GWs normalize the Make sure the GWs normalize the format of the called numbers so the format of the called numbers so the VoIP core deals with a single call VoIP core deals with a single call format (E.164 = country+city+local).format (E.164 = country+city+local).
Inter provider VoIP Inter provider VoIP servicesservices
What happens when you want to What happens when you want to extend the reach of your VoIP extend the reach of your VoIP services by interconnecting with services by interconnecting with other ITSP?other ITSP?
Tandem coding (VoIP->PSTN-Tandem coding (VoIP->PSTN->VoIP)>VoIP)
Open Settlement ProtocolOpen Settlement Protocol
Tandem CodingTandem Coding
In the case where a call is passed back from In the case where a call is passed back from the VoIP network to the PSTN and then the VoIP network to the PSTN and then resampled & compressed the call has been resampled & compressed the call has been sampled and compressed twice and sampled and compressed twice and therefore the call quality will degrade very therefore the call quality will degrade very rapidly.rapidly.
Examples:Examples:VoIP to GSM via the PSTN.VoIP to GSM via the PSTN.VoIP to the PSTN via another carrier with VoIP to the PSTN via another carrier with
compression gear.compression gear.Other VoIP carrier doesn’t want to “risk” Other VoIP carrier doesn’t want to “risk”
interconnects over VoIP (inter-ITSP QoS interconnects over VoIP (inter-ITSP QoS management issues)management issues)
Open Open Settlement Settlement ProtocolProtocol (OSP) (OSP)
Open Settlement Protocol (OSP), client-server protocol Open Settlement Protocol (OSP), client-server protocol defined by the ETSI TIPHON standards organization. defined by the ETSI TIPHON standards organization. Designed to offer billing and accounting record Designed to offer billing and accounting record consolidation for voice calls that traverse ITSP boundaries. consolidation for voice calls that traverse ITSP boundaries. It also allows service providers to exchange traffic with It also allows service providers to exchange traffic with each other without establishing multiple bilateral peering each other without establishing multiple bilateral peering agreements by using a 3rd party clearinghouse to enable agreements by using a 3rd party clearinghouse to enable extending the reach of their network.extending the reach of their network.
3rd party clearinghouse with an OSP server will allow 3rd party clearinghouse with an OSP server will allow services such as route selection, call authorization, call services such as route selection, call authorization, call accounting, and inter-carrier settlements, including all the accounting, and inter-carrier settlements, including all the complex rating and routing tables necessary for efficient complex rating and routing tables necessary for efficient and cost-effective interconnections. The OSP based and cost-effective interconnections. The OSP based clearinghouses provide the least cost and the best route-clearinghouses provide the least cost and the best route-selection algorithms based on the a wide variety of selection algorithms based on the a wide variety of parameters.parameters.
How How it worksit works Step 1: customer places call via the PSTN to a VoIP Step 1: customer places call via the PSTN to a VoIP
Gateway, which authenticates the customer by Gateway, which authenticates the customer by communicating with a RADIUS servercommunicating with a RADIUS server
Step 2: The originating VoIP gateway attempts to locate the Step 2: The originating VoIP gateway attempts to locate the termination point within it's own network by communicating termination point within it's own network by communicating with a gatekeeper using H.323 RAS. If there's no appropriate with a gatekeeper using H.323 RAS. If there's no appropriate route, the gatekeeper tells the gateway to search for a route, the gatekeeper tells the gateway to search for a termination point elsewhere.termination point elsewhere.
Step 3: The gateway contacts an OSP server at the 3rd party Step 3: The gateway contacts an OSP server at the 3rd party clearinghouse. The gateway establishes an SSL connection clearinghouse. The gateway establishes an SSL connection to the OSP server and sends an authorization request to the to the OSP server and sends an authorization request to the clearinghouse. The authorization request contains pertinent clearinghouse. The authorization request contains pertinent information about the call, including the destination number, information about the call, including the destination number, the device ID, and the customer ID of the gateway.the device ID, and the customer ID of the gateway.
Step 4: The OSP server processes the information and, Step 4: The OSP server processes the information and, assuming the gateway is authorized, returns routing details assuming the gateway is authorized, returns routing details for the possible terminating gateways that can satisfy the for the possible terminating gateways that can satisfy the request of the originating gateway.request of the originating gateway.
How it works (2How it works (2)) Step 5: The Clearinghouse creates an Step 5: The Clearinghouse creates an
authorization token, signs it with the certificate authorization token, signs it with the certificate and private key, and then replies to the and private key, and then replies to the originating gateway with a token and up to 3 originating gateway with a token and up to 3 selected routes. The originating gateway uses selected routes. The originating gateway uses the IP address supplied by the clearinghouse to the IP address supplied by the clearinghouse to setup the call.setup the call.
Step 6: The originating gateway sends the token Step 6: The originating gateway sends the token it received from the settlement server in the it received from the settlement server in the setup message to the terminating gateway.setup message to the terminating gateway.
Step 7: The terminating gateway accepts the call Step 7: The terminating gateway accepts the call after validating the token and completes the call after validating the token and completes the call setup.setup.
Voice Speech Quality Voice Speech Quality (VSQ)(VSQ)
MOS: ITU P.800 & P.830, scale from 1 (bad) to 5 MOS: ITU P.800 & P.830, scale from 1 (bad) to 5 (excellent), based on human perception (excellent), based on human perception (subjective), most widely used by VoIP vendors (subjective), most widely used by VoIP vendors when comparing codec quality, the oldest model.when comparing codec quality, the oldest model.
PSQM (Perceptual Speech Quality Measurement), PSQM (Perceptual Speech Quality Measurement), ITU P.861, compares input and output speech ITU P.861, compares input and output speech (automated), developed by KPN Research(automated), developed by KPN Research
PAMS (Perceptual Analysis Measurement System), PAMS (Perceptual Analysis Measurement System), Developed by British Telecom, “Objectively” Developed by British Telecom, “Objectively” predict results of subjective speech quality testspredict results of subjective speech quality tests
PESQ (Perceptual Evaluation of Speech Quality) PESQ (Perceptual Evaluation of Speech Quality) ITU P.862, latest standard (January 2001), ITU P.862, latest standard (January 2001), currently the most accurate model for automated currently the most accurate model for automated voice quality perception, improves over PSQM and voice quality perception, improves over PSQM and PAMSPAMS
Sources of potential VSQ Sources of potential VSQ problemsproblems
Delay jitter: variance in delay (zero, little or Delay jitter: variance in delay (zero, little or excessive delay)excessive delay)
Encoding and decoding of voice Encoding and decoding of voice (PCM/ADPCM/low bit-rate codecs/CLEP)(PCM/ADPCM/low bit-rate codecs/CLEP)
Time-Clipping (Front end clipping) introduced by Time-Clipping (Front end clipping) introduced by Voice Activity Detectors (VAD)Voice Activity Detectors (VAD)
Temporal signal loss and dropouts introduced by Temporal signal loss and dropouts introduced by packet lesspacket less
Environmental noise, including background noiseEnvironmental noise, including background noise Signal attenuation and gain/attenuation variancesSignal attenuation and gain/attenuation variances Level clippingLevel clipping Transmission channel errorsTransmission channel errors
Echo: What makes it a Echo: What makes it a problem?problem?
An An analog leakageanalog leakage path path between analog Tx and Rx paths between analog Tx and Rx paths
SufficientSufficient delay delay in echo return in echo return Sufficient Sufficient echo amplitudeecho amplitude
When all of the following conditions are true, echo is perceived as annoying:
How the packet voice How the packet voice impact on echo impact on echo
perception ?perception ?
Bits don’t leak - Echo is not introduced on digital linksBits don’t leak - Echo is not introduced on digital links The packet segment of the voice connection introduces a The packet segment of the voice connection introduces a
significant delay (typically 30 ms in each direction). significant delay (typically 30 ms in each direction). The introduction of delay causes echoes (from analog tail The introduction of delay causes echoes (from analog tail
circuits) that are normally indistinguishable from side tone to circuits) that are normally indistinguishable from side tone to become perceptible.become perceptible.
Because the delay introduced by packet voice is unavoidable, the Because the delay introduced by packet voice is unavoidable, the voice gateways must prevent the echovoice gateways must prevent the echo..
WAN PSTN PSTN
Low delay, potential echo sources
Large delay,no echo sources
Identify and Isolate the Identify and Isolate the echo problemecho problem
IdentifyIdentify the echo problem. Which side the echo problem. Which side hears echo? Calls to which numbers hears echo? Calls to which numbers hear echo ?hear echo ?
IsolateIsolate the problem as much as the problem as much as possible and try to find a scenario possible and try to find a scenario where the echo is reproducible.where the echo is reproducible.
Whenever I hear echo, the problem is Whenever I hear echo, the problem is at the OTHER end !!at the OTHER end !!
Basic securityBasic security GWs/GKs w/ACLs with source ip (yes, can be GWs/GKs w/ACLs with source ip (yes, can be
spoofed) appears to be the #1 source of spoofed) appears to be the #1 source of protection against un-authorized calls.protection against un-authorized calls.
Run your VoIP network isolated from any Run your VoIP network isolated from any public network using your prefered flavor public network using your prefered flavor (physical seperation, VLAN, MPLS, etc..)(physical seperation, VLAN, MPLS, etc..)
VoIP packets are _not_ encrypted, if this is an VoIP packets are _not_ encrypted, if this is an issue used IPSec! Beware that software issue used IPSec! Beware that software crypto will add delay and jitter, use hardware crypto will add delay and jitter, use hardware crypto for better performance (should add crypto for better performance (should add predictable delay and jitter)predictable delay and jitter)
Note: CRTP doesn't work with IPSec, Note: CRTP doesn't work with IPSec, remember this when designing the bandwidth remember this when designing the bandwidth budget.budget.
Questions?Questions?