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