Test Plan for VoIP

Slide 2

Test Plan Main Objectives

Validate functionality
Compliance with standards
Performance verification
Price/performance evaluation tool

This should take place before commercial
deployment

Slide 3

Functionalities

Verifies that all functions work
properly

Functionalities

Under Stress

Verifies that all functions work
properly under stress

Falt-Insertion

Test

Verifies that the system reacts as
expected in non-legal conditions

Long-Term

Stability

Verifies that all functions work properly and
are consistant in long term stability test

Performance

Test

Verify the performance of the system
which should comply with the system
requirements

Phase 1

Phase 2

Test Strategy

Testing VoD systems involves five aspects grouped into two

phases

Slide 4

Enterprise

SS7
Network

MGC

IP/ATM Network

SS7
Network

Home/SOHO

MGC

SIP -T

RTP

MGC

Class 5
Switch

Signaling
GW

SIGTRAN

RTP

RAS

Billing System

Class 5
Switch

NMS

SIGTRAN

Typical VoIP Architecture

Slide 5

What has to be Tested?

GW and MG
GK and MGC
Signaling GW
Interactive voice response and voice mails
Billing and prepaid system
Network Management System (NMS)

03:24:10 PM

Slide 6

Some Considerations

Number of anticipated users
Estimated traffic per user
Compression method to be used
Packet structure
Network performance: latency, packet loss,
available bandwidth
The infrastructure: Frame Relay, ATM, VSAT,
XDSL, WLL

Slide 7

Criteria and Assumptions

End user behavior according to Poisson
statistic model
Average call duration is 180 seconds
End user traffic: 0.01–0.15 Er

Slide 8

Definitions

Blocking: a percentage that refers to the calls that get
a busy signal because all lines are in use
Busy Hour Traffic (BHT): the amount of call traffic
handled by a group of phone lines during the busiest
hour of the busiest day for your system (defined in
units of Erlangs or CCS)
Centi-Call Seconds (CCS): a unit of BHT also commonly
used for traffic measurement (36 CCS equals 1 Erlang
of traffic)
Erlang: a unit of BHT which represents the continuous
use of a single line for 1 hour e.g., 30 calls of 2 minutes
holding time each would equal 1 Erlang of traffic

Slide 9

Erlang B Calculations

Purpose: to calculate number of lines
(trunks)
BHT = (number of anticipated end users) x
0.05
Blocking = Required Grade of Service/100 =
number of failed call attempts/total number
of call attempts
For Erlang calculations,refer to:

http://www.erlang.com/calculator/

Slide 12

Avoiding Bottlenecks

T1 supports up to 18 Er with GOS = 5%
E1 supports up to 24.8 Er with GOS = 5%
N(t) = 18 Er/0.05 Er = 360 customers
N(e) = 24.8 Er/0.05 Er = 496 customers
Simultaneous calls can be made according to the
number of trunks e.g., 24/23/30 (T1-CAS/T1-PRI/E1-
PRI)
However, the limitation will be derived from two
other factors:

Compression method
Guaranteed bandwidth

Slide 13

Stress Conditions

400 RTP calls per session and 120 setup
calls per second
Create jitter/packet loss/miss order packets
and latency in uniform mode
VAD and silence suppression mechanism
should be activated
RTP packets will consist of 1 frame per
packet and 3 frames per packet

Slide 14

Gateway (1)

Functionality, operation under stress, QoS and

GoS

Compression and decompression
Bandwidth utilization
Silence suppression and VAD
DTMF detection and generation
Jitter suppression and echo cancellation
Fall-back to PSTN mechanism
Alternative rerouting mechanism
IVR for 2-stage dialing
Voice quality evaluation

Slide 15

Gateway (2)

Fax over IP: Packet Loss Recovery
Mechanism

Redundant packet transmission (T.38)
Resent mechanism (TCP/IP)
Forward Error Correction (FEC)

FAX/voice switch over mechanism

Slide 16

Gateway Testing

QPro

InterSim

RTP

323Sim

HUB

QPro

HUB

MediaPro

Slide 17

Gatekeeper

Registration mechanism
Admission and authorization mechanism
Routing directory service:

Least cost routing
Best cost routing

H.225 (RAS) compliancy
Routed mode operation
RSVP and bandwidth allocation

Slide 18

Gatekeeper Testing

InterSim

RTP

RAS

HUB

MediaPro

323Sim

Slide 19

IVR: Interactive Voice Response

DTMF integrity
Test under stress
Recorded playback voice quality (voice mail)
Reliability tests for voice recognition
mechanisms
Testing the effects of latency, jitter, packet
loss and packets sent in the wrong order

Slide 20

IVR Testing

InterSim

RTP

323Sim

HUB

QPro

HUB

MediaPro

IVR/Voice-
Mail

Slide 21

Billing and Prepaid Systems

Call Detail Record (CDR) integrity

CLID
Call duration
Called ID
Personal Identification Number (PIN)

Functionality under stress
Prepaid number verification
Charging system integrity

Slide 22

Billing/Prepaid System Testing

InterSim

RTP

RAS

HUB

MediaPro

323Sim

CDR

Billing/Prepa
id System

Slide 24

NMS Testing

InterSim

RTP

RAS

HUB

MediaPro

323Sim

NMS

Slide 25

Testing Capabilities: Future Plans

SIP environment testing

Proxy server
Soft-switch
Protocol mediators
IP PABX
Redirect server
MGC

Signaling gateway

SIGTRAN decoding and statistics

Add-on services and unified messaging testing

Document Outline