qos and admission probability study for a sip-based central managed ip telephony system

Presentación

Jose Saldana

Jenifer Murillo

Julián Fernández Navajas

José Ruiz Mas

Eduardo Viruete Navarro

José I. Aznar

O M U N I C A C I O N E S

TRUPO DE

CECNOLOGÍAS

GDE LAS

CPS - University of Zaragoza, Spain

<>

http://www.ntms-conf.org/


INTRODUCTION RELATED WORKS SYSTEM ARCHITECTURE SYSTEM IMPLEMENTATION TESTS AND RESULTS CONCLUSIONS

Index

NTMS February 7-10, 2011. Paris QoS and Admission Probability for a SIP-based Telephony System

4

Introduction

- VoIP is replacing traditional telephony systems.

- Software-based solutions allow a simple PC to assume the role of the PBX.

- This is interesting for SME that want to avoid the costs of proprietary systems.

INTRODUCTION RELATED WORKS SYSTEM ARCHITECTURE SYSTEM IMPLEMENTATION RESULTS CONCLUSIONS


5

Introduction

- VoIP is a real-time service, but it uses a network designed for best effort services.

- Users demand a QoS similar to PSTN.

- Need of solutions to add quality to IP networks.

- Overprovisioning is not the best solution.

- Call Admission Control (CAC) can be used.



6

Introduction

- The offices are grouped into countries and geographical zones.


Country 1

Country 2

Country 3

Country 4

Country 5

Zone 1

Zone 2

Zone 3

IP network

PSTN


7

Introduction

- Central managed system: connection of different offices via IP, and sharing lines between offices. Cost savings by establishing calls from the GW of the destination country.


PSTN

IP networkOffice 1 Office 2

GatewayGateway

IP call Local call

Country 1 Country 2


8

Introduction

- Control element: Local agent (SIP proxy)


IP network

Data centre

PBX

Local

agent

Local

agentLocal

agent

Office 1 Office 2

Office i

Gateway

Gateway

Gateway


9

Introduction

- Control element: Local agent (SIP proxy)


IP network

Data centre

PBX

Local

agent

Local

agentLocal

agent

Office 1 Office 2

Office i

Gateway

Gateway

Gateway


10

Introduction

- SIP proxy allows the CAC to be seamlessly integrated into the system.

- The PBX and the terminals do not have to be modified.

- SIP Redirect messages can be used to decrease blocking probability.

- A SIP proxy does not require a high processing capacity.



11

Motivation of this work

Study the system in terms of

- QoS parameters

- OWD (One Way Delay)

- Packet Loss

- Jitter

- R-factor

- Establishment delay

- Admission probability



12

R-factor - Defined by ITU G.107 (E-Model)

- Ranges from 0 (bad quality) to 100 (good)

- Acceptable for R > 70

- Dependence on delay and packet loss

- Widely accepted quality estimator for VoIP services



Index


14

Signaling protocols

- H.323: Used by many proprietary solutions.

- SIP (Session Initiation Protocol): Becoming very popular. Many open-source PBX use it.

- SIP proxies: Used to add scalability, transferring workload from the network core to the borders.



15

CAC systems

- Measurement-based: Use the state of the network to take admission decisions

- Parameter-based: Some measurements are carried out during system’s set up, and a maximum number of calls is set.



16

- «Rule of the thumb»: Bandwidth-delay product.

- «Stanford model»: Division by sqrt(N) (N:number of TCP flows).

- Other proposal: time-limited buffer. Interesting for this work. Limits OWD. But penalizes big packets.

Buffer size and buffer policies



Index


18

- Offices in different countries

- Dial plan only at the PBX

- Internet used for VoIP traffic

- The system does not use any reservation protocol

- VoIP traffic is the only real-time one we are going to take care of in a special way

- A parameter-based CAC is used

IP Telephony system



19

- All signaling messages pass through it

- Counts the number of calls

- In charge of admission decissions

Local Agent


INVITE

100 Trying

INVITE

100 Trying

183 Session

Progress

183 Session

Progress

INVITE

480 Temporarily

Unavailable

480 Temporarily

Unavailable

480 Temporarily

Unavailable

IP phone IP phoneLocal agent Local agent

Office 1 Office i

database

PBX


20

- All signaling messages pass through it

- Counts the number of calls

- In charge of admission decissions

Local Agent


INVITE

100 Trying

INVITE

100 Trying

183 Session

Progress

183 Session

Progress

INVITE

480 Temporarily

Unavailable

480 Temporarily

Unavailable

480 Temporarily

Unavailable

IP phone IP phoneLocal agent Local agent

Office 1 Office i

database

PBX

There is no place for this call


21

Different call types


IP network

Office 1 Office 2

Gateway Gateway

Type 1

Type 2

Type 4 Type 6

Type 5

Type 3

User in a country

without office


22

- Calls to PSTN can be redirected if there are no available lines in a gateway

Redirected calls


INVITE

100 TryingINVITE

100 Trying

183 Session

Progress

183 Session

Progress

INVITE

302 Moved

Temporarily

IP phone SIP proxy SIP proxy Gateway

INVITE

SIP proxy

database

Gateway

INVITE

100 Trying

100 Trying

Offic

e 1

PBX

Offic

e i

Offic

e j

database


23


Redirected calls


INVITE

100 TryingINVITE

100 Trying

183 Session

Progress

183 Session

Progress

INVITE

302 Moved

Temporarily


INVITE

SIP proxy

database

Gateway

INVITE

100 Trying

100 Trying

Offic

e 1

PBX

Offic

e i

Offic

e j

database

There is no place for this call


24


Redirected calls


INVITE

100 TryingINVITE

100 Trying

183 Session

Progress

183 Session

Progress

INVITE

302 Moved

Temporarily


INVITE

SIP proxy

database

Gateway

INVITE

100 Trying

100 Trying

Offic

e 1

PBX

Offic

e i

Offic

e j

database

Can it be established from another office?


25


Redirected calls


INVITE

100 TryingINVITE

100 Trying

183 Session

Progress

183 Session

Progress

INVITE

302 Moved

Temporarily


INVITE

SIP proxy

database

Gateway

INVITE

100 Trying

100 Trying

Offic

e 1

PBX

Offic

e i

Offic

e j

database

Try Office j


26


Redirected calls


INVITE

100 TryingINVITE

100 Trying

183 Session

Progress

183 Session

Progress

INVITE

302 Moved

Temporarily


INVITE

SIP proxy

database

Gateway

INVITE

100 Trying

100 Trying

Offic

e 1

PBX

Offic

e i

Offic

e j

database

Accept


Index


28

- Xen Virtualization-based testbed

- Each computer is translated into a VM

- Bandwidth of office’s routers emulated with Linux tc (Traffic Control)

- Codec used: G.729a with 2 samples per packet

Testbed


Virtual network

Physical

machine

xenbr0

Public IP address


29

- Off-the-self tools:

- SIP Proxy: OpenSIPS 1.4

- PBX: Asterisk 1.6.0.1

- Softphone: PJSUA 1.0

- Gateways: Emulated with PJSUA 1.0

- Admission probability: We need a bigger scenario (Testbed has size limitations). Matlab simulations.

Software tools


pj


Index


31

- Generator: D-ITG

- Network delays and dejitter buffer effect are added offline

QoS measurements


Traffic

Generation

RouterTraffic

Capture

Real Traffic in a testbed

Network

delays

+

Dejitter

buffer

Offline post-processing

Traffic

Trace

Final

Results

VoIP

Background

Buffer

policies


32

- Size distribution

- 50% 40 bytes

- 10% 576 bytes

- 40% 1500 bytes

- Only UDP, in order to avoid flow control: always the same background traffic.

- Different rates to saturate the access router

- Network does not loose packets

Background traffic



33

- Step-like graphs

- When the bandiwidht is not enough, QoS falls dramatically

High capacity buffer


70

75

80

400 450 500 550 600 650 700 750 800 850 900 950 1000

R-f

acto

r

background traffic (kbps)

R-factor

1 call

5 calls

10 calls

15 calls

20 calls


34

- The graphs present a slope

- Acceptable R values are obtained with more background traffic

Time-limited buffer (60ms)


60

65

70

75

80

85

400 450 500 550 600 650 700 750 800 850 900 950 1000

R-f

acto

r


R-factor 1 call

5 calls

10 calls

15 calls

20 calls


35

- Limits the maximum delay

- But increases packet loss for BG traffic



0

5

10

15

20

25

30

35

40

45

1 2 3 4 5 6 7 8 9 10 11 12 13 14 15

% p

acke

t lo

ss

Number of calls. Background traffic=800kbps

Packet loss of each trafficRTP

1500 bytes

572 bytes

40 bytes


36

- VoIP traffic is protected because of its small size



0

100

200

300

400

500

600

700

800

900

1000

1 2 3 4 5 6 7 8 9 10 11 12 13 14 15

Ban

dw

idth

(kb

ps

at e

th le

vel)

Number of calls. Background traffic=800kbps

Bandwidth of each trafficRTP1500 bytes572 bytes40 bytes


37

- OWD presents a limit



75

100

125

150

175

400 450 500 550 600 650 700 750 800 850 900 950 1000

OW

D (

ms)


One Way Delay1 call

5 calls

10 calls

15 calls

20 calls


38

- Packet loss grows indefinitely



0

1

2

3

4

5

6

400 450 500 550 600 650 700 750 800 850 900 950 1000

pe

rce

nta

ge o

f d

isca

rde

d p

acke

ts


Percentage of packet loss1 call

5 calls

10 calls

15 calls

20 calls


39

- Jitter (IPDV)



2

3

4

5

6

7

8

9

10

400 450 500 550 600 650 700 750 800 850 900 950 1000

IPD

V (

ms)


IPDV1 call

5 calls

10 calls

15 calls

20 calls


40

- Redirecting calls can increase this delay.

- Measured with simulation: From INVITE to arrival at the destination.

- Considered delays:

- Network delay at the LANs: Negligible

- Processing time: Proxy and PBX

- Queuing delay at the router

- Network delay at the WAN

Establishment delay



41

Establishment delay


INVITE

INVITE

INVITE

302 Moved

IP phone SIP proxy SIP proxy

INVITE

INVITE

Offic

e 1

PBX

Offic

e i

Offic

e j

Proc. Proxy

Queuing

Network

Proc. PBX

Network

Proc. Proxy

Queuing

Network

Proc. PBX

Network

Proc. Proxy

IP phone

Processing delays


42

Establishment delay


INVITE

INVITE

INVITE

302 Moved


INVITE

INVITE

Offic

e 1

PBX

Offic

e i

Offic

e j

Proc. Proxy

Queuing

Network

Proc. PBX

Network

Proc. Proxy

Queuing

Network

Proc. PBX

Network

Proc. Proxy

IP phone

Queuing delays


43

Establishment delay


INVITE

INVITE

INVITE

302 Moved


INVITE

INVITE

Offic

e 1

PBX

Offic

e i

Offic

e j

Proc. Proxy

Queuing

Network

Proc. PBX

Network

Proc. Proxy

Queuing

Network

Proc. PBX

Network

Proc. Proxy

IP phone

WAN delays


44

- Different RTT values

- Independent of the number of offices

Establishment delay


50

100

150

200

250

300

350

25 50 75 100 125

Esta

blis

hm

en

t d

ela

y (m

s)

RTT (ms)

Establishment delay 2 offices

4 offices

6 offices

8 offices


45

- Matlab simulations

- Each office has 25 users

- Gateways have 6 lines

- CAC limit=6 (variable in some tests)

- Different values of and number of offices

Admission probability


... ...

IP network

M1 M2AI12AP1

AO1

AI21 AP2

AO2

N1 N2

PSTN PSTN

...

...

...

...

Office 1 Office 2


46



80

85

90

95

100

3 3,5 4 4,5 5

Pe

rce

nta

ge o

f ad

mit

ted

co

nfe

ren

ces

(%)

λ (conferences per hour per user)

Admission probability 3 offices

5 offices

7 offices

10 offices

15 offices


47



80

85

90

95

100

3 3,5 4 4,5 5

Pe

rce

nta

ge o

f ad

mit

ted

co

nfe

ren

ces

(%)

λ (conferences per hour per user)

Admission probability 3 offices

5 offices

7 offices

10 offices

15 offices

Increasing the number of offices is beneficial


48

- Influence of CAC limit



80

85

90

95

100

1 2 3 4 5 6 7 8 9 10 11 12 13

Ad

mis

sio

n p

rob

abili

ty (

%)

CAC limit

Admission probability2 offices and isolated mode

2 offices and sharing mode

4 offices and isolated mode





49




80

85

90

95

100

1 2 3 4 5 6 7 8 9 10 11 12 13

Ad

mis

sio

n p

rob

abili

ty (

%)

CAC limit







Sharing mode is better than isolated mode


50




80

85

90

95

100

1 2 3 4 5 6 7 8 9 10 11 12 13

Ad

mis

sio

n p

rob

abili

ty (

%)

CAC limit







Increasing the number of offices is beneficial


51




80

85

90

95

100

1 2 3 4 5 6 7 8 9 10 11 12 13

Ad

mis

sio

n p

rob

abili

ty (

%)

CAC limit







In Sharing mode, the increase of CAC limit improves the Asmission probability


Index


53

- A SIP-based IP telephony system has been designed and tested.

- SIP proxies are used in order to implement a CAC.

- QoS measurements show the better performance of a time-limited buffer.

- Establishment delay does not depend on the number of offices.

- Sharing the gateways improves admission.

Conclusions


Presentación

Jose Saldana

Jenifer Murillo

Julián Fernández Navajas

José Ruiz Mas

Eduardo Viruete Navarro

José I. Aznar

O M U N I C A C I O N E S

TRUPO DE

CECNOLOGÍAS

GDE LAS

CPS - University of Zaragoza, Spain



55

Xen Virtualization



56

PJSUA interface


qos and admission probability study for a sip-based central managed ip telephony system

Education

sipbased telephony system

introduction sip proxy

sip proxies

blocking probability

sip redirect messages

admission control cac

softwarebased solutions

traditional telephony