QoE Assurance Division

Solutions for ToIP – VoIP service assurance and service performance

Problems to solve Telephony and voice over IP solutions are launched as enterprises

are migrating their PABX to IP based PBXs

IP VPNs or core IP networks are being stressed with more and more VoIP traffic

Serious issues may happened as IP networks have not being designed for such real time traffic

Carriers/SPs are trying to solve the issues by deploying QOS in their networks

Symmetricom VoIP/ToIP solution helps SPs, enterprises to assess and monitor the quality of the network (IP SLAs) as well as verifying the service availability

Symmetricom solution benefits Independent and integrated network performance

monitoring and service performance monitoring solution for:

Assessing network performance before deployment, verifying IP QoS (Quality Of Services) and COS (Class Of Traffic) performances by stressing WAN links or IP backbone by

– Injecting RTP traffic with different TOS/DiffServ settings in order to monitor performance improvements or lack of performances and issues diagnosis

– Monitoring critical KPIs (Key Performance Indicators) such as OWD (one way delay), PDV (packet delay variation), PL (packet Loss) as required for carrying voice traffic over IP as defined in ITU Y1540/41 and IETF RFC 2330

Monitoring IP SLSs for IP VPN service providers by monitoring above KPIs

Monitoring VoIP traffic (monitor and score customer traffic – voice calls) for tracking VoIP usage and performances

Stimulating SIP signaling of hosted IPBX or IP-Centrex and score service performance and service availability

Active SIP testing Part 1

Solution main features

Q-xxx network probe acts as both SIP UAS and SIP UAC Multiple UAS per Q-xxx (simulates an end user) Supported codecs include: G.711a, G.711u, G.729 and G.723 with

audio (MOS) scoring using ITU G.107 model

SIP supported transport layers: UDP or TCP Supported SIP commands:

REGISTER (with time out) – several SIP servers supported INVITE and RE-INVITE TRANSFERT BYE

Encryption: MD5 digest Support TOS value in RTP packets when SIP test runs (see part 2)

Easy test setup using Excel spreadsheets or XML templates

Monitoring REGISTERBackbone NetworkMain Office

• Register• Invite• MOS• BYE• Response Code

IPBX, SIPProxy/register

Monitoring

Quality Monitoring Quality Monitoring

PSTN

Remote Office

• MOS(passive)

• MOS(passive)

Proxy Server• Register• Invite• MOS• BYE• Response Code

NetworkProbes

NetworkProbes

REGISTERREGISTER

A Q-101 probe can be used to simulate end user calls and monitor IPBXs as well all Registrar proxy

Monitoring call performancesBackbone NetworkMain Office

• Register• Invite• MOS• BYE• Response Code

IPBX, SIPProxy/register

Monitoring

Quality Monitoring Quality Monitoring

Remote Office

• MOS(passive)

• MOS(passive)

Proxy Server• Register• Invite• MOS• BYE• Response Code

NetworkProbes

NetworkProbes

INVITE

INVITE

PSTN

Once a Q-101 probe has been successfully register, according to the test template, the probe can simulate user calls and traffic and monitor the signaling network as well as the data network and their performances

Rating voice qualityBackbone NetworkMain Office

• Register• Invite• MOS• BYE• Response Code

IPBX, SIPProxy/register

Monitoring

Quality Monitoring Quality Monitoring

Remote Office

• MOS(passive)

• MOS(passive)

Proxy Server• Register• Invite• MOS• BYE• Response Code

NetworkProbes

NetworkProbes

MOS

MOSMOS

MOS

MOS

MOS

PSTN

Using ITU G107 MOS standard, voice quality is rated (active test between probes or passive )

OSS integrationBackbone NetworkMain Office

• Register• Invite• MOS• BYE• Response Code

IPBX, SIPProxy/register

Monitoring

Quality Monitoring Quality Monitoring

Remote Office

• MOS(passive)

• MOS(passive)

Proxy Server• Register• Invite• MOS• BYE• Response Code

NetworkProbes

NetworkProbes

BYE

BYE

ReportReport

SNMP TRAP/ ALARMS

PSTN

KPIs are monitored as well as cause of failure, alarms or SNMP traps can be used to alert network administrator – SLS can be setup per codec type (up to 16 SLS)

OSS integration

24/7 Service Monitoring

Provide statistics per location (probe)

02 January 2007

SIP Servers

Provide statistics per serveror per location (probe)

NOC dashboards

Real time monitoring per probe or location

Support unlimited number of SIP servers

Per Server, Per Call Diagnostics

Per SIP server diagnostics, per cause of failure

Per call diagnostics

Alarms and Errors

Alarms and SNMP traps are generated when an error condition occurs:

INVITE failure condition REGISTER failure condition RTP Call Session failure condition Reported error code MOS value below threshold KPIs monitoring

Test Setup – Excel or HTML

Test Definition Template

Version 2 (Do not change this value)

Test Type 2 (1=DATA, 2=VOIP, 3=IPTV)

Scope 2 (0=Private, 1=Group, 2=Public)

App Type SIP (VFACTOR or SIP)

Test Name Hitachi SIP Test (max 80 chars) must be unique!

Test Description To Test and Monitor Hitachi SIP Server(max 80 chars)

Test Duration 230 (in minute)

Result Upload 15 (in second)

Repeat Count 1 (-1=infinite, valid for 1 to n)

Repeat Interval 0 (in minute)

Sip Session Definition

Sip Server 220.110.155.190 (max 64 chars)

Server Port 5060

Transport Protocol 1 Valid = 1(UDP) or 2(TCP)Note:

Register Refresh Interval 30 (valid >= 1 min) Codec Type Sample Size0(PCMU) 10

Invite Refresh Interval 3 (valid >= 1 min) 8(PCMA) 2018(G729) 10

SLA Mask 0Jitter Buffer Length is validated based on the "Sample Size" of the codec specified.

TOS 0 The min value is the "Sample Size" and the max value is the 1000x"Sample Size".Also, the value need to be a multiply number of the "Sample Size"

Call List

Call List Id Call Duration (min) Delay Between Calls (min) Call Address 1 .. NCall List 1 2 1 6003@220.110.155.190 6100@220.110.155.190 6101@220.110.155.190

Sip User List (Note: Codec Type = 0(PCMU), 8(PCMA) or 18(G729))

IP Address Username Password Sip Port RTP Port Codec Type Num of Codec Sample Jitter Buffer Length Test Start Delay Call List Id172.16.1.182 6002 432178 5060 9900 0 1 40 0 Call List 1

SIP server instantiation

UAS/UAC call list

UAS/UAC definition

Active RTP monitoring Part 2

VoIP backbone

In order to assure voice quality, IP backbone monitoring is required as defined in ITU Y1540 and Y1541

KPIs that must be monitored and matched against performance indicators are: OWD PDV PL

Solution must also test DiffServ/TOS settings in order to monitor specific COS (Class Of Services)

Site A

Site B Site C

Site B

Site A

Hosted servicesRTP •codec,•TOS,.

NetworkProbes

Monitoring RTP backbone

RTP SLS, RTP Performance(Delay, Jitter, Loss)

TOS 1TOS 1 TOS 2TOS 2

OSS integration

SLS and Alarms monitoring

RTP KPIs monitoring dashboard

PDV monitoring

PL monitoring

Delay monitoring

Scheduling and reporting

Test scheduling Reports scheduling (RTP only) are defined through templates and forwarded as pdf to a list of email

A VoIP lifecycle solution

For deployment/pre deployment Assess Network Performance per ITU Y1540/41 Assess Network Quality Of Service for real time VoIP traffic readiness

Accelerate deployments Diagnostics/Troubleshooting VoIP network troubleshooting (dissociate issues coming from the network and issues

coming from appliances such as NAT devices, proxy,..) Codec and network QoS mismatch (diagnose impacts of network COS on various

type of codec)

For operations Monitor VoIP service availability and service performance Monitor service levels of actual VoIP calls Hosted services VoIP KPIs monitoring IP Centrex service monitoring Voice backbone monitoring

AssureEnableAccelerate

Conclusions

Assess infrastructure and Network Quality Of Service for real time VoIP traffic readiness.

Enable new service deployment - Hosted services VoIP, IP Centrex

Assure service availability and service performance