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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 [email protected] [email protected] [email protected]
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