Open-‐‑Source Based Prototype for QoS Monitoring and QoE Estimation in Telecommunication Environments
Sebastian Schumann Slovak University of Technology
Bratislava, Slovakia
Cardiff, UK – 15. September 2011
Introduction • Implementation for Quality of Service (QoS) and
Experience (QoE) monitoring • Works in Real-time Transport Protocol (RTP) based
telecommunication environments • Analysis
o QoS parameters are evaluated o QoE is determined with the E-Model
• Output o R-Factor, one-way delay, packet-loss probability o Graphical representation
Environment • Usage of Voice over IP (VoIP) increased over the last
years • It is not always possible to enforce QoS, esp. in
unmanaged networks • Size of measured network does not matter • Measurement system
o Measurement points (probes) are distributed o Central reporting unit collects and evaluates the data
• Focus on widespread networks, not system components
Motivation • ngnlab.eu targets distributed VoIP environments
and open-source based solutions • Commercial solutions are expensive, only for
operators • Main goals
o Easy but flexible measurement design o A non-intrusive online monitoring o Informative results o Ability to determine the geographical and technical
source of degradations
“Competition”
Theory • E-model used to determine QoE (calculated acc.
several network parameters) • Objective (i.e., calculated) value can be mapped
to the subjective Mean Opinion Score (MOS) • Impacts on speech quality are
o One-way delay o Packet-loss probability o Packet-loss distribution o Speech codec
• Measurement and evaluation of values allow calculation of QoS/QoE during the call
Correlation between MOS value and R-‐‑Factor
Measured Impairments I • One-way delay • Measured by halving
the Round-Trip-Time (RTT) value of the voice packets (estimation)
• Both directions possible • RTT determination using
measured values o Time-stamp in PCAP o Time-stamp in RTCP
• RTT1=A2-A1-D2 • RTT2=A3-A2-D3
DLSR .. delay sender report A1 .. 1st SR passes ME A2 .. following SR D2 .. DL btw reception of SR1 and transmission of SR2
Measured Impairments II • Packet loss probability • Determined by recording the sequence number of
each RTP packet that passes the ME • The loss probability is updated after every 100 RTP
packets o The time distance is a good balance between the applied
load on the ME, the network load, and the actuality of the measurement results on the EE
Measured Impairments III • Packet loss distribution calculated acc. the patent
of McGowan o Overall packet loss probability (Ppl) o Average length of all loss sequences
• Speech codec is determined by parsing the Session Description Protocol (SDP) during the session establishment procedure
• Knowledge is important in relation to the used compression method and its robustness against packet loss (packet loss robustness factor)
Network setup
Application • Measurement probes
o PCAP library captures packet for analysis o Perl script extracts required information from each packet o HTTP is used to exchange measured parameters
• Central reporting unit o Java application o Real-time monitoring with three detail levels (monitoring
unit, call, details) o Adjustable color indication when pre-set thresholds are
reached
GUI
Measurement setup
Results I • Non-degraded
measurement • Normal values • Delay in path 2+4 high
due to public network
Results II • Degraded
measurement • One-way delay on the
Internet higher (20x) in paths 2+4
• R-Factor decreased as well
• Knowing network and taking packet loss into account, low upload on office B is determined
Summary • QoS and QoE can be measured using the designed
prototype • Implementation is scalable to smaller or larger Telco
networks (probes can be distributed accordingly) • Implementation can compete with professional
equipment to a certain extent • Extensions open but easily possible
o Alarms o Visual network status display in real-time o Follow-up calls for neg. quality calls o Recording of call samples possible as well
Recommended
