1 understanding voip from backbone measurements marco mellia, dario rossi robert birke, and michele...
TRANSCRIPT
1
Understanding VoIP from Backbone Measurements
Marco Mellia, Dario Rossi Robert Birke, and Michele PetraccaINFOCOM 07’, Anchorage, Alaska, USA
Young J. Won
Oct. 15, 2007
2
Outline
• Introduction
• Measurement methodology
• The FastWeb network
• Measurement results
• Conclusion
3
Introduction
• VoIP has long been indicated as the technology that will trigger convergence.
• Traffic monitoring and characterization have been seen as a key methodology to understand telecommunication technology and operation– This paper presents the first extended set of measure
ment results collected via passive monitoring of VoIP traffic
4
Measurement Methodology
• Identification of RTP/RTCP over UDP flows
• Measurement indexes– Call duration– Call round trip time– Flow packet loss probability– Flow jitter (Inter-Packet-Gap variation)– Flow equivalent mean opinion score (eMOS)
A computational model by ITU-T the predicts subjective quality of packetized voice.
5
Why?
• Why we are interested in this?– Curiosity– Extracting basic parameter values for
OPNET and later modeling– Basis for the theorical scenario analysis
models
6
Identification of RTP/RTCP over UDP Flows
• Conditions to Check– The version field must be set to 2– The payload type field must have an admissible value or the same
SSCR (Synchronized Source Identifier)– The UDP port > 1024 or the same payload type
8
Measurement Summary• July 15, 2006 - 10 am to 2 pm (4 hr)
– 240GB packet header– 150,000 phone calls, RTP and RTCP over UDP
• Voice transport, G.711a Codec– Two 64kbps streams, 50 packets per sec– Packetization time is set to 20 ms
• No per-class differentiation
• The maximum values are observed between 10 am to 2 pm– More than 1300 simultaneous calls per minute– Drops in lunch break and in the early afternoon
9
Measurement Results
• Two probe nodes located in a PoP located in Turin, and a Gateway node located in Milan
• Tstat was run on the probe to take live traffic measurements– Passive monitoring tool by the Politecnico di Torino– RTP/RTCP over UDP or tunneled TCP
10
HAG Distribution
• Home Access Gateway (HAG)– Offers Ethernet ports to PCs, VideoBox, Plain Old Telephone
12
User-Centric Measurements (Call Duration)
• User-Centric measurements– Average phone duration, 106s: heavy-tailed distribution– Long (97s), Local (113s)– My observation: Very similar to FLOW Lifetime measurement
13
eMOS CDF
• eMOS– Excellent ( eMOS>4 )– Good ( eMOS[3:4] )
• eMOS >= 3.6– The same quality as trad
itional PSTN phone calls
14
Network-Centric Measurements (RTT)
• All measurements present RTT values smaller than 200 ms for more than 97% of calls– RTT cannot be considered as a
major impairment of VoIP call quality
15
Network-Centric Measurements (Jitter)
• Jitter is traced smaller than 15ms– Inter-packet-gap is 20ms
16
Network-Centric Measurements (Packet Loss)
• Average loss probability: 2.8%
• The bottom picture showed that packet loss probability has little correlation with the actual network load