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Understanding VoIP from Backbone Measurements

Marco Mellia, Dario Rossi Robert Birke, and Michele PetraccaINFOCOM 07’, Anchorage, Alaska, USA

Young J. Won

Oct. 15, 2007

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Outline• Introduction• Measurement methodology• The FastWeb network• Measurement results• Conclusion

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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

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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.

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Why?• Why we are interested in this?

– Curiosity– Extracting basic parameter values for

OPNET and later modeling– Basis for the theorical scenario analysis

models

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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

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The FastWeb Network

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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

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Measurement Results• Two probe nodes located in a PoP locat

ed 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

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HAG Distribution

• Home Access Gateway (HAG)– Offers Ethernet ports to PCs, VideoBox, Plain Old Telephone

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Number of Phone Calls Tracked

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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

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eMOS CDF• eMOS

– Excellent ( eMOS>4 )– Good ( eMOS[3:4] )

• eMOS >= 3.6– The same quality as trad

itional PSTN phone calls

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Network-Centric Measurements (RTT)• All measurements present R

TT values smaller than 200 ms for more than 97% of calls– RTT cannot be considered as a

major impairment of VoIP call quality

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Network-Centric Measurements (Jitter)

• Jitter is traced smaller than 15ms– Inter-packet-gap is 20ms

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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

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Conclusion• This paper presented an extensive measure

ment campaign focusing on VoIP traffic characterization

• Use eMOS model to compare the quality of VoIP to traditional PSTN phone calls

• In FastWeb, only the packet loss probability affected the quality of VoIP