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Understanding VoIP from Backbone Measurements

Understanding VoIP from Backbone Measurements. Marco Mellia, Dario Rossi Robert Birke, and Michele Petracca INFOCOM 07’, Anchorage, Alaska, USA Young J. Won Oct. 15, 2007. Outline. Introduction Measurement methodology The FastWeb network Measurement results Conclusion. Introduction.

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Understanding VoIP from Backbone Measurements

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  1. Understanding VoIP from Backbone Measurements Marco Mellia, Dario Rossi Robert Birke, and Michele Petracca INFOCOM 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 measurement 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

  7. The FastWeb Network

  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

  11. Number of Phone Calls Tracked

  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 traditional PSTN phone calls

  14. Network-Centric Measurements (RTT) • All measurements present RTT values smaller than 200ms 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

  17. Conclusion • This paper presented an extensive measurement 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

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