1. An T. Le - USF 2006 - VoIP Packet...1 VOIP Packet loss, packet labeling and packet classification An T. Le

2. An T. Le - USF 2006 - VoIP Packet...2 VoIP Properties Real-time stream requested Typical Internet applications use TCP/IP, whereas VoIP uses RTP/UDP/IP. In VoIP, voice is sending over IP network in IP packets. Latency Packet Loss …There will be no “re-send please” in VoIP

3. An T. Le - USF 2006 - VoIP Packet...3 OSI 7 layers model

4. An T. Le - USF 2006 - VoIP Packet...4 IP packets

5. An T. Le - USF 2006 - VoIP Packet...5 VoIP packet size VoIP packet size is very important. The packet size relates to the delay times that needs for sending and receiving packet The lost of packets is impacting to the quality of reconstructed voice stream. S = VoIP size = Playload + RTP header + UDP/IP header S = s+12bytes+28bytes S = s+40bytes = s + 320bits Headers size is large, payload size (s) is expected small enough to reduce delay time

6. An T. Le - USF 2006 - VoIP Packet...6 VoIP bandwidth request R=CODEC rate (bps) n= number of packet in second n=R/s where s is a size of payload in second If s=R then n=1 (badly delay) H= Header size BW= n(R/n+H) = R(1+H/s) Examples: If R=64kbps, s=1.28kbps (n=50 or length of payload is 20ms) then R=(64k+40*8*50)=80kbps If R=64kbps, s=0.64kbps (n=100 or length of payload is 10ms) then R=(64k+40*8*100)=96kbps

7. An T. Le - USF 2006 - VoIP Packet...7 VoIP – End to End Stream

8. An T. Le - USF 2006 - VoIP Packet...8 VoIP – End to End Stream with delay

9. An T. Le - USF 2006 - VoIP Packet...9 VoIP’s QoS Latency (end to end) CODE and DECODE processing delays Completed (Header + payload) transfer delays Application delays Propagation delays … Packet loss Lost in transmission Lost in congestion (jitter) …Lost a bit in header - lost a packet

10. An T. Le - USF 2006 - VoIP Packet...10 Packet loss Ratio and QoS Packet loss vs MOS (Mean of Opinion Score)  (source: http://www.kineto.com )http://www.kineto.com

11. An T. Le - USF 2006 - VoIP Packet...11 Paket loss Ratio and QoS: Reconstruction and examples Reconstruction: If packet is large, interleaving is may not used due to real time is requested Samples: http://www.voiptroubleshooter.comhttp://www.voiptroubleshooter.com Silence Insertion Replay last packet G.711 Appendix 1 5% loss rate 10% loss rate 20% loss rate 40% loss rate

12. An T. Le - USF 2006 - VoIP Packet...12 Packet loss analysis Lost by delay: As a real-time system, a long delay packet (>500ms) is considered as loss. Lost by bad receiving: Uncompleted header packet, consider as loss Uncompleted payload packet, consider as loss if using any LPC codec

13. An T. Le - USF 2006 - VoIP Packet...13 Loss caused by delay Congestion will cause delay then loss Quality of transmission link: Capacity=BW.log 2 (1+SNR) SNR is not independently with BW Improve SNR by repeater with amplifier

14. An T. Le - USF 2006 - VoIP Packet...14 Optimal Packetization in VoIP Optimal packet’s size to reduce end to end delay Optimal packet’s size to minimum loss ratio (usually that makes packet smaller) Three main (but not independently) parameters Bandwidth budget Min = 8000+320/0.1 = 11.2kbps (with G.729) Avg = 64000+320/0.02 = 80kbps (with G.711) Delay threshold (max = 240ms ?) MOS threshold (min=3 ?)

15. An T. Le - USF 2006 - VoIP Packet...15 VoIP – Bandwidth request

16. An T. Le - USF 2006 - VoIP Packet...16 R-factor vs MOS An IP phone monitoring application using the SNMP The R-factor is described in the ITU-T G.107 recommendation which defines a computing model known as an E-model. The R-factor is a well-tried tool for transmission planning and for determining the combined impact of various transmission parameters which influence the call quality. All appropriate transmission parameters are put together to calculate the R-factor as follows: E-EFF R = RO - IS - ID - I E-EFF + A where ROis the basic signal-to-noise ratio, ISis a sum of all impairments occurring during speech transmission, IDis a degradation factor representing all impairments caused by the voice signal delay, I E-EFF includes packet loss, A is an advantage factor (permitted range is from 0 to 20)

17. An T. Le - USF 2006 - VoIP Packet...17 Acceptable MOS and R Scores for Narrowband CODECs (source: voicetroubeshooter.com) User OpinionR FactorMOS Score Maximum obtainable for G.711934.4 Very satisfied90-1004.3-5.0 Satisfied80-904.0-4.3 Some users satisfied70-803.6-4.0 Many users dissatisfied60-703.1-3.6 Nearly all users dissatisfied50-602.6-3.1 Not recommended0 - 501.0-2.6

18. An T. Le - USF 2006 - VoIP Packet...18 Packet labeling Packet classification Concern about Class of Service (CoS) Fast detect Packet class Give a label to packet Use Packet classification Packet labeling can be done by HW or SW Packet classification, usually, done by HW (Programmable Logic Controller)

19. An T. Le - USF 2006 - VoIP Packet...19 Packet labeling Packet classification (cont) Bit 8-15 in IP header: TOS: Type of Service 0001020304050607 PrecedenceDTRM0 Precedence. 3 bits. ValueDescription 0Routine. 1Priority. 2Immediate. 3Flash. 4Flash override. 5CRITIC/ECP. 6Internetwork control. 7Network control. D. 1 bit. Minimize delay. ValueDescription 0Normal delay. 1Low delay. T. 1 bit. Maximize throughput. ValueDescription 0Normal throughput. 1High throughput. R. 1 bit. Maximize reliability. ValueDescription 0Normal reliability. 1High reliability. M. 1 bit. Minimize monetary cost. ValueDescription 0Normal monetary cost. 1Minimize monetary cost.

20. An T. Le - USF 2006 - VoIP Packet...20 Packet labeling Packet classification (cont) RTP header, the use of PT (Payload Type) bits (9-15) PTNameType Clock rate (Hz) Audio channels References 0PCMUAudio80001RFC 3551 3GSMAudio80001RFC 3551 7LPCAudio80001RFC 3551 8PCMAAudio80001RFC 3551 12QCELPAudio80001 18G729Audio80001 19reservedAudio 20- 24 27 29- 30 35-71 77-95

21. An T. Le - USF 2006 - VoIP Packet...21 Packet labeling Packet classification (cont) Use of UDP header Source PortDestination Port LengthChecksum Data:::

22. An T. Le - USF 2006 - VoIP Packet...22 Packet labeling Packet classification (cont) Use of Payload ?

23. An T. Le - USF 2006 - VoIP Packet...23 Conclusion Packet loss by processing by end to end delay by congestion by bad transceiver and link Solutions for packet loss Use “good" CODEC Optimal packet (based on Bandwidth, CODEC, desire of delay and MOS Need of Real time QoS monitor and adaptive variable packet size protocol. Packet labeling and classification Fast packet class detection

24. An T. Le - USF 2006 - VoIP Packet...24 Reference 1- Voice over Internet protocol (VoIP) Goode, B.; Proceedings of the IEEE Volume 90, Issue 9, Sept. 2002 Page(s):1495 - 1517 Performance comparison between VBR speech coders for adaptive VoIP applications Beritelli, F.; Casale, S.; Ruggeri, G.; Communications Letters, IEEE Volume 5, Issue 10, Oct. 2001 Page(s):423 - 425 Congestion Avoidance Using DYnamic COdec MAnagement: A solution for ISP Alcuri, L.; Saitta, F.; Fasciana, M.L.; Communications, 2005 Asia-Pacific Conference on 03-05 Oct. 2005 Page(s):886 - 890 On packet loss concealment artifacts and their implications for packet labeling in voice over IP Praestholm, S.; Jensen, S.S.; Andersen, S.V.; Murthi, M.N.; Multimedia and Expo, 2004. ICME '04. 2004 IEEE International Conference on Volume 3, 27-30 June 2004 Page(s):1667 - 1670 Vol.3 Assessment of effects of packet loss on speech quality in VoIP Ding, L.; Goubran, R.A.; Haptic, Audio and Visual Environments and Their Applications, 2003. HAVE 2003. Proceedings. The 2nd IEEE Internatioal Workshop on 20-21 Sept. 2003 Page(s):49 - 54 Voice-quality monitoring and control for VoIP Manousos, M.; Apostolacos, S.; Grammatikakis, I.; Mexis, D.; Kagklis, D.; Sykas, E.; Internet Computing, IEEE, Volume 9, Issue 4, July-Aug. 2005 Page(s):35 - 42 Digital Object Identifier 10.1109/MIC.2005.92 ……. Others listed in this presentation

25. An T. Le - USF 2006 - VoIP Packet...25 THANK YOU