Communications Network Research Institute

EQUAL Project

Mirosław (Mirek) Narbutt Piotr Soboński

Project goal

Real-time implementation of the ITU-T E-Model quality contours in a playout buffer module of a VoIP application.


If you would like to read a short summary of the EQUAL project please click the following link: Download EQUAL summary

Project duration

October 2007 − December 2008

Funding Agency

Enterprise Ireland (under the Commercialisation Fund Proof of Concept 2007 programme). The project has been co-funded by the European Regional Development Fund (ERDF) under the Southern & Eastern Regional Operational Programme 2007-13.

Investing in your future.


An extension to the ITU-T E-model developed for the pruposes of predicting VoIP transmission quality has recently been adopted by the ITU-T as Appendix I to the ITU-T G.109 Recommendation [1], [2], [3], [4]. To date, this method has been realized as an off-line application layer tool for evaluating various playout buffer algorithms [6], [7] and has been shown to be particularly effective in assessing the performance of Voice over WLAN systems [8], [9], [10], [11], [12], [13], [14]. This project proposes to develop a real-time implementation of this method that is capable of being integrated into adaptive tuning schemes for optimising VoIP quality.


A fundamental trade-off exists between average buffering delay and late packet loss. A large buffering time causes an increase in the overall delay but decreases the packet loss, while a small buffering time decreases the delay but increases the packet loss. A good playout algorithm should be able to keep the buffering as short as possible while minimizing the number of packets that arrive too late to be played out. These two conflicting goals have led to various adaptive playout algorithms, which can be grouped into four categories:

Although this statistical method of evaluating playout algorithms can be useful, it does not provide a direct link to perceived conversational speech quality. Instead, we predict user satisfaction from time varying loss/delay impairments using the ITU-T standardized quality contours.


The main objective of this project is to realise an adaptive de-jitter playout buffer for VoIP applications that is controlled by the real-time implementation of the E-model and which seeks to achieve the highest user satisfaction under all network conditions. The intention is to control the playout buffer module using the voice transmission quality predictor in order to minimize the effect of transmission impairments and thus maximize user satisfaction.


Implementation has been divided on two parts:


Experimental test bed

Both modified RTP tools applications (i.e. RTPsend and RTPdump) were tested through network emulations, and in a real wireless LAN environment. Realistic voice sources were used with active and silent periods (in accordance with ITU-T recommendation P.59 [5]). Voice packets were generated at the sender every 10ms during active periods. No packets were generated during silent periods. The total duration of each test was 1 hour. The simulation scenarios (see Figure 1) covered a wide range of network conditions regarding delay and jitter behaviour.

For network emulations the NISTnet 2.1.0 network emulation software has been chosen [24]. Both RTPsend and RTPdump applications were running in different networks. Voice traffic flow was sent from RTPsend to RTPdump application through a NISTnet router that modelled various delay patterns. Figure 2 shows delay patterns observed during network emulations.

Fig. 1

Figure 1: Test simulation setup for the EQUAL module.

Fig. 2A Fig. 2C Fig. 2B

Figure 2: Delay patterns used in experiments: (A) delay = 0ms, jitter varies; (B) both delay and jitter vary; (C) delay = 100ms, jitter varies.

Experimental results

Both the E-Model method and the playout buffer module have been successfully implemented in the RTP tools software. We tested its performance through network simulations and via experiments in a real wireless environment.

We compared the performance of reference playout algorithms (with fixed playout deadlines, Ramjee’s [14],and Moon’s [18]) with our optimized algorithm. For comparison purposes we have chosen network traces: A, B, C. Table 1 compares the maximal values of rating factor R achieved by the reference algorithms with the values of R obtained using our optimized algorithm. As an example graphical results from fixed playout are shown below for trace A, B and C.

Transmission factor R
Trace A
Trace B
Trace C
Fixed playout delay (160 ms)
Ramjee’s alg.
Moon’s alg.
Optimized R
(our solution)

Table.1. Algorithm performance comparison

Results have shown that our quality driven buffer adaptation provides the highest R score among examined playout adaptation schemes.

Key Benefits

Prototype Details/Requirements

Results for trace A

Trace A − fixed playout (pD=160ms)

Trace A − Ramjee’s alg. (α=0.998002, β=4)

Trace A − Moon’s alg. (p=0.5%, W=1200)

Trace A − optimized playout

Results for trace B

Trace B − fixed playout (pD=160ms)

Trace B − Ramjee’s alg. (α=0.998002, β=4)

Trace B − Moon’s alg. (p=0.5%, W=1200)

Trace B − optimized playout

Results for trace C

Trace C − fixed playout (pD=160ms)

Trace C − Ramjee’s alg. (α=0.998002, β=4)

Trace C − Moon’s alg. (p=0.5%, W=1200)

Trace C − optimized playout


  1. Appendix I (01/2007), "The E-model based quality contours for predicting speech transmission quality and user satisfaction from time-varying transmission impairments"
  2. ITU-T Recommendation G.107 (03/2003), "The E-Model, a computational model for use in transmission planning"
  3. ITU-T Recommendation G.113 Appendix I (05/2002). "Provisional planning values for the equipment impairment factor Ie and packet-loss robustness factor Bpl"
  4. ITU-T Recommendation G.109 (09/1999), "Definition of categories of speech transmission quality"
  5. ITU-T Recommendation P.59 (03/1993), "Artificial conversational speech"
  6. M.Narbutt, M. Davis "Assessing the Quality of VoIP Transmission Affected by Playout Buffer Scheme", Proc. of the ETSI/IEEE Measurement of Speech and Audio Quality in Networks Conference 2005 (MESAQIN 2005),Prague, June 2005.
  7. M.Narbutt, A. Kelly, L. Murphy, P. Perry, "Adaptive VoIP Playout scheduling: Assessing User Satisfaction", IEEE Internet Computing Magazine, vol. 09, No.4, July/August 2005
  8. M.Narbutt, M. Davis "Gauging VoIP Call Quality from 802.11b Resource Usage", Proc of the IEEE International Symposium on a World of Wireless, Mobile and Multimedia Networks (WoWMoM06), Buffalo-NY, June 2006
  9. M.Narbutt, M. Davis, "Experimental investigation on VoIP performance and the resource utilization in 802.11b WLANs", Proc of the 31st IEEE Conference on Local Computer Networks (LCN’06), Tampa, November 2006
  10. M.Narbutt, M. Davis, "Effect of Free Bandwith on VoIP Performance in 802.11b WLANs", IEE Irish Signals and Systems Conference 2006 (ISSC 2006), Dublin, June 2006
  11. M.Narbutt, M. Davis, "Experimental tuning of AIFSN and CWmin parameters to prioritize voice over data transmission in 802.11e WLAN networks", ACM Cross-layer optimized wireless networks symposium, International Conference on Wireless Communications and Mobile Computing (IWCMC 2007), Hawaii, August 2007
  12. M.Narbutt, M. Davis, "The capability of the EDCA mechanism to support voice traffic in a mixed voice/data transmission over 802.11e WLANs - an experimental investigation", IEEE Conference on Local Computer Networks (LCN’07), Dublin, Ireland, October 2007
  13. M.Narbutt, M. Davis, "Experimental Tuning of the AIFSN Parameter to Prioritize Voice over Data Transmission in 802.11e WLAN Networks", IEEE International Conference on Signal Processing and Communications (ICSPC 2007), Dubai, November 2007
  14. R. Ramjee, J. Kurose, D. Towsley, and H. Schulzrinne, "Adaptive playout mechanisms for packetized audio applications in wide-area networks", Proceedings of the IEEE Infocom, June 1994
  15. J.C. Bolot and A. Vega-Garcia, "Control mechanisms for packet audio in the Internet", Proceedings of the IEEE Infocom 1996, April’96;
  16. M.Narbutt, L. Murphy, "VoIP Playout Buffer Adjustment using Adaptive Estimation of Network Delays", Proceedings of the 18-th International Teletraffic Congress − ITC-18, p.1171-1180, September 2003
  17. N. Shivakumar, C. J. Sreenan, B. Narendran, and P. Agrawal, "The Concord algorithm for synchronization of networked multimedia streams", Proceedings of the IEEE International Conference on Multimedia Computing and Systems, May 1995
  18. S. B. Moon, J. Kurose, and D. Towsley, "Packet audio playout delay adjustment: Performance bounds and algorithms", ACM/Springer Multimedia Systems, Vol. 6, January 1998
  19. D. L. Stone, K. Jeffay, "An empirical study of delay jitter management policies", ACM/Springer Multimedia Systems Journal, Vol. 2, No. 6, January, ’95
  20. A. P. Markopoulou, PhD Dissertation: "Assessing the Quality of Multimedia Communications over Internet Backbone Networks", Stanford University, 2003
  21. K. Fujimoto, S. Ata, M. Murata, "Adaptive playout buffer algorithm for enhancing perceived quality of streaming applications", Proceedings of the IEEE GLOBECOM 2002 Conference, vol. 21, no. 1, pp. 2463 − 2469, Nov.’02
  22. L. Sun, E. Ifeachor, "Prediction of Perceived Conversational Speech Quality and Effects of Playout Buffer Algorithms", Proceedings of the IEEE ICC’03, Anchorage, USA, May 2003, pp. 1-6
  23. RTPtools software
  24. NISTnet, Network Emulator software
  25. MGEN, The Multi-Generator Toolset