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Integrating HEC with circuit breakers and multipath RTP to improve RTC media quality

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Abstract

The use of hybrid error correction (HEC) schemes that integrate both, forward error correction as well as negative acknowledgments, has proven to be beneficial in substantially improving reliability while keeping transmission rates under control. In this paper we introduce a framework that combines HEC with circuit breakers and multipath media as a way to increase quality without affecting bandwidth performance. Circuit breakers consist of selectively transmitting media frames based on network measurements in order to minimize congestion. On the other hand, multipath media takes advantage of the fact that most modern mobile devices incorporate multiple network interfaces that can be used to transmit traffic simultaneously through many possible routes. We specifically focus on the aforementioned integration scheme when applied in the context of real time communications and quantify it as a function of network impairments by means of standard quality scores for a wide range of speech and video codecs.

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References

  1. Schulzrinne, H., Casner, S., Frederick, R., Jacobson, V. (2003). RTP: A transport protocol for real-time applications. RFC 3550 (Internet Standard). Updated by RFCs 5506, 5761, 6051, 6222.

  2. Baugher, M., McGrew, D., Naslund, M., Carrara, E., Norrman, K.: The secure real-time transport protocol (SRTP). RFC 3711 (Internet Standard).

  3. 3GPP TS 23.107. (2009). Technical specification group services and system aspects; quality of service (QoS) concept and architecture (release 9), v9.0.0.

  4. Lazzaro, J.: Framing real-time transport protocol (RTP) and RTP control protocol (RTCP) packets over connection-oriented transport. RFC 4571 (Internet Standard).

  5. McNeill, K., Liu, M., Rodriguez, J. (2006). An adaptive jitter buffer play-out scheme to improve voip quality in wireless networks. In IEEE Military Communications Conference, 2006. MILCOM 2006 (pp. 1–5). doi:10.1109/MILCOM.2006.302119.

  6. Holmer, S., Shemer, M., Paniconi, M. (2013). Handling packet loss in webrtc. In ICIP (pp. 1860–1864).

  7. Zhai, F., Eisenberg, Y., Pappas, T., Berry, R., & Katsaggelos, A. (2006). Rate-distortion optimized hybrid error control for real-time packetized video transmission. IEEE Transactions on Image Processing, 15(1), 40–53. doi:10.1109/TIP.2005.860353.

    Article  Google Scholar 

  8. Herrero, R., Cadirola, M. (2014). Effect of fec mechanisms in the performance of low bit rate codecs in lossy mobile environments. In Principles, Systems and Applications of IP Telecommunications, IPTComm ’14.

  9. Ding, J. W., Deng, D. J., Lo, Y. K., & Park, J. H. (2013). Perceptual quality based error control for scalable on-demand streaming in next-generation wireless networks. Telecommunication Systems, 52(2), 445–459. doi:10.1007/s11235-011-9447-y.

    Google Scholar 

  10. Li, A. (2007). RTP payload format for generic forward error correction. RFC 5109 (Proposed Standard).

  11. Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., Peterson, J., Sparks, R., Handley, M., Schooler, E. (2002). SIP: Session initiation protocol. RFC 3261 (Proposed Standard). Updated by RFCs 3265, 3853, 4320, 4916, 5393, 5621, 5626, 5630, 5922, 5954, 6026, 6141, 6665.

  12. Matsuzono Matsuzono, K., Detchart Detchart, J., Cunche Cunche, M., Roca Roca, V., Asaeda Asaeda, H. (2010). Performance analysis of a high-performance real-time application with several al-fec schemes. In Proceedings of the 2010 IEEE 35th Conference on Local Computer Networks, LCN ’10 (pp. 1–7).

  13. Ott, J., Wenger, S., Sato, N., Burmeister, C., Rey, J. Extended RTP profile for real-time transport control protocol (RTCP)-based feedback (RTP/AVPF). RFC 4585 (Proposed Standard).

  14. Ott, J., Carrara, E. Extended Secure RTP profile for real-time transport control protocol (RTCP)-based feedback (RTP/SAVPF). RFC 5124 (Proposed Standard).

  15. Gruen, J., Gorius, M., Herfet, T. (2013). Interactive rtp services with predictable reliability. In IEEE Third International Conference on Consumer Electronics, Berlin (ICCE-Berlin), 2013 (pp. 371–375). doi:10.1109/ICCE-Berlin.2013.6698014

  16. Sarker, Z., Singh, V., Perkins, C. (2014). An evaluation of rtp circuit breaker performance on lTE networks. In IEEE Conference on Computer Communications Workshops (INFOCOM WKSHPS) (pp. 251–256). doi:10.1109/INFCOMW.2014.6849240

  17. Singh, V., Ahsan, S., Ott, J. (2013). Mprtp: Multipath considerations for real-time media. In Proceedings of the 4th ACM Multimedia Systems Conference, MMSys ’13 (pp. 190–201). ACM, New York, NY, USA. doi:10.1145/2483977.2484002

  18. Karkkainen, T., Ott, J. (2014). S.A.L.E.: Multipath RTP (MPRTP). IETF Internet Draft—work in progress 10

  19. Globisch, R., Sanchez, Y., Schierl, T., Ferguson, K., Wiegand, T.: Retransmission timeout estimation for low-delay applications using multipath rtp. In 28th International Conference on Advanced Information Networking and Applications Workshops (WAINA), 2014 (pp. 759–764). doi:10.1109/WAINA.2014.124.

  20. Shacham, N., McKenney, P.E. (1990) Packet recovery in high-speed networks using coding and buffer management. In INFOCOM’90, Ninth Annual Joint Conference of the IEEE Computer and Communication Societies

  21. Singer, D., Desineri, H. A general mechanism for RTP header extensions. RFC 5285 (Internet Standard)

  22. Jacobson, V. (1988). Congestion avoidance and control. In Symposium Proceedings on Communications Architectures and Protocols, SIGCOMM ’88 (pp. 314–329). ACM, New York, NY, USA. doi:10.1145/52324.52356.

  23. Abbas, S., Mosbah, M., Zemmari, A., Bordeaux, U. (1996). Itu-t recommendation g.114, one way transmission time. In International Conference on Dynamics in Logistics. LDIC 2007. Lect. Notes in Comp. Sciences: Springer.

  24. Jammeh, E., Mkwawa, I., Khan, A., Goudarzi, M., Sun, L., & Ifeachor, E. (2010). Quality of experience (QOE) driven adaptation scheme for voice/video over IP. Telecommunication Systems, 49(1), 99–111. doi:10.1007/s11235-010-9356-5.

    Article  Google Scholar 

  25. Li, M., & Lee, C. Y. (2014). A cost-effective and real-time QOE evaluation method for multimedia streaming services. Telecommunication Systems, 59(3), 317–327. doi:10.1007/s11235-014-9938-8.

    Article  Google Scholar 

  26. Fujimoto, K., Ata, S., & Murata, M. (2004). Adaptive playout buffer algorithm for enhancing perceived quality of streaming applications. Telecommunication Systems, 25(3), 259–271.

    Article  Google Scholar 

  27. Ecotronics: Kapanga softphone. http://www.kapanga.net

  28. Hemminger, S. (2005). Network emulation with NetEm. http://developer.osdl.org/shemminger/netem/LCA2005%5C_paper.pdf

  29. Jakes, W. C., & Cox, D. C. (1994). Microwave mobile communications. New York: Wiley-IEEE Press.

    Book  Google Scholar 

  30. ITU-T: G.711 (2006) Pulse code modulation (pcm) of voice frequencies. Tech. Rep. G.711, International Telecommunication Union, Geneva

  31. ITU-T: G.722. (2006). 7 khz audio-coding within 64 kbit/s. Tech. Rep. G.722, International Telecommunication Union, Geneva

  32. Salami, R., Laflamme, C., Bessette, B., Adoul, J. (1997). Description of itu-t recommendation g.729 annex a: Reduced complexity 8 kbit/s cs-acelp codec. In: Proceedings of the 1997 IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP ’97)-Volume 2 - Volume 2, ICASSP ’97 (p. 775). IEEE Computer Society, Washington, DC, USA.

  33. 3GPP: Ts 26.071. (2008). Mandatory speech codec speech processing functions; AMR speech codec; general description. Tech. Rep. TS 26.071, 3rd Generation Partnership Project

  34. 3GPP2: C.s0014-a. (2004). Enhanced variable rate codec, speech service option 3 for wideband spread spectrum digital systems. Tech. Rep. C.S0014-A, 3rd Generation Partnership Project 2

  35. 3GPP. (2008). Ts 26.190 : Speech codec speech processing functions; adaptive multi-rate—wideband (amr-wb) speech codec; transcoding functions. Tech. Rep. TS 26.190, 3rd Generation Partnership Project

  36. Valin, J., Vos, K., Terriberry, T. (2012). Definition of the Opus Audio Codec. RFC 6716 (Proposed Standard)

  37. S. Andersen A. Duric. (2004). H.A.R.H.W.K.J.L.: Internet low bit rate codec (ILBC). RFC 3951 (Proposed Standard).

  38. ITU-T Recommendation P.862. (2001). Perceptual evaluation of speech quality (PESQ): An objective method for end-to-end speech quality assessment of narrow-band telephone networks and speech codecs . Tech. rep., International Telecommunication Union, Geneva, Switzerland.

  39. ITU-T Recommendation P.862.2. (2007). Wideband extension to Recommendation P.862 for the assessment of wideband telephone networks and speech codecs. Tech. rep., International Telecommunication Union, Geneva, Switzerland.

  40. ITU-T: H.263. (2005). Video coding for low bit rate communication. Tech. Rep. H.263, International Telecommunication Union, Geneva

  41. ITU-T: H.264. (2014). Advanced video coding for generic audiovisual services. Tech. Rep. H.264, International Telecommunication Union, Geneva

  42. Bankoski, J., Koleszar, J. (2011). L.Q.J.S.P.W.Y.X.: VP8 data format and decoding guide. RFC 6386.

  43. ITU-T: J.247. (2008). Objective perceptual multimedia video quality measurement in the presence of a full reference. Tech. Rep. J.247, International Telecommunication Union, Geneva

  44. Chong, H.M., Matthews, H.S. (2004). Comparative analysis of traditional telephone and voice-over-internet protocol (voip) systems. In: IEEE International Symposium on Electronics and the Environment, 2004. Conference Record (pp. 106–111). doi:10.1109/ISEE.2004.1299697

  45. Ribadeneira, A.F. (2007). An analysis of the MOS under conditions of delay, jitter and packet loss and an analysis of the impact of introducing piggybacking and reed solomon FEC for VOIP. Master’s thesis, Georgia State University, USA.

  46. Gonia, K. (2004). Latency and QoS for voice over IP. SANS Institute: Tech. rep.

  47. Walsh, T. J., & Kuhn, D. R. (2005). Challenges in securing voice over IP. IEEE Security and Privacy, 3(3), 44–49.

    Article  Google Scholar 

  48. ITU-T Recommendation P.863: Tech. rep., International Telecommunication Union, Geneva, Switzerland.

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  1. Northeastern University, 360 Huntington Avenue, Boston, MA, 02115-5000, USA

    Rolando Herrero

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  1. Rolando Herrero
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Herrero, R. Integrating HEC with circuit breakers and multipath RTP to improve RTC media quality. Telecommun Syst 64, 211–221 (2017). https://doi.org/10.1007/s11235-016-0169-z

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