Analytical model and comparative evaluation of application layer loss in the context of media encapsulation in wireless RTC
- 75 Downloads
Internet firewalls are typically configured to allow web friendly traffic transported by means of the Transmission Control Protocol (TCP) and, simultaneously as security measure, reject any other type of transport. This is a problem for real time communications that heavily rely on transport types like User Datagram Protocol or those associated to the Internet Security Protocol. A mechanism to overcome this limitation is through stream tunneling where media frames that would be typically rejected by firewalls are encapsulated on top of TCP transport. In this paper, a novel mathematical model that links application layer packet loss to bursty network packet loss in the context of fading channels characteristic of wireless communications is presented for both, datagram and stream encapsulation. This model is compared with experimental scenarios of tunneling applied to state of the art speech codecs such that quality scores are obtained and correlated against theoretical results.
KeywordsRTC Tunneling Encapsulation Firewall Traversal Security
- 1.3GPP TS 23.107. (2009). Technical specification group services and system aspects; quality of service (QoS) concept and architecture (release 9), v9.0.0.Google Scholar
- 2.3GPP: Ts 26.071. (2008). Mandatory speech codec speech processing functions; amr speech codec; general description. Technical report TS 26.071, 3rd Generation Partnership Project.Google Scholar
- 3.3GPP: Ts 26.190. (2008). Speech codec speech processing functions; adaptive multi-rate-wideband (AMR-WB) speech codec; transcoding functions. Technical report TS 26.190, 3rd Generation Partnership Project.Google Scholar
- 4.3GPP. (2014). Feasibility study on IMS firewall traversal. TR 33.830, 3rd Generation Partnership Project (3GPP).Google Scholar
- 5.3GPP. (2015). Tunnelling of IP multimedia subsystem (IMS) services over restrictive access networks; Stage 3. TS 24.322, 3rd Generation Partnership Project (3GPP).Google Scholar
- 6.3GPP2: C.s0014-a. (2004). Enhanced variable rate codec, speech service option 3 for wideband spread spectrum digital systems. Technical report, C.S0014-A, 3rd Generation Partnership Project 2.Google Scholar
- 7.Barton, M., Lemberg, H., Sarraf, M., & Hamilton, C. (2010). Performance analysis of packet loss concealment in mobile environments with a two-state loss model. In 2010 IEEE international workshop technical committee on communications quality and reliability (CQR) (pp. 1–6) doi: 10.1109/CQR.2010.5619908.
- 9.Bu, T., Liu, Y., & Towsley, D. (2006) On the TCP-friendliness of VoIP traffic. In Proceedings IEEE INFOCOM 2006. 25TH IEEE international conference on computer communications (pp. 1–12). doi: 10.1109/INFOCOM.2006.245.
- 10.Chong, H. M., & Matthews, H. S. (2004). Comparative analysis of traditional telephone and voice-over-internet protocol (VoIP) systems. In 2004 IEEE international symposium on electronics and the environment, 2004. Conference record (pp. 106–111). doi: 10.1109/ISEE.2004.1299697.
- 11.Cocker, E., Ghazzi, F., Speidel, U., Dong, M. C., Wong, V., Vinck, A. J. H., Yamamoto, H., Yokoo, H., Morita, H., Ferreira, H., Emleh, A., McFadzien, R., Palelei, S., & Eimann, R. (2014). Measurement of buffer requirement trends for real time traffic over TCP. In 2014 IEEE 15th international conference on high performance switching and routing (HPSR) (pp. 120–124). doi: 10.1109/HPSR.2014.6900891.
- 12.Ecotronics. Kapanga softphone. http://www.kapanga.net.
- 13.Ellis, M., Pezaros, D. P., Kypraios, T., & Perkins, C. (2012). Modelling packet loss in RTP-based streaming video for residential users. In 2012 IEEE 37th conference on local computer networks (LCN) (pp. 220–223). doi: 10.1109/LCN.2012.6423613.
- 14.Epiphaniou, G., Maple, C., Sant, P., & Reeve, M. (2010). Affects of queuing mechanisms on RTP traffic: Comparative analysis of jitter, end-to-end delay and packet loss. In ARES ’10 international conference on availability, reliability, and security, 2010 (pp. 33–40). doi: 10.1109/ARES.2010.67.
- 15.Gonia, K. (2004). Latency and QoS for Voice over IP. SANS Institute, Technical report.Google Scholar
- 16.Hemminger, S. (2005). Network emulation with NetEm. http://developer.osdl.org/shemminger/netem/LCA2005_paper.pdf.
- 17.Herrero, R. (2016). Integrating HEC with circuit breakers and multipath RTP to improve RTC media quality. Telecommunication Systems. doi: 10.1007/s11235-016-0169-z.
- 18.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.Google Scholar
- 19.Hwang, H., Yin, X., Wang, Z., & Wang, H. (2009). The internet measurement of VoIP on different transport layer rotocols. In 2009 International conference on information networking (pp. 1–3).Google Scholar
- 20.ITU-T: G.711. (2006). Pulse code modulation (PCM) of voice frequencies. Technical report, G.711, International Telecommunication Union, Geneva.Google Scholar
- 21.ITU-T Recommendation P.501. (2009). Test Signals for use in Telephonometry. Technical report, International Telecommunications Union, Geneva, Switzerland.Google Scholar
- 22.ITU-T Recommendation P.863. (2014). Perceptual objective listening quality assessment (POLQA). Technical Report, International Telecommunication Union, Geneva, Switzerland.Google Scholar
- 23.Kent, S., & Seo, K. (2005). Security architecture for the internet protocol. RFC 4301 (Proposed standard). http://www.ietf.org/rfc/rfc4301.txt.
- 24.Mahy, R., Matthews, P., & Rosenberg, J. (2010). Traversal using relays around NAT (TURN). RFC 5766 (Internet standard).Google Scholar
- 25.McNeill, K., Liu, M., & Rodriguez, J. (2006). An adaptive jitter buffer play-out scheme to improve VoIP quality in wireless networks. In Military communications conference, 2006. MILCOM 2006. IEEE (pp. 1–5). doi: 10.1109/MILCOM.2006.302119.
- 27.Pouffary, Y., & Young, A. (1997). ISO Transport Service on top of TCP (ITOT). RFC 2126 (Proposed standard). http://www.ietf.org/rfc/rfc2126.txt.
- 28.Psaras, I., & Tsaoussidis, V. (2007). The TCP minimum RTO revisited. In I. F. Akyildiz, R. Sivakumar, E. Ekici, J. C. de Oliveira, & J. McNair (Eds.), Networking. Lecture notes in computer science (Vol. 4479, pp. 981–991). Berlin: Springer.Google Scholar
- 29.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.Google Scholar
- 30.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) (Vol. 2, p. 775). Washington, DC: IEEE Computer Society.Google Scholar
- 31.Sanchez-Iborra, R., Cano, M. D., & Garcia-Haro, J. (2013). Performance evaluation of QoE in VoIP traffic under fading channels. In 2013 World congress on computer and information technology (WCCIT) (pp. 1–6). doi: 10.1109/WCCIT.2013.6618721.
- 32.Satoda, K., Nihei, K., & Yoshida, H. (2014). Quality evaluation of voice over multiple TCP connections. In 2014 International conference on computing, networking and communications (ICNC) (pp. 141–146). doi: 10.1109/ICCNC.2014.6785320.
- 33.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.Google Scholar
- 34.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). New York, NY: ACM. doi: 10.1145/2483977.2484002.
- 35.Valin, J., Vos, K., & Terriberry, T. (2012). Definition of the Opus audio codec. RFC 6716 (Proposed standard).Google Scholar