以 RTP 代理伺服器為輔助的網路電話品質量測

Voice Quality Monitoring Assisted by RTP Proxy Servers

李霓雅(Ni-Ya Li)

指導教授 ： 吳坤熹

國立暨南國際大學/科技學院/資訊工程學系/碩士(2010年)

https://doi.org/10.6837/NCNU.2010.00100

摘要

自從網路電話(Voice over Internet Protocol, 簡稱VoIP)問世以來，聲音品質一直都是網路電話服務面臨的最大挑戰。隨著網路電話的使用者越來越多，其聲音品質也就越受到關注。以往聲音品質的量測方式，都是藉由控制影響聲音品質的變數並在使用者端進行測量。然而，在真實網路上，網路電話的管理者想要在使用者端進行聲音品質量測是非常困難的一件事。而目前市面上有多家軟硬體網路電話廠商，要實做一套能夠在各家產品上都適用的聲音品質量測系統幾乎是不可能的。
在此同時，由於近年來網路位址不足的原因，Network Address Translator (NAT，俗稱網路分享器)這個解決方法廣為佈建在大大小小的各種網路中。然而NAT的存在造成了某些點對點的網路應用服務受到影響，網路電話服務就是其中之一。為了讓網路電話能順利穿越NAT，目前的網路電話系統中都會建置一台RTP Proxy Server。
在本文中，我們將提出一套利用上述的RTP Proxy Server為輔助，在網路上測量聲音品質的量測系統。因為此量測系統並沒有在使用者端進行任何的修改，所以並不侷限於特定的軟硬體網路電話；此系統將量測後所得到的數據記錄下來，以圖形的方式呈現，希望能夠在使用者對於網路電話的聲音品質有疑慮的時候，提供給網路電話服務的管理者一份客觀的除錯依據。
我們並以實作驗證，利用此簡易量測法所獲得的數值，相較於利用使用者端設備進行量測的正規方式，誤差不到百分之一。可說是以較低的成本，所獲得一項相當令人滿意的量測成果。

關鍵字

網路電話 ； 聲音品質量測 ； RTP代理伺服器

並列摘要

Since the Voice over Internet Protocol (VoIP) application was introduced, voice quality has always been a big issue. As more and more people use VoIP applications, the quality issue now becomes critical. Traditionally, the measurement of voice quality has to perform the test on both client-sides. However, in a real network, it is not always possible for VoIP service providers to control the IP phone directly and measure the voice quality on client-sides. Because there are many VoIP products made from different manufacturers, right now, it is almost impossible to find a measurement system which is applicable to all VoIP products.
Meanwhile, in recent years, because of the exhaustion of IP addresses, Network Address Translator (NAT) was introduced to mitigate the shortage of IP addresses. Nevertheless, NAT causes serious problems for many peer-to-peer Internet applications, such as VoIP. Thus, VoIP applications need solutions for NAT traversal. For the past years, there are lots of NAT traversal mechanisms suggested, such as static assignment, Virtual Private Network (VPN), and relay-based proxy servers. RTP Proxy Server is a relay-based proxy server, which is the most popular one among these NAT traversal mechanisms. Nowadays, in most VoIP systems there exists a RTP Proxy Server to relay RTP packets and solve the problem of NAT traversal.
In this thesis, we design a monitoring system, named RTP-M which works with RTP Proxy Server to measure the VoIP quality. Because this system is independent with client-sides, it can be applied to any VoIP end devices. Moreover, RTP-M depicts the measured voice quality in graphical forms which are more intuitive for human beings. We hope that our RTP-M can provide VoIP administrators with the troubleshooting information when users have any complaint about voice quality.
Our implementation shows that, the voice quality measured by RTP-M, has only some negligible error compared to voice quality measured by the formal way on the client-sides. Considering the convenience and low lost, the precise is fairly satisfactory.

並列關鍵字

Voice over Internet Protocol (VoIP) ； Voice Quality Monitoring ； RTP Proxy Server

參考文獻

[1] ITU-T, “Method for subjective determination of transmission quality,” ITU-T Recommendation P.800, August 1996.

[2] ITU-T, “Perceptual evaluation of speech quality (PESQ): An objective method for end-to-end speech quality assessment of narrow-band telephone networks and speech codecs,” ITU-T Recommendation P.862, February 2001.

[3] ITU-T, “The E-Model, a computational model for use in transmission planning,” ITU-T Recommendation G.107, March 2005.

[4] R. G. Cole, J. H. Rosenbluth, “Voice over IP Performance Monitoring,” ACM Computer Communication Review, vol. 31, no. 2, pp.9-24, April 2001.

[5] J. Rosenberg, H. Schulzrinne, G. Camarillo, A. Johnston, J. Peterson, R. Sparks, M. Handley, E. Schooler, “SIP: Session Initiation Protocol,” IETF RFC 3261, June 2002.