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1、QoS Measurement of VoIP End-pointsKazuumi Koguchi Wenyu Jiang Henning SchulzrinneInformation Technology R&D Center, Mitsubishi Electric Corporation 5-1-1 Ofuna, Kamakura, Kanagawa, 247-8501Department of Computer Science, Columbia University 1214 Amsterdam Avenue, New York, NY, 10027 U.S.A.E-mail: ko
2、guchiisl.melco.co.jp, wenyu, hgscs.columbia.eduAbstract VoIP (Voice over IP) is a service that requires synergy between the underlying network for transport and the end-points responsible for voice processing. We evaluate the end-to-end quality and performance of several VoIP end-points. In particul
3、ar, we focus on the following aspects: mouth-to-ear (M2E) delay, clock skew and behavior under packet loss. Our measurement results show that M2E delay depends mostly on the receiving end-point, and when hardware IP phones act as receivers, they achieve low average M2E delay (45-90ms) in a LAN envir
4、onment. For software VoIP clients as receivers, their average M2E delays range from 65ms to over 400ms. We find that all tested hardware IP phones support some form of packet concealment and it works well for up to two consecutive losses at 20 ms packet intervals.Keyword IP Telephony, voice over IP,
5、 quality of service, end-to-end, delay, jitter1. IntroductionRecently VoIP (Voice over IP) service has become popular and several kinds of VoIP end-points have been commercialized. There are extensive literature on netrwork quality of service (QoS), such as in the areas of Diffserv 1 and Intserv 2.
6、However, little has been done on studying the QoS of VoIP end-points. VoIP is a service that requires synergy between the network for transport and the end-points responsible for voice processing. Depending on the implementation, such as what playout algorithms these end-points use, whether they are
7、 hardware or software based, their performance may differ dramatically. Therefore it is worth to measure and compare QoS using VoIP end-points.In this paper, we evaluate the QoS of a number of VoIP end-points, including hardware based IP phones and software based end-points. We examined the followin
8、g QoS aspects, mouth-to-ear (M2E) delay, clock skew and behavior under packet loss. The rest of the paper is organized as follows. Section 2 describes our experiment setup. Section 3 presents the measurement results and section 4 concludes the paper and lists future work.2. Experiment Setup2.1 Measu
9、rement system configurationOur measurement system configuration is shown in Fig. 1. We measure mouth-to-ear delay by recording both the original audio and the receivers output audio in a two-channel (stereo) mode simultaneously. Using a fork cable, the original audio is output to both an VoIP end-po
10、int and a PC for recording. The end-point is connected to another end-point through LAN. The receivers output is also connected to the PC for recording. After recording we measure delay between the original audio and receivers output. We use a software which calculates a most likely delay offset bas
11、ed on auto-correlation in the frequency domain. We have confirmed its precision is within 1ms by using an audio mixer which can insert a known delay offset.Fig. 1 Measurement System Configuration2.2 End-point devicesWe evaluate several IP phones which are shown in Table 1. Phone (d) is a 1-line PSTN
12、/IP gateway. sipc 3 is a SIP 4 user agent and developed by IRT lab in Columbia university. Phone (e), (f) and (g) are software VoIP clients. We installed (e) and (f) on a AMD K7 PC with Windows 2000/XP dual-boot, and a Pentium-3 notebook with Windows XP. Phone (g) is a software client which can call
13、 from PC to PSTN. In addition, we used mobile phone to measure delay to PSTN for reference.Table 2 shows some parameters in the experiments.Table 1 List of tested end-pointsEnd-pointtype/platformphone (a)hardwarephone (b)hardwarephone (c)hardwarephone (d)gateway, hardwaresipcSolaris, Ultra10phone (e
14、)Win 2000/XP(K7)& Win XP(P3)phone (f)Win 2000/XP(K7)& Win XP(P3)phone (g)Win NT(P2)GSM phoneGSM 1900 USTable 2 Parameters in the experimentsEnd-pointcodecsilencesuppressionpacketintervalphone (a)G.711 -lawN20msphone (b)G.711 -lawG.729YY20ms20msphone (c)G.711 -lawN20msphone (d)G.711 -lawN30mssipcG.71
15、1 -lawN20msphone (e)G.711 -lawY20msphone (f)G.723.1Y30msphone (g)?Y60ms3. Measurement Results3.1 Mouth-to-ear delay without jitter Fig.2 shows an example plot of M2E delay over time. The notation experiment 1-2 means part 2 of call no.1 and experiment 2-1 means part 1 of call no.2. Between call no.1 and call no.2 we hang up and call up. There is a short pause between part 1 and part 2 to save part 1s audio to disk. We transmitted the same audio data for each experiment.Fig. 2 M2E delay from phone (a) to (b)In experiment 1-1, the M2E delay from phone (a) t
