jueves, 29 de julio de 2010


IP telephony testbed

Within Mitretek Systems' Advanced Telecommunications Laboratory (ATL), a VoIP testbed has been set up. The testbed consists of two PC clients, two IP gateways, four analog phones, and a PC-based private branch exchange (PBX) (see Figure 1). The purpose of the testbed is to measure and calibrate traffic generated from the voice conversations on the PCs and the analog phones. To collect this information, a packet sniffer, Observer from Network Instruments, has also been installed in the VoIP testbed.

Figure 1: IP telephony testbed connectivity
The IP gateways use the standard H.323 call control protocol and several standard codecs: GSM120, G.711, and G.723.1. The PC-based PBX served as a voice switch in the testbed. The PC clients were connected to a hub with private IP address space. Different codecs were used in measuring the packet delay, jitter, packet loss, and bandwidth utilization.
Both PC-to-PC and phone-to-phone testing was conducted in an intranet environment. In the phone-to-phone experiment, the end-to-end delay was measured as 274 ms. A large portion of the delay is due to the analog/digital conversion, compression, packetization, and OS overhead in the IP gateways. When faster digital signal processors (DSPs) become available, the delay introduced by the equipment should be significantly reduced. The purpose of conducting intranet testing is to provide a performance baseline. When this experiment is expanded to the Internet, the additional delay elevated by Internet can be better understood.
Table 1: Packet size distribution over 30 seconds

Packet Size (Bytes)
<=64 65-84 85-128 129-512 513-1024 >1024
6.4 kbps
Listener 0.20% 81.20% 0.70% 0.30% 0.00% 17.50%
Talker 0.10% 98.30% 0.90% 0.70% 0.00% 0.00%
5.333 kbps
Listener 0.00% 81.40% 1.10% 0.30% 0.00% 17.20%
Talker 0.10% 98.10% 1.30% 0.50% 0.00% 0.00%
64 kbps
Listener 2.30% 0.30% 1.90% 50.60% 0.00% 44.80%
Talker 4.20% 0.60% 3.50% 91.70% 0.00% 0.00%

Table 1 represents the results from a 30-second half-duplex voice session. The columns are packet sizes. With the G.723.1 coder, 81.2% of the packets from the Listener were 64 to 84 bytes in size. When using G.711, the talker sends data out in 310 frames. The remaining packets from the talker to listener are called control packets from the application ranging in size from 64 to 128 bytes. When using the G.723 codecs, the data are in 78-byte frames. When the listener sends data to the talker, the packet size is between 84 and 100 bytes, regardless of the codec. Other packets are from the Observer software or are call control packets from NetMeeting.
Additional tests will be taken in a mixed voice and data environment. The data behavior is highly application dependent. Instead of using a generic data generator to produce the necessary data load, a script language, SILK by Sagueway, which emulates the application transactions, will be programmed to produce Microsoft Word, Microsoft Excel, Microsoft PowerPoint, and Web traffic. It is felt that combinations of these application transactions represent the actual network data behavior better than the traffic generated by the network management device. However, the emulation approach is appropriate only for a reasonable system size. For a large-network system, simulation and analytic models are required to facilitate the analysis.

C.I 18878408

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