Measurement and Analysis of Skype VoIP Traffic in 3G UMTS Systems

1CDF0.80.60.40.2interarrival timesof sent packetsinterarrival timesof received packetspayload observed interarrival[Byte] packets timesender 3 3 20.05 ssender 67 1888 30.00 msrcver 3 3 20.25 srcver 67 1039 54.43 ms00 200 400 600packet interarrival time [ms]Figure 9: Single run with 16 kbps bottleneck and 8000 bit buffer in the routerpackets. At first glance, the CDF of the receiver has a very unexpected shape. About 90 percentof all packets have an interarrival time of practically 0ms, while the time between the remainingpackets is about 500ms. The buffer in the router was set to 8000 Bit, while simultaneouslylimiting the speed of the link to 16 kbps. Skype used a total packet size of 872 Bit. Thus, atmost 9 packets (872*9= 7848 Bit) fit into the buffer of the router. To emulate a link speed of16 kbps, the router fills its buffer and delays the data for exactly 500ms. This way, a bandwidthof 8000 Bits/500ms = 16 kbps is achieved on a physical 100Mbit/s link. This has two majorimplications. At first the interarrival time of the packets within a burst is 872 Bit/100 Mbit/s,which is in the range of 10 −6 s ≈ 0 ms and explains the shape of the CDF in Figure 9. Secondly,packet loss occurs in bursts during the 500ms, in which the buffer of the router is delayed.Finally, the table in Figure 9 shows the payload size and the observed number of packets ofthis size for both the sender and the receiver, as well as the corresponding interrarival times. Thepackets with a payload of 67 Bytes represent the voice connection. Only 1039 of the 1888 sentpackets arrive at the receiver, which corresponds to a packet loss of about 45 percent and explainsthe increase in the average interarrival time from 30ms to 54.43ms. An interesting observation isthat Skype obviously sends some smaller packets (3 Bytes) every 20 seconds, which we believeto be quality feedback. This would also explain how a Skype client is able to display the packetloss of the remote side in its technical info window during an active call.4.2 Measurements in UMTS NetworkAll experiments in this section were done using our UMTS setup as described in Section 2.2.During all measurements the clients used a different codec for the main audio connection, sending108 Byte of payload every 60ms instead of 67 Byte every 30ms like in the previous scenarios.Since this codec was used starting with the first audio packet, Skype seems to choose this codecbased on local information, like access type (modem or LAN) to the Internet, or due to exchangedpackets before measuring the link quality. This assumption is supported by the fact,that emulating the link properties (delay, bandwidth) with dummynet did not cause Skype to usethe same codec. There was nearly no packet loss in any of the experiments. In total, 11 out of8