Skype over UMTS. Tobias Hoßfeld www3.informatik.uni-wuerzburg.de. University of Würzburg Informatik III (Distributed Systems) Prof. Dr. P.

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1 Inmatik III () Prof. Dr. P. Tran-Gia Skype over UMTS Talk (25+5min) ITG Fachgruppe VoIP over Wireless 15th May 2006, WürzburgW

2 P2P Applications Across Mobile Networks When Do We Need Rate Control Dedicated Channels in UMTS? Source Traffic Model Wireless System Permance and perceived quality of service? P2P Application Measurements in a Laboratory UMTS Network with time-varying Loads Measurement and Analysis of Skype VoIP Traffic in 3G UMTS Systems 2

3 Measurement and Analysis of Skype VoIP Traffic in UMTS, Andreas Binzenhöfer, Kurt Tutschku, Institute of Computer Science, Department of. Würzburg, Germany. Markus Fiedler Blekinge Institute of Technology, Department of Telecommunication Systems. Karlskrona, Sweden. Second EuroNGI Workshop on Wireless and Mobility WP.IA.8.2 and WP.IA.8.3 Lake Como, Italy,

4 Skype over UMTS Mobile VoIP as inexpensive alternative voice calls Skype VoIP application is very popular users online on Thursday, , 11:27 UMTS UMTS Access Access Network Internet VoIP user not interested in network permance indicators Perceived quality of service is measured (PESQ values) in a test-bed emulating rate-controlled dedicated channels and dynamic changes of the (emulated) network in a public German UMTS network 4

5 Bottleneck LAN: Emulated Rate-Controlled DCH in UMTS Bottleneck LAN scenario using a traffic shaping router We start a voice call with 16 kbps and increase up to 384 kbps Cisco Cisco dual dual Ethernet Ethernet router router shapes shapes outgoing outgoing traffic traffic both both interfaces interfaces gateway gateway signaling signaling traffic traffic to to the the Internet Internet Internet kbps Skype Skype user user A A sends sends to to B B Skype Skype user user B B receives receives cable sound warding records records into into wav-file wav-file original original and and received received wav-file wav-file required required PESQ PESQ computation computation 5

6 Reference PESQ Value w/o Influence of Network PESQ is not linear and very sensitive at its upper value 4.5 is best, -0.5 is worst quality Measurement setup does not falsify PESQ value measurements original wav-file 4.50 encode decode sound has to be warded to another machine in order to save it to a wav-file on disk degenerated wav-file 4.02 cable received wav-file 3.93 Remarks: ilbc codec (instead of ISAC) used due to 500 MHz CPU 6

7 Bottleneck LAN: Throughput and Packet Interarrival Times Throughput includes 67 Bytes payload, 28 Bytes UDP/IP, 14 Bytes Ethernet header Total size of Skype 872 bit Packets are sent every 30 ms Throughput of sender is about 29 kbps 30 Router has a buffer of 8 kbit 9 s fit into buffer Bottleneck link of 16 kbps Buffer is emptied every 500 ms 500 ms / 30 ms ~ 17 s, i.e. 8 lost s or 46% loss ratio 1 throughput [kbps] min/mean/max throughput of sender min/mean/max throughput of receiver CDF s in buffer are then transmitted buffer is via emptied 100 Mbps every LAN500 ms interarrival times of sent s interarrival times of received s bandwidth [kbps] bandwidth restricted to 16 kbps interarrival time [ms] 7

8 Bottleneck LAN: PESQ and Packet Loss Ratio Linear relationship between loss and PESQ bottleneck links below 32 kbps 8

9 Measurement Setup in a Public UMTS Network Skype Skype user user A A via via UMTS UMTS sends/receives sends/receives uplink: uplink: kbit/s kbit/s downlink: downlink: kbit/s kbit/s UMTS UMTS Access Access Network Internet uplink: uplink: kbit/s kbit/s downlink: downlink: kbit/s kbit/s Skype Skype user user B B via via DSL DSL receives/sends receives/sends DSL DSL Access Access Network 9

10 Packet Interarrival Times (PITs) UMTS Uplink UMTS client constantly sends a voice every 60 ms of 108 Byte payload Another ilbc codec is used Due to jitter PITs at DSL receiver spread around mean PESQ 2.24 instead of 2.95 in bottleneck LAN with 64 kbps UMTS s are sent at a discrete resolution of 1 ms Discretization happens at sender, probably by the PCMCIA UMTS card deviation of sent and received throughput differs about 1200bps CDF DSL receiver with 1024 kbps UMTS sender with 64 kbps s (sorted by arrival times) UMTS sender with 64 kbps DSL receiver with 1024 kbps interarrival time [ms] interarrival time [ms] 10

11 Packet Interarrival Times (PITs) UMTS Downlink Packets of 108 Byte are sent by DSL sender every 60 ms Interarrival times of s arriving at UMTS receiver are discrete at 40, 60, and 80 ms UMTS transmission time interval (TTI) has a value of 10 ms Every 5th is retransmitted (FER adaptation of outer loop?) Deviation of sent and received throughput differs about 300 bps 1 80 CDF interarrival time [ms] UMTS receiver with 384 kbps DSL sender with 128 kbps interarrival time [ms] UMTS receiver 384 kbps s (sorted by arrival times) 11

12 Emulate Dynamic Changes in UMTS Dynamic scenario using a software tool: NistNet CPU power of machines 1.2GHz ISAC codec used shapes shapes traffic traffic only only connection connection between between Katie Katie and and Salma Salma gateway gateway signaling signaling traffic traffic to to the the Internet Internet eth0 eth2 eth1 Internet Katie Katie Salma Salma sends sends to to B B receives receives Audio file (51s) is repeated with a pause of 5s in between PESQ value computed intervals of 56s Network characteristics evaluated using moving average (5min) 12

13 ISAC Codec Reacts on Dynamic Changes Packet sent times depend on codec, independent on loss Variable bit rate by increasing size, i.e. more Packet loss loss receiver sender sender Time between s [ms] Application adapts onto Time [ms] experienced x 10 6 end-to-end Time [ms] x 10 6 quality (here: network measurements) Packet loss 0.2 loss size 200 Packet sizes 13

14 Dynamic Scenario: PESQ and Packet Loss PESQ computation: 51s s within 50min (synchronization) 0.35 Packet loss PESQ Time [ms] x

15 Dynamic Scenario: PESQ and Jitter Jitter is defined as standard deviation of end-to-end delay Mean end-to-end delay is set to 60 ms Jitter [ms] PESQ Time [ms] x

16 Dynamic Scenario: PESQ and RTT RTT>500ms results in strong PESQ degradation If RTT>4s Skype relays connection over third party machine PESQ Application-driven routing based on experienced endto-end quality (here: network measurements) max 2.8 mean min round trip time [ms] 16

17 Application-driven QoS Adaptation and Routing based on experienced end-to-end quality connection is relayed application re-routing Internet Skype user C Skype user A based on experienced end-to-end quality codec is changed QoS adaptation Skype user B 17

18 Conclusion and Outlook Measurement of Skype considering the perceived voice quality UMTS sufficient to make mobile VoIP calls with Skype possible Variable bit rate ISAC codec used if CPU power above 600 MHz Connection relayed if... loss too high (>25%) round trip time too high (>4s) Based on experienced end-to-end quality Skype implements dynamic QoS adaptation onto environment application-driven re-routing intelligence is moved to the edge Outlook Impact of noise traffic / network congestion on the PESQ Characterization of Skype traffic patterns 18

19 Thank you! Any questions? Answers Melanie Brotzeller Michael Duelli Tommy Zinner Frank Bennewitz 19

1 University of Würzburg. Institute of Computer Science Research Report Series. Measurement and Analysis of Skype VoIP Traffic in 3G UMTS Systems

1 University of Würzburg. Institute of Computer Science Research Report Series. Measurement and Analysis of Skype VoIP Traffic in 3G UMTS Systems University of Würzburg Institute of Computer Science Research Report Series Measurement and Analysis of Skype VoIP Traffic in 3G UMTS Systems Tobias Hoßfeld 1, Andreas Binzenhöfer 1, Markus Fiedler 2,

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