Measurements of Congestion Responsiveness of Windows Streaming Media (WSM)

Presented By:-

Ashish Gupta

Roadmap

Introduction

Graphs and Observations

Conclusion

Comments

Introduction

AIM: Characterize the bitrate response of

“Intelligent streaming” and content “encoding” of WSM

Variable Parameters: Network characterstics like bottleneck

bandwidth Content encoding characterstics

Windows Streaming Media (WSM)

Intelligent Streaming: Content bitrate is adjusted according to current

available bandwidth Server and Client determine current bandwidth

Multiple bitrate encoded files are used for streaming

Intelligent degradation in image quality is done first (both server and client support)

Audio is reconstructed to preserve quality

WSM (Contd.)

WSM server decides the current available bandwidth

It requires multiple bitrate encoded stream support by the media creator

Some media data parameters like Thinning, Maximum bitrate etc. can be set during transfer initiation.

Roadmap

Introduction

Experimental Setup

Graphs and Observations

Conclusion

Comments

Experimental Setup (as shown in the paper)

Setup (Contd.)

Client has media tracker s/w for capturing the client side performance statistics

Each side of the router has linux machine capture offered and achieved network loads.

Internet behavior is achieved by router by setting the internet latencies (normal internet traffic)

Roadmap

Introduction

Experimental Setup

Graphs and Observations

Conclusion

Comments

Graphs

All graphs in the next few slides consider single bitrate encoding

Network bandwidth 725 Kbps and latency set to measured internet latency equal to 45 ms

WSM provided with 60 sec clip encoded at 340 Kbps, roughly equal to fair capacity

Graphs (Contd.)

Observations

20-40 percent of packet loss occurred during buffering phase

WSM playing can be partitioned into buffering phase and playout phase

Previous experiment is repeated with 540Kbps clip which is much above the fare share bandwidth

Graphs

Observations

Media Thinning functionality provided by “intelligent streaming” was NOT used

Competing TCP flow was denied of fair share of available capacity

During buffering period heavy packet losses were encountered

Graphs

Observations

No media thinning during the buffering phase

Bitrate is increased even during losses during buffering phase (Fire-hose approach)

Transmission rate decreased below fair share bandwidth during playout phase

Graph

Now buffering phase and post-buffering phase is studied independently

Graphs are plotted “transmission bitrate of WSM/TCP” vs

“Content encoded bitrate” “loss rate” vs “Content encoded bitrate”

Different capacity n/w is considered without induced losses for “Buffering phase only”

Graphs (Buffering: 250Kbps capacity)

Graphs (Buffering: Capacity 725 Kbps)

Graphs (Buffering: Capacity 1500 Kbps)

Observations

Buffering rate is proportional to content encoding rate until the encoding rate exceeds bottleneck capacity

Beyond this loss rate can be as high as 80%

WSM has high loss rate due to higher sending rate

Observation (Contd.)

The behavior of WSM changes between 340 Kbps-548 Kbps

At 548Kbps encoding rate is there is a significant drop in loss rate

Graphs (Playout: Capacity 250 Kbps)

Graphs (Playout: Capacity 725 Kbps)

Graphs (Playout: Capacity 1500 Kbps)

Observations

In post-buffering bit rate is proportional encoding bit-rate till bottleneck capacity

For still higher encoding bit rate incurs losses upto 40 percent initially

Bitrate goes lower then TCP flow due to media thinning

Accurate information of “thinning” behavior over post buffering period is missed due to average values

Graphs: Type of traffic (340 Kbps)

Packet Sequence number vs Time for 340 Kbps clip

Observations

WSM traffic is bursty in nature

Retransmission happens for any dropped packet

Reason for spikes: shortening of transmission time between packets bursts

Graphs (Multiple bit rate)

Aim: Explore number of bitrate vs responsiveness

Ten set of clips were used 1st clip (1128 Kbps) 2nd clip (1128, 764 Kbps) . 10th clip (1128,.. 764, 548,.., 282, 148..Kbps)

Study of Bitrate vs lowest bitrate contained in the clip is made

Graph (Buffering: Multiple bit rate)

Graph (Buffering: Multiple bit rate)

For Buffering: WSM chooses the bitrate which is just lower than bottleneck bandwidth otherwise it takes the lowest capacity available

WSM specific observations

Bottleneck bandwidth: Available bandwidth is measured in WSM using

three large packets. Two packet pair estimate is used to get the bottleneck bandwidth

Bottleneck bandwidth is estimated only once before each session and then RTCP messages are used to control retransmission

Frequency of RTCP message increases with increasing loss rate

Graphs (Induced losses)

Loss induced by router to check the behavior of WSM in case of loss due to network

For further discussion bottleneck bandwidth: 725 Kbps Encoded bitrate: 548 Kbps (largest below 725

Kbps)

Graph (Buffering: network induced loss)

Observations (Buffering: network induced loss) With high loss induced loss rate

TCP decreases its flow WSM increases its flow to compensate for

high loss rate

During the initial losses WSM buffering uses “fire-hose” approach

Graph (post-buffering: network induced loss)

Observations (post-buffering: network induced loss) Loss rate of 3-5% causes WSM to thin stream

Above 5% loss rate there is no change in thinned bitrate

Graphs (Induced losses MBR clip)

Next experiment uses multiple bitrate (MBR) encoded ( 548, 340, 282, 148, 106, 58…. Kbps)

Graphs (Induced losses with MBR)

Graphs (Induced losses with MBR)

Instead of thinning it chooses to stream at lower bitrate

Graphs (sudden change in induced loss)

This shows that WSM is unaffected by the sudden change in network loss

WSM model

WSM model: Should have two phases buffering and post

buffering Bursty Buffering is TCP unfriendly In some post buffering period WSM is more

than TCP friendly

Conclusion

During buffering TCP friendliness can only be achieved if encoded bitrate is less then estimated capacity. Otherwise, buffering is done at encoding rate

During playout Thinning or streaming low bitrate encoded

stream is done This can also be TCP unfriendly if encoding

rate is more than half capacity and less than full capacity

Conclusion

This study shows that content provider can make judicious decision encoding rates number of encoding levels

It provides the researcher with more accurate model of streaming media traffic

Comments

Study is exhaustive

It doesn’t propose analytical model for estimation of number encoding bitrates levels and there values