Self-clocked Rate Adaptation for Conversational Video in LTE

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CAPITALS

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SELF-CLOCKED RATE ADAPTATION FOR CONVERSATIONAL VIDEO IN LTE CSWS’14

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Self-clocked Rate Adaptation for Conversational Video in LTE | ACM 978-1-4503-2991-0/14/08 | © Ericsson AB 2014 | Page 2

It’s not only about bitrate

Julia rehearsing for a figure skating show

with her friend Hilda over Skype

Latency is important too !!

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› Two types

– Rate based congestion control

› GCC = Google CC (congestion control)

› MTSI (3GPP TS26.114)

– Self-clocked congestion control

› TCP

› TFWC (TCP Friendly Window based Congestion Control)

› SCReAM (Self-clocked Rate Adaptation for Multimedia)

Endpoint congestion control

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› Rate based CC seems

to be unable to combat

congestion in LTE well

› Root cause, packet

conservation principle is

not followed.

› ECN is one solution to

problem

Endpoint congestion control, the problem

ECN :Explicit Congestion Notification

* http://www.uppersideconferences.com/volte2011/volte2011program.html

“Expanding VoLTE With Video Capabilities”

*

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› File transfer with TCP

LEDBAT

› Low latency despite

quite a high bitrate !

› Packet conservation

principle followed

› Can the same principles

be used for

conversational video ?

Endpoint congestion control, the solution ?

36 38 40 42 44 46 48 50 52 540

0.1

0.2

0.3

0.4TCP segment delay [s]

36 38 40 42 44 46 48 50 52 540

1

2

3

4

5

6Throughput [Mbps]

T[s]

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› Example use case

– WebRTC

› Audio + video

› Possibly also a data

channel

› Varying number of

streams

Properties of conversational media

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› In general no guaranteed bitrate *

› LTE bearers are allocated to individual

terminals

– Low statistical multiplexing

› Throughput dictated by channel quality,

number of users in cell and historical

bitrate.

– May change frequently, very little grace time

to reduce bitrate

Properties of LTE channels

* QoS framework enables to prioritize e.g. for a given minimum bitrate

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1. Low latency and packet loss

2. Ability to coexist with other traffic over the same bearer

3. Stable bitrate

› #3 may be hard to fulfill given the two other requirements

Requirements on a solution

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› IETF : draft-johansson-

rmcat-scream-cc-02

› Chromium

implementation in

progress

The adaptation framework

RTP

Congestion

control

Video coder

Queue

RTP packets

Rate

control

UDP socket

Transmission

scheduler

Queue

length

RTCP

TSTX

RTPSN, RTPsize

Frame

skip

Rate

adjustment

CWND, RTT

bytesInFlight

UDP socket

Dejitter

buffer

(optional)

Video decoder

RTP

Tx channel

RTCP report

generator

RTP payload

decapsulator

TSRX

RTPSN, SSRC,

Sender Receiver

Frame

drop

~TCP

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› RFC4585 Transport layer feedback message

RTCP feedback

0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |V=2|P| FMT | PT | length | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | SSRC of packet sender | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | SSRC of media source | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Timestamp (32bits) | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Last recv. sequence nr (16b) | ECN echo |Q|R|R|R|R|R|R|R| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | ACK vector (32b) |

+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

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› CWND : Congestion window, determines max number of

bytes in flight, size depends on estimated network queuing

delay and packet loss or ECN

› Bytes in flight: The total size of the transmitted but not yet

acknowledged RTP packets.

› OWD : One way (extra) delay. An estimate of how much

the queues in the network increase

Buzz words

SNSN-1SN-2SN-3SN-4SN-5SN-6SN-7Transmitted

RTP SN

ACKed

RTP SNSN-3SN-5SN-6SN-7 Bytes in flight

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› Inspired by TCP and LEDBAT

› Additional features

– Adaptive delay target

– Random CWND reduction

Congestion Control OWD’ = TSRx-TSTx

Update baseDelay

OWD = OWD’-baseDelay

bytesInFlight -= ackedBytes

lossEvent

Done

Flowchart is executed

for each received

RTCP packet

Yes

No

baseDelay is the

min OWD’ over a

sliding window

offTarget = (owdTarget-owd)/

owdTarget

cwnd += gain * offTarget *

ackedBytes * maxMtu /

cwnd

doFastStart = false

doFastStart = false

cwnd *= beta

cwndHdRoom =

maxBytesInFlight*K

cwnd = min(cnwd,cwndHdRoom)

maxBytesInFlight is the

max number of bytes in

flight over a 1s window

K is an extra headroom

factor (e.g 1.5)

OWD < owdTarget/2

&&

doFastStart

No

cwnd +=

ackedBytes

Yes

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› Additional feature

– Allow transmission even though bytes in

flight > CWND in certain cases.

Transmission Scheduling

Done


every Tpace

Yes

RTPsize = size of next

RTP paket in queue

Transmit next RTP packet

bytesInFlight += RTPsize

Tpace = RTPsize/

max(50000,BW)

No

Sender queue

Empty ?

Tpace =

0.001

BW = CWND*8/RTT

bytesInFlight +

RTPsize<CWND?

Yes

No

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› Additional actions, in

case sender queue

grows too large

– Skip frames

– Drop RTP packets

Media Rate Control Estimate

Davg: Average sender

queue delay

Davg > Dth

||

lossEvent?

Davg == 0 ? doFastStart = false

bitRate *= 1.0-Davg

doFastStart ==

true ?

rampUpT =

rampUpTnormal

rampUpT =

rampUpTfast

incr = Tfr/rampUpT

bitRate *= (1.0+incr)

Done


every frameperiod Tfr,

for instance every

Tfr=40ms

Yes

Yes

No

No

No

Yes

lossEvent ?

bitRate *= beta

YesNo

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› Note the more sparse ACKs during the congestion event

~15.5 – 16.0s

Example 1 LTE channel

0

0.1

0.2

Frame delay and OWD [s]

0

0.2

0.4

0.6

Temp. bitrate [Mbps]

14 14.5 15 15.5 16 16.5 170

5000

10000

T[s]

Network congestion control

Video frame delay

Audio frame delay

OWD

Bytes in flight

CWND

Bytes newly ACKed

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› Rate based CC increases

bitrate very slowly low

average bitrate

› Self-clocked CC

increases bitrate quickly

High bitrate at low load

› Still Self-clocked CC is

more stable at higher

load levels

Comparison Self-clocked vs Rate based CC

2 4 6 8 10 12 14 160

200

400

600

800Video frame 95%ile, user 90%ile latency [ms]

2 4 6 8 10 12 14 160

1

2

3

4

5Average packet loss rate [%]

Users/cell

Rate based CC

Self-clocked CC

2 4 6 8 10 12 14 16

200

400

600

800

1000

1200

Users/cell

Average bitrate [kbps]

Rate based CC

Self-clocked CC

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› Self-clocked rate adaptation performs considerably better

than rate based congestion control algorithms

› Possible remaining issues to solve

– Over-reaction at handover

– Over-reaction to congestion events

– Negative impact from other cross traffic

– Possibly frequent and large media bitrate changes

› Possible solution to the above listed issues

– ECN

– Network assisted rate adaptation

Discussion

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› Possible to implement well functioning conversational video

over LTE

› Self-clocking is the key !

› Leverages on novel TCP features

› Solution realized with RTCP feedback

– Solution can be complemented with more network centric support

for enhanced performance

Conclusion

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› Three scenarios

Fairness enforcement

Host E

Host F

Host DHost A

Host B

Host C

Video A

Video B

Video C

A’s bitrate is higher than B’s and

C’s. Fairness enforcement in host

A, B and C tries to balance by

means of random CWND reduction

(5-10%) and frame blanking.

Video A

Video B

Video B

Scenario 1

Difficult

Host A

Host B

Host C

Host D

Video A

Video B

Video C

A

B

C


C’s. Source quench bit is set in A’s

feedback. A’s CWND is reduced

proportional to the amount of

feedback with source quench bit

set. B and C grabs free capacity

BW

manager

Scenario 3

Simple Host A

Host B

Host C

Host D

Video A

Video B

Video C

A

B

C


C’s. Bandwidth manager in host C,

balances the bitrate allocation.

BW

manager

Scenario 2

Simple

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› Scatter plots of IP packet delays

vs video frame delay indicate

additional queuing in sender

queue is strongly correlated to

network queuing.

› Conclusion: Without a sender

queue, the delay would likely

occur in the network instead

Additional delay in sender queue

0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.40

0.1

0.2

0.3

0.4

0.5

0.6

0.7

0.8

Vid

eo

fra

me

de

lay [s]

IP packey delay [s]

Video and IP packet delays, 1s average, scatter plot

0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.40

0.1

0.2

0.3

0.4

0.5

0.6

0.7

0.8

Vid

eo

fra

me

de

lay [s]

IP packey delay [s]

Video and IP packet delays, 1s average, scatter plot

Downlink, 4 users/cell

Downlink, 10 users/cell

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› SCReAM reacts quickly to reduced throughput.

Example 2 Bottleneck with changing BW

0 5 10 15 20 25 30 35 40 45 500

0.2

0.4

0.6Video frame delay and OWD [s]

0 5 10 15 20 25 30 35 40 45 500

1

2


0 5 10 15 20 25 30 35 40 45 500

1

2

x 104 Network congestion control

T[s]

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› Three features :

– Fast start

– Adaptive delay target

– Congestion window validation

Example 3 Competing FTP

0 5 10 15 20 25 30 35 400

0.2

0.4Frame delay and OWD [s]

0 5 10 15 20 25 30 35 400

0.5

1


0 5 10 15 20 25 30 35 400

1

2

x 104

T[s]

Network congestion control

Video frame delay

OWD

Delay target

CWND

Bytes in flight

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› Congestion window validation is crucial for the stability of

the algorithm

Example 4 Congestion window validation

0 10 20 30 40 50 60 70 800

0.2

0.4

IP packet delay [s]

0 10 20 30 40 50 60 70 800

1

2

Throughput [Mbps]

0 10 20 30 40 50 60 70 800

2

4

6

8x 10

4 CWND and bytes in flight

T [s]

CWND

in flight

0 10 20 30 40 50 60 70 800

0.2

0.4

IP packet delay [s]

0 10 20 30 40 50 60 70 800

1

2

Throughput [Mbps]

0 10 20 30 40 50 60 70 800

2

4

6

8x 10

4 CWND and bytes in flight

T [s]

CWND

in flight

CWV off CWV on

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› RTT = 100ms

› 3 RMCAT flows compete

Competing RMCAT flows

0 20 40 60 80 100 120 140 160 180 2000

500

1000

1500

Throughput [kbps]

0 20 40 60 80 100 120 140 160 180 2000

0.05

0.1

0.15

0.2

0.25

Video frame delay [s]

T[s]

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› 5 flows with different RTT

RTT fairness

0 10 20 30 40 50 600

500

1000

1500

Throughput [kbps]

0 10 20 30 40 50 600

0.1

0.2

0.3

0.4Video frame delay [s]

T[s]

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› Two RMCAT flows

compete, changing

bandwidth

› RTT 200ms

Variable BW

0 20 40 60 80 100 1200

500

1000

1500

Throughput [kbps]

0 20 40 60 80 100 1200

0.2

0.4

Video frame delay [s]

0 20 40 60 80 100 1200

5

10AQM packet loss rate [%]

T[s]

Documents

Self-clocked Rate Adaptation for Conversational Video in LTE