Self-Clocked Rate Adaptation for Multimedia
draft-ietf-rmcat-scream-cc-00
The information below is for an old version of the document.
| Document | Type |
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| Authors | Ingemar Johansson , Zaheduzzaman Sarker | ||
| Last updated | 2015-05-03 | ||
| RFC stream | Internet Engineering Task Force (IETF) | ||
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draft-ietf-rmcat-scream-cc-00
RMCAT WG I. Johansson
Internet-Draft Z. Sarker
Intended status: Informational Ericsson AB
Expires: November 3, 2015 May 2, 2015
Self-Clocked Rate Adaptation for Multimedia
draft-ietf-rmcat-scream-cc-00
Abstract
This memo describes a rate adaptation algorithm for conversational
video services. The solution conforms to the packet conservation
principle and uses a hybrid loss and delay based congestion control
algorithm. The algorithm is evaluated over both simulated Internet
bottleneck scenarios as well as in a LTE (Long Term Evolution) system
simulator and is shown to achieve both low latency and high video
throughput in these scenarios.
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on November 3, 2015.
Copyright Notice
Copyright (c) 2015 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
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the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
1.1. Wireless (LTE) access properties . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Overview of SCReAM Algorithm . . . . . . . . . . . . . . . . 3
3.1. Congestion Control . . . . . . . . . . . . . . . . . . . 4
3.2. Transmission Scheduling . . . . . . . . . . . . . . . . . 5
3.3. Media Rate Control . . . . . . . . . . . . . . . . . . . 5
4. Detailed Description of SCReAM . . . . . . . . . . . . . . . 5
4.1. SCReAM Sender . . . . . . . . . . . . . . . . . . . . . . 5
4.1.1. Constants and Parameter values . . . . . . . . . . . 7
4.1.2. Network congestion control . . . . . . . . . . . . . 11
4.1.2.1. Congestion window update . . . . . . . . . . . . 12
4.1.2.2. Transmission scheduling . . . . . . . . . . . . . 16
4.1.3. Video rate control . . . . . . . . . . . . . . . . . 17
4.2. SCReAM Receiver . . . . . . . . . . . . . . . . . . . . . 19
5. Feedback Message . . . . . . . . . . . . . . . . . . . . . . 20
6. Discussion . . . . . . . . . . . . . . . . . . . . . . . . . 22
7. Conclusion . . . . . . . . . . . . . . . . . . . . . . . . . 22
8. Open issues . . . . . . . . . . . . . . . . . . . . . . . . . 22
9. Source code . . . . . . . . . . . . . . . . . . . . . . . . . 23
10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 23
11. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 23
12. Security Considerations . . . . . . . . . . . . . . . . . . . 23
13. Change history . . . . . . . . . . . . . . . . . . . . . . . 23
14. References . . . . . . . . . . . . . . . . . . . . . . . . . 24
14.1. Normative References . . . . . . . . . . . . . . . . . . 24
14.2. Informative References . . . . . . . . . . . . . . . . . 24
Appendix A. Additional features . . . . . . . . . . . . . . . . 25
A.1. Packet pacing . . . . . . . . . . . . . . . . . . . . . . 25
A.2. Stream prioritization . . . . . . . . . . . . . . . . . . 26
A.3. Q-bit semantics (source quench) . . . . . . . . . . . . . 28
A.4. Frame skipping . . . . . . . . . . . . . . . . . . . . . 29
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 30
1. Introduction
Congestion in the internet is a reality and applications that are
deployed in the internet must have congestion control schemes in
place not only for the robustness of the service that it provides but
also to ensure the function of the currently deployed internet. As
the interactive realtime communication imposes a great deal of
requirements on the transport, a robust, efficient rate adaptation
for all access types is considered as an important part of
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interactive realtime communications as the transmission channel
bandwidth may vary over time. Wireless access such as LTE, which is
an integral part of the current internet, increases the importance of
rate adaptation as the channel bandwidth of a default LTE bearer
[QoS-3GPP] can change considerably in a very short time frame. Thus
a rate adaptation solution for interactive realtime media, such as
WebRTC, must be both quick and be able to operate over a large span
in available channel bandwidth. This memo describes a solution,named
SCReAM, that is based on the self-clocking principle of TCP and uses
techniques similar to what is used in a new delay based rate
adaptation algorithm, LEDBAT [RFC6817]. Because neither TCP nor
LEDBAT was designed for interactive realtime media, a few extra
features are needed to make the concept work well within this
context. This memo describes these extra features.
1.1. Wireless (LTE) access properties
[I-D.draft-sarker-rmcat-cellular-eval-test-cases] introduces the
complications that can be observed in wireless environments.
Wireless access such as LTE can typically not guarantee a given
bandwidth, this is true especially for default bearers. The network
throughput may vary considerably for instance in cases where the
wireless terminal is moving around.
Unlike wireline bottlenecks with large statistical multiplexing it is
not possible to try to maintain a given bitrate when congestion is
detected with the hope that other flows will yield, this because
there are generally few other flows competing for the same
bottleneck. Each user gets its own variable throughput bottleneck,
where the throughput depends on factors like channel quality, network
load and historical throughput. The bottom line is, if the
throughput drops, the sender has no other option than to reduce the
bitrate. In addition, the grace time, i.e. allowed reaction time
from the time that the congestion is detected until a reaction in
terms of a rate reduction is effected, is generally very short, in
the order of one RTT (Round Trip Time).
2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC2119 [RFC2119]
3. Overview of SCReAM Algorithm
The core SCReAM algorithm has similarities to concepts like self-
clocking used in TFWC [TFWC] and follows packet conservation
principles. The packet conservation principle is described as an
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important key-factor behind the protection of networks from
congestion [FACK].
The packet conservation principle is realized by including an
indication of the highest received sequence number in the feedback,
see Section 5, from the receiver back to the sender, the sender keeps
a list of transmitted packets and their respective sizes. This
information is then used to determine how many bytes can be
transmitted. A congestion window puts an upper limit on how many
bytes can be in flight, i.e. transmitted but not yet acknowledged.
The congestion window is determined in a way similar to LEDBAT
[RFC6817]. This ensures that the e2e latency is kept low. The basic
functionality is quite simple, there are however a few steps to take
to make the concept work with conversational media. These will be
briefly described in sections Section 3.1 to Section 3.3.
The rate adaptation solution constitutes three parts- congestion
control, transmission scheduling and media rate adaptation. All
these three parts reside at the sender side. The receiver side
algorithm is very simple in comparison as it only generates
acknowledgements to received RTP packets.
3.1. Congestion Control
The congestion control sets an upper limit on how much data can be in
the network (bytes in flight); this limit is called CWND (congestion
window) and is used in the transmission scheduling.
The SCReAM congestion control method, uses LEDBAT [RFC6817] to
measure the OWD (one way delay). The SCReAM sender calculates the
congestion window based on the feedback from SCReAM receiver. The
congestion window is allowed to increase if the OWD is below a
predefined target, otherwise the congestion window decreases. The
delay target is typically set to 50-100ms. This ensures that the OWD
is kept low on the average. The reaction to loss events is similar
to that of loss based TCP, i.e. an instant reduction of CWND.
LEDBAT is designed with file transfers as main use case which means
that the algorithm must be modified somewhat to work with rate-
limited sources such as video. The modifications are
o Congestion window validation techniques. These are similar in
action as the method described in [I-D.ietf-tcpm-newcwv].
o Fast start for bitrate increase. It makes the video bitrate ramp-
up within 5 to 10 seconds. The behavior is similar to TCP
slowstart. The fast start is exited when congestion is detected.
The fast start state can be resumed if the congestion level is
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low, this to enable a reasonably quick rate increase in case link
throughput increases.
o Adaptive delay target. This helps the congestion control to
compete with FTP traffic to some degree.
3.2. Transmission Scheduling
Transmission scheduling limits the output of data, given by the
relation between the number of bytes in flight and the congestion
window similar to TCP. Packet pacing is used to mitigate issues with
coalescing that may cause increased jitter and/or packet loss in the
media traffic.
3.3. Media Rate Control
The media rate control serves to adjust the media bitrate to ramp up
quickly enough to get a fair share of the system resources when link
throughput increases.
The reaction to reduced throughput must be prompt in order to avoid
getting too much data queued up in the RTP packet queues. The media
bitrate is decreased if the RTP queue size exceeds a threshold.
In cases where the sender frame queues increase rapidly such as the
case of a RAT (Radio Access Type) handover it may be necessary to
implement additional actions, such as discarding of encoded video
frames or frame skipping in order to ensure that the RTP queues are
drained quickly. Frame skipping means that the frame rate is
temporarily reduced. Discarding of old video frames is a more
efficient way to reduce media latency than frame skipping but it
comes with a requirement to repair codec state, frame skipping is
thus to prefer as a first remedy. Frame skipping is described as an
optional to implement feature in this specification.
4. Detailed Description of SCReAM
4.1. SCReAM Sender
This section describes the sender side algorithm in more detail. It
is split between the network congestion control and the video rate
adaptation.
Figure 1 shows the functional overview of a SCReAM sender. The RTP
application interaction with congestion control is described in
[I-D.ietf-rmcat-app-interaction]. Here we use a more decomposed
version of the implementation model in the sense that the RTP packets
may be queued up in the sender, the transmission of these RTP packets
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is controlled by a transmission scheduler. A SCReAM sender
implements rate control and a queue for each media type or source,
where RTP packets containing encoded media frames are temporarily
stored for transmission, the figure shows the details for when two
video sources (a.k.a streams) are used.
---------------------------- -----------------------------
| Video encoder | | Video encoder |
---------------------------- -----------------------------
^ | ^ ^ | ^
(1)| (2)| (3)| (1)| (2)| (3)|
| RTP | | RTP |
| V | | V |
| ------------- | | ------------- |
----------- | |-- ----------- | |--
| Rate | (4) | Queue | | Rate | (4) | Queue |
| control |<----| | | control |<----| |
| | |RTP packets| | | |RTP packets|
----------- | | ----------- | |
------------- -------------
| |
--------------- --------------
(5)| |(5)
RTP RTP
| |
v v
-------------- ----------------
| Network | (8) | Transmission |
| congestion |<-------->| scheduler |
| control | | |
-------------- ----------------
^ |
| (7) |(6)
---------RTCP---------- RTP
| |
| v
-------------
| UDP |
| socket |
-------------
Figure 1: SCReAM sender functional view
Video frames are encoded and forwarded to the queue (2). The media
rate adaptation adapts to the size of the RTP queue and controls the
video bitrate (1). The RTP packets are picked from each queue based
on some defined priority order or simply in a round robin fashion
(5). A transmission scheduler takes care of the transmission of RTP
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packets, to be written to the UDP socket (6). In the general case
all media must go through the transmission scheduler and is allowed
to be transmitted if the number of bytes in flight is less than the
congestion window. Audio frames can however be allowed to be
transmitted immediately as audio is typically low bitrate and thus
contributes little to congestion, this is something that is left as
an implementation choice. RTCP packets are received (7) and the
information about bytes in flight and congestion window is exchanged
between the network congestion control and the transmission scheduler
(8).
4.1.1. Constants and Parameter values
A set of constants are defined in Table 1, state variables are
defined in Table 2. And finally, local variables are described in
Table 3.
An init value [] indicates an empty array.
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+-------------------------------+------------------------+----------+
| Constant | Explanation | Value |
+-------------------------------+------------------------+----------+
| OWD_TARGET_LO | Min OWD target | 0.1s |
| OWD_TARGET_HI | Max OWD target | 0.4s |
| MAX_BYTES_IN_FLIGHT_HEAD_ROOM | Headroom for | 1.1 |
| | limitation of CWND | |
| GAIN | Gain factor for | 1.0 |
| | congestion window | |
| | adjustment | |
| BETA | CWND scale factor due | 0.6 |
| | to loss event | |
| BETA_R | Target rate scale | 0.8 |
| | factor due to loss | |
| | event | |
| BYTES_IN_FLIGHT_SLACK | Additional slack [%] | 10% |
| | to the congestion | |
| | window | |
| RATE_ADJUST_INTERVAL | Interval between video | 0.1s |
| | bitrate adjustments | |
| FRAME_PERIOD | Video coder frame | |
| | period [s] | |
| TARGET_BITRATE_MIN | Min target_bitrate | |
| | [bps] | |
| TARGET_BITRATE_MAX | Max target_bitrate | |
| | [bps] | |
| RAMP_UP_TIME | Timespan [s] from | 10s |
| | lowest to highest | |
| | bitrate | |
| PRE_CONGESTION_GUARD | Guard factor against | 0.0..0.2 |
| | early congestion | |
| | onset. A higher value | |
| | gives less jitter | |
| | possibly at the | |
| | expense of a lower | |
| | video bitrate. | |
| TX_QUEUE_SIZE_FACTOR | Guard factor against | 0.0..2.0 |
| | RTP queue buildup | |
+-------------------------------+------------------------+----------+
Table 1: Constants
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+-------------------------+--------------------+--------------------+
| Variable | Explanation | Init value |
+-------------------------+--------------------+--------------------+
| owd_target | OWD target | OWD_TARGET_LO |
| owd_fraction_avg | EWMA filtered | 0.0 |
| | owd_fraction | |
| owd_fraction_hist | Vector of the last | [] |
| | 20 owd_fraction | |
| owd_trend | OWD trend, | 0.0 |
| | indicates | |
| | incipient | |
| | congestion | |
| owd_trend_mem | Low pass filtered | 0.0 |
| | version of | |
| | owd_trend | |
| owd_norm_hist | Vector of the last | [] |
| | 100 owd_norm | |
| mss | Maximum segment | 1000 |
| | size = Max RTP | |
| | packet size [byte] | |
| min_cwnd | Minimum congestion | 2*MSS |
| | window [byte] | |
| in_fast_start | True if in fast | true |
| | start state | |
| cwnd | Congestion window | min_cwnd |
| | [byte] | |
| cwnd_i | Congestion window | 1 |
| | inflection point | |
| bytes_newly_acked | The number of | 0 |
| | bytes that was | |
| | acknowledged with | |
| | the last received | |
| | acknowledgement | |
| | i.e. bytes | |
| | acknowledged since | |
| | the last CWND | |
| | update [byte]. | |
| | Reset after a CWND | |
| | update | |
| send_wnd | Upper limit of how | 0 |
| | many bytes that | |
| | can be transmitted | |
| | [byte]. Updated | |
| | when CWND is | |
| | updated and when | |
| | RTP packet is | |
| | transmitted | |
| t_pace | Approximate | 0.001 |
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| | estimate of inter- | |
| | packet | |
| | transmission | |
| | interval [s], | |
| | updated when RTP | |
| | packet transmitted | |
| age_vec | A vector of the | [] |
| | last 20 RTP packet | |
| | queue delay | |
| | samples | |
| frame_skip_intensity | Indicates the | 0.0 |
| | intensity of the | |
| | frame skips | |
| since_last_frame_skip | Number of video | 0 |
| | frames since the | |
| | last skip | |
| consecutive_frame_skips | Number of | 0 |
| | consecutive frame | |
| | skips | |
| target_bitrate | Video target | TARGET_BITRATE_MIN |
| | bitrate [bps] | |
| target_bitrate_i | Video target | 1 |
| | bitrate inflection | |
| | point i.e. the | |
| | last known highest | |
| | target_bitrate | |
| | during fast start. | |
| | Used to limit | |
| | bitrate increase | |
| | close to the last | |
| | know congestion | |
| | point | |
| rate_transmit | Measured transmit | 0.0 |
| | bitrate [bps] | |
| rate_acked | Measured | 0.0 |
| | throughput based | |
| | on received | |
| | acknowledgements | |
| | [bps] | |
| rate_rtp | Measured bitrate | 0.0 |
| | from the media | |
| | encoder [bps] | |
| rate_rtp_median | Median value of | 0.0 |
| | rate_rtp, computed | |
| | over more than 10s | |
| | [bps] | |
| s_rtt | Smoothed RTT [s], | 0.0 |
| | computed similar | |
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| | to method depicted | |
| | in [RFC6298] | |
| rtp_queue_size | Size of RTP | 0 |
| | packets in queue | |
| | [bits] | |
| rtp_size | Size of the last | 0 |
| | transmitted RTP | |
| | packets [byte] | |
| frame_skip | Skip encoding of | false |
| | video frame if | |
| | true | |
+-------------------------+--------------------+--------------------+
Table 2: State variables
+------------------+------------------------------------------------+
| Variable | Explanation |
+------------------+------------------------------------------------+
| owd | OWD = One way delay with base delay subtracted |
| | [s]. This is an estimate of the network |
| | queueing delay. |
| owd_fraction | OWD as a fraction of the OWD target |
| owd_norm | OWD normalized to OWD_TARGET_LO |
| owd_norm_mean | Average OWD norm over the last 100 samples |
| owd_norm_mean_sh | Average OWD norm over the last 20 samples |
| owd_norm_var | OWD norm variance over the last 100 samples |
| off_target | Relation between OWD and OWD target |
| scl_i | A general scalefactor that is applied to the |
| | CWND or target_bitrate increase |
| x_cwnd | Additional increase of CWND, used when |
| | send_wnd is computed |
| pace_bitrate | The allowed RTP packet transmission rate, used |
| | in the computation of t_pace [bps] |
| age_avg | Average RTP queue delay [s] |
| increment | Allowed target_bitrate increase |
| current_rate | Max of rate_transmit and rate_acked |
+------------------+------------------------------------------------+
Table 3: Local temporary variables
4.1.2. Network congestion control
This section explains the network congestion control, it contains two
main functions
o Computation of congestion window at the sender: Gives an upper
limit to the number of bytes in flight i.e. how many bytes that
have been transmitted but not yet acknowledged.
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o Transmission scheduling at the sender: RTP packets are transmitted
if allowed by the relation between the number of bytes in flight
and the congestion window. This is controlled by the send window.
Unlike TCP, SCReAM is not a byte oriented protocol, rather it is an
RTP packet oriented protocol. Thus it keeps a list of transmitted
RTP packets and their respective sending times (wall-clock time).
The feedback indicates the highest received RTP sequence number and a
timestamp (wall-clock time) when it was received. In addition, an
ACK list is included to make it possible to determine lost packets.
4.1.2.1. Congestion window update
The congestion window is computed from the one way (extra) delay
estimates (OWD) that are obtained from the send and received
timestamp of the RTP packets. LEDBAT [RFC6817] explains the details
of the computation of the OWD. An OWD sample is obtained for each
received acknowledgement. No smoothing of the OWD samples occur,
however some smoothing occurs anyway as the computation of the CWND
is in itself a low pass filter function.
SCReAM uses the terminology "Bytes in flight (bytes_in_flight)" which
is computed as the sum of the sizes of the RTP packets ranging from
the RTP packet most recently transmitted down to but not including
the acknowledged packet with the highest sequence number. As an
example: If RTP packet was sequence number SN with transmitted and
the last ACK indicated SN-5 as the highest received sequence number
then bytes in flight is computed as the sum of the size of RTP
packets with sequence number SN-4, SN-3, SN-2, SN-1 and SN.
CWND is updated differently depending on whether the congestion
control is in fast start or not and if a loss event is detected. A
Boolean variable in_fast_start indicates if the congestion is in fast
start state.
A loss event indicates one or more lost RTP packets within an RTT.
This is detected by means of inspection for holes in the sequence
number space in the acknowledgements with some margin for possible
packet reordering in the network. As an alternative, a timer for
loss detection similar to TCP RACK may be used.
Below is described the actions when an acknowledgement from the
receiver is received.
bytes_newly_acked is updated.
The OWD fraction and an average of it are computed as
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owd_fraction = owd/owd_target
owd_fraction_avg = 0.9* owd_fraction_avg + 0.1* owd_fraction
The OWD fraction is sampled every 50ms and the last 20 samples are
stored in a vector (owd_fraction_hist). This vector is used in the
computation of an OWD trend that gives a value between 0.0 and 1.0
depending on how close to congestion it is. The OWD trend is
calculated as follows
Let R(owd_fraction_hist,K) be the autocorrelation function of
owd_fraction_hist at lag K. The 1st order prediction coefficient is
formulated as
a = R(owd_fraction_hist,1)/R(owd_fraction_hist,0)
The prediction coefficient a has positive values if OWD shows an
increasing trend, thus an indication of congestion is obtained before
the OWD target is reached. The prediction coefficient is further
multiplied with owd_fraction_avg to reduce sensitivity to increasing
OWD when OWD is very small. The OWD trend is thus computed as
owd_trend = max(0.0,min(1.0,a*owd_fraction_avg))
owd_trend_mem = max(0.99*owd_trend_mem, owd_trend)
The owd_trend is utilized in the media rate control and to determine
when to exit slow start. owd_trend_mem is used to enforce a less
aggressive rate increase after congestion events.
An off target value is computed as
off_target = (owd_target - owd) / owd_target
A temporal variable is scl_i is computed as
scl_i = max(0.2, min(1.0, (abs(cwnd-cwnd_i)/cwnd_i*4)^2))
scl_i is used to limit the CWND increase when close to the last known
max value, before congestion was last detected.
The congestion window update depends on whether a loss event has
occurred, and if the congestion control is if fast start or not.
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____________________________________________________________________
On loss event:
If a loss event is detected then in_fast_start is set to false and
CWND is updated according to
cwnd_i = cwnd
cwnd = max(min_cwnd,cwnd*BETA)
otherwise the CWND update continues
____________________________________________________________________
in_fast_start = true:
in_fast_start is set to false and cwnd_i=cwnd if owd_trend >= 0.2 and
otherwise CWND is updated according to
cwnd = cwnd + bytes_newly_acked*scl_i
____________________________________________________________________
in_fast_start = false:
Values of off_target > 0.0 indicates that the congestion window can
be increased. This is done according to the equations below.
gain = GAIN*(1.0 + max(0.0, 1.0 - owd_trend/ 0.2))
The equation above limits the gain when near congestion is detected
gain *= scl_i
This equation limits the gain when CWND is close to its last known
max value
cwnd += gain * off_target * bytes_newly_acked * mss / cwnd
Values of off_target <= 0.0 indicates congestion, CWND is then
updated according to the equation
cwnd += GAIN*off_target*bytes_newly_acked*mss/cwnd
The equations above are very similar to what is specified in
[RFC6817]. There are however a few differences.
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o [RFC6817] specifies a constant GAIN, this specification however
limits the gain when CWND is increased dependent on near
congestion state and the relation to the last known max CWND
value.
o [RFC6817] specifies that the CWND increased is limited by an
additional function controlled by a constant ALLOWED_INCREASE.
This additional limitation is removed in this specification.
____________________________________________________________________
A number of final steps in the congestion window update procedure are
outlined below
____________________________________________________________________
Resume fast start:
Fast start can be resumed in order to speed up the bitrate increase
in case congestion abates. The condition to resume fast start
(in_fast_start = true) is that owd_trend is less than 0.2 for 1.0
seconds or more.
____________________________________________________________________
Competing flows compensation, adjustment of owd_target:
Competing flows compensation is needed to avoid that flows congestion
controlled by SCReAM are starved out by flows that are more
aggressive in their nature. The owd_target is adjusted according to
the owd_norm_mean_sh whenever owd_norm_var is below a given value.
The condition to update owd_target is fulfilled if owd_norm_var <
0.16 (indicating that the standard deviation is less than 0.4).
owd_target is then update as:
owd_target = min(OWD_TARGET_HI,max(OWD_TARGET_LO, owd_norm_mean_sh*
OWD_TARGET_LO*1.1))
____________________________________________________________________
Final CWND adjustment step:
The congestion window is limited by the maximum number of bytes in
flight over the last 1.0 seconds according to
cwnd = min(cwnd, max_bytes_in_flight*MAX_BYTES_IN_FLIGHT_HEAD_ROOM)
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This avoids possible over-estimation of the throughput after for
example, idle periods.
Finally cwnd is set to ensure that it is at least min_cwnd
cwnd = max(cwnd, MIN_CWND)
4.1.2.2. Transmission scheduling
The principle is to allow packet transmission of an RTP packet only
if the number of bytes in flight is less than the congestion window.
There are however two reasons why this strict rule will not work
optimally:
o Bitrate variations: The video frame size is always varying to a
larger or smaller extent, a strict rule as the one given above
will have the effect that the video bitrate have difficulties to
increase as the congestion window puts a too hard restriction on
the video frame size variation, this further can lead to
occasional queuing of RTP packets in the RTP packet queue that
will prevent bitrate increase because of the increased RTP queue
size.
o Reverse (feedback) path congestion: Especially in transport over
buffer-bloated networks, the one way delay in the reverse
direction may jump due to congestion. The effect of this is that
the acknowledgements are delayed with the result that the self-
clocking is temporarily halted, even though the forward path is
not congested.
Packets are transmitted at a pace given by the send window, computed
below
The send window is computed differently depending on OWD and its
relation to the OWD target.
o If owd > owd_target:
The send window is computed as
send_wnd = cwnd-bytes_in_flight
This enforces a strict rule that helps to prevent further queue
buildup.
o If owd <= owd_target:
A helper variable
x_cwnd=1.0+BYTES_IN_FLIGHT_SLACK*max(0.0,
min(1.0,1.0-owd_trend/0.5))/100.0
is computed. The send window is computed as
send_wnd = max(cwnd*x_cwnd, cwnd+mss)-bytes_in_flight
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This gives a slack that reduces as congestion increases,
BYTES_IN_FLIGHT_SLACK is a maximum allowed slack in percent. A
large value increases the robustness to bitrate variations in the
source and congested feedback channel issues. The possible
drawback is increased delay or packet loss when forward path
congestion occur.
4.1.3. Video rate control
The video rate control is operated based on the size of the RTP
packet send queue and observed loss events. In addition, owd_trend
is also considered in the rate control, this to reduce the amount of
induced network jitter.
A variable target_bitrate is adjusted depending on the congestion
state. The target bitrate can vary between a minimum value
(target_bitrate_min) and a maximum value (target_bitrate_max).
For the overall bitrate adjustment, two network throughput estimates
are computed :
o rate_transmit: The measured transmit bitrate
o rate_acked: The ACKed bitrate, i.e. the volume of ACKed bits per
time unit.
Both estimates are updated every 200ms.
The current throughput current_rate is computed as the maximum value
of rate_transmit and rate_acked. The rationale behind the use of
rate_acked in addition to rate_transmit is that rate_transmit is
affected also by the amount of data that is available to transmit,
thus a lack of data to transmit can be seen as reduced throughput
that may itself cause an unnecessary rate reduction. To overcome
this shortcoming; rate_acked is used as well. This gives a more
stable throughput estimate.
The bitrate is updated at regular intervals, given by
RATE_ADJUST_INTERVAL and differently depending the fast start state
The rate change behavior depends on whether a loss event has
occurred, and if the congestion control is if fast start or not.
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____________________________________________________________________
On loss event:
First of all the target_bitrate is updated if a new loss event was
indicated and the rate change procedure is exited.
target_bitrate_i = target_bitrate
target_bitrate = max(BETA_R* target_bitrate, TARGET_BITRATE_MIN)
If no loss event was indicated then the rate change procedure
continues.
____________________________________________________________________
in_fast_start = true:
An allowed increment is computed based on the congestion level and
the relation to target_bitrate_i
scl_i = (target_bitrate - target_bitrate_i)/ target_bitrate_i
increment = TARGET_BITRATE_MAX* RATE_ADJUST_INTERVAL/RAMP_UP_TIME*
(1.0- min(1.0, owd_trend/0.1))
increment *= max(0.2, min(1.0, (scl_i*4)^2))
target_bitrate += increment
target_bitrate is reduced further if congestion is detected.
target_bitrate *= (1.0- PRE_CONGESTION_GUARD*owd_trend)
____________________________________________________________________
in_fast_start = false:
target_bitrate_i is updated to the current value of target_bitrate if
in_fast_start was true the last time the bitrate was updated.
A pre-congestion indicator is computed as
pre_congestion = min(1.0, max(0.0, owd_fraction_avg-0.3)/0.7)
pre_congestion += owd_trend
The target bitrate is computed as
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target_bitrate=current_rate*(1.0-
PRE_CONGESTION_GUARD*pre_congestion)-TX_QUEUE_SIZE_FACTOR
*rtp_queue_size
____________________________________________________________________
Final step:
As a final step, the target bitrate is limited such that it is kept
within reasonable bounds.
In cases where input stimuli to the media encoder is static, for
instance in "talking head" scenarios, the target bitrate is not
always fully utilized. This may cause undesirable oscillations in
the target bitrate in the cases where the link throughput is limited
and the media coder input stimuli changes between static and varying.
To overcome this issue, the target bitrate is capped to be less than
a given multiplier of a median value of the history of media coder
output bitrates. A rate_rtp_limit is computed as
rate_rtp_limit = max(br, max(rate_rtp,rtp_rate_median))
A multiplier is applied to rate_rtp_limit, depending on congestion
history
rate_rtp_limit *= (2.0-1.0*owd_trend_mem)
The target_bitrate is then limited by rate_rtp_limit
target_bitrate = min(target_bitrate, rate_rtp_limit)
Finally the target_bitrate is enforced to be within the defined min
and max values
target_bitrate =
min(TARGET_BITRATE_MAX,max(TARGET_BITRATE_MIN,target_bitrate))
4.2. SCReAM Receiver
The SCReAM receiver is very simple in its implementation. The task
is to feedback acknowledgements of received packets. For that
purpose a set of state variables are needed, these are explained in
Table 4.
One set of state variables are maintained per stream.
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+-----------------------------+-----------------------------+-------+
| Variable | Explanation | Init |
| | | value |
+-----------------------------+-----------------------------+-------+
| rx_timestamp | The wall clock timestamp | 0 |
| | when the latest RTP packet | |
| | was received | |
| highest_rtp_sequence_number | The highest received | 0 |
| | sequence number | |
| ack_vector | A 16 bit vector that | 0 |
| | indicates received RTP | |
| | packets with a sequence | |
| | number lower than | |
| | highest_rtp_sequence_number | |
| n_loss | An 8 bit counter for the | 0 |
| | number of lost RTP packets, | |
| | separate counters are | |
| | maintained for each SSRC | |
| n_ECN | An 8 bit counter for the | 0 |
| | number of ECN-CE marked RTP | |
| | packets, separate counters | |
| | are maintained for each | |
| | SSRC | |
| pending_feedback | Indicates that an RTP | false |
| | packet was received and | |
| | that an RTCP packet can be | |
| | generated when RTCP timing | |
| | rules permit | |
| last_transmit_t | Last time an RTCP packet | -1.0 |
| | was transmitted, this is | |
| | used to ensure that RTCP | |
| | feedback is generated | |
| | fairly for all streams. | |
+-----------------------------+-----------------------------+-------+
Table 4: State variables
Upon reception of an RTP packet, the state variables in Table 4
should be updated and the RTCP processing function should be
notified. An RTCP packet is later generated based on the state
variables, how often this is done depends on the RTCP bandwidth.
5. Feedback Message
The feedback is over RTCP [RFC3550] and is based on [RFC4585]. It is
implemented as a transport layer feedback message (RTPFB), see
proposed example in Figure 2. The feedback control information part
(FCI) consists of the following elements.
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o Highest received RTP sequence number: The highest received RTP
sequence number for the given SSRC
o n_lost: Ackumulated number of lost RTP packets for the given SSRC
o Timestamp: A timestamp value indicating when the last packet was
received which makes it possible to compute the one way (extra)
delay (OWD).
o n_ECN: Ackumulated number of ECN-CE marked RTP packets for the
given SSRC
o Source quench bit (Q): Makes it possible to request the sender to
reduce its congestion window. This is useful if WebRTC media is
received from many hosts and it becomes necessary to balance the
bitrates between the streams.
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 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Highest recv. seq. nr. (16b) | n_lost | n_ECN |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Timestamp (32bits) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|Q| Reserved for future use |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Figure 2: Transport layer feedback message
To make the feedback as frequent as possible, the feedback packets
are transmitted as reduced size RTCP according to [RFC5506].
The timestamp clock time is recommended to be set to a fixed value
such as 1000Hz, defined in this specification. The n_lost and n_ECN
makes it possible to take necessary actions on the detection of lost
and ECN marked packets.
Section 4 describes the main algorithm details and how the feedback
is used.
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6. Discussion
This section covers a few open discussion points
o RTCP feedback overhead: SCReAM benefits from a relatively frequent
feedback. Experiments have shown that a feedback rate roughly
equal to the frame rate gives a stable self-clocking and
robustness against loss of feedback. With a maximum bitrate of
1500kbps the RTCP feedback overhead is in the range 10-15kbps with
reduced size RTCP, including IP and UDP framing, in other words
the RTCP overhead is quite modest and should not pose a problem in
the general case. Other solutions may be required in highly
asymmetrical link capacity cases. Worth notice is that SCReAM can
work with as low feedback rates as once every 200ms, this however
comes with a higher sensitivity to loss of feedback and also a
potential reduction in throughput.
o AVPF mode: The RTCP feedback is based on AVPF regular mode. The
SCReAM feedback is transmitted as reduced size RTCP so save
overhead, it is however required to transmit full compound RTCP at
regular intervals, this interval can be controlled by trr-int
depicted in [RFC4585].
o BETA, CWND scale factor due to loss: The BETA value is recommended
to be higher than 0.5. The reason behind this is that congestion
control for multimedia has to deal with a source that is rate
limited. A file transfer has "unlimited" source bitrate in
comparison. The outcome is that SCReAM must be a little more
aggressive than a file transfer in order to not be out competed.
7. Conclusion
This memo describes a congestion control algorithm for RMCAT that it
is particularly good at handling the quickly changing condition in
wireless network such as LTE. The solution conforms to the packet
conservation principle and leverages on novel congestion control
algorithms and recent TCP research, together with media bitrate
determined by sender queuing delay and given delay thresholds. The
solution has shown potential to meet the goals of high link
utilization and prompt reaction to congestion. The solution is
realized with a new RFC4585 transport layer feedback message.
8. Open issues
A list of open issues.
o Describe how clock drift compensation is done
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o Describe how FEC overhead is accounted for in target_bitrate
computation
o Investigate the impact of more sparse RTCP feedback, for instance
once per RTT
o Describe ECN behavior
9. Source code
Source code for SCReAM is available in two formats :
o C++ code, that is apt for experimentation. The code maitained as
Visual Studio project. This code can possibly be included in
simulators such as NS3. Avaliable at
https://github.com/EricssonResearch/scream
o OpenWebRTC implementation : Work in progress, see
http://www.openwebrtc.io/ for information about the OpenWebRTC
project
10. Acknowledgements
We would like to thank the following persons for their comments,
questions and support during the work that led to this memo: Markus
Andersson, Bo Burman, Tomas Frankkila, Frederic Gabin, Laurits Hamm,
Hans Hannu, Nikolas Hermanns, Stefan Haakansson, Erlendur Karlsson,
Daniel Lindstroem, Mats Nordberg, Jonathan Samuelsson, Rickard
Sjoeberg, Robert Swain, Magnus Westerlund, Stefan Aalund.
11. IANA Considerations
A new RFC4585 transport layer feedback message needs to be
standardized.
12. Security Considerations
The feedback can be vulnerable to attacks similar to those that can
affect TCP. It is therefore recommended that the RTCP feedback is at
least integrity protected.
13. Change history
A list of changes:
o -05 to WG-00 : First version of WG doc, moved additional features
section to Appendix. Added description of prioritization in
SCReAM. Added description of additional cap on target bitrate
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o -04 to -05 : ACK vector is replaced by a loss counter, PT is
removed from feedback, references to source code added
o -03 to -04 : Extensive changes due to review comments, code
somewhat modified, frame skipping made optional
o -02 to -03 : Added algorithm description with equations, removed
pseudo code and simulation results
o -01 to -02 : Updated GCC simulation results
o -00 to -01 : Fixed a few bugs in example code
14. References
14.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July
2006.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, April 2009.
[RFC6298] Paxson, V., Allman, M., Chu, J., and M. Sargent,
"Computing TCP's Retransmission Timer", RFC 6298, June
2011.
[RFC6817] Shalunov, S., Hazel, G., Iyengar, J., and M. Kuehlewind,
"Low Extra Delay Background Transport (LEDBAT)", RFC 6817,
December 2012.
14.2. Informative References
[FACK] "Forward Acknowledgement: Refining TCP Congestion
Control", 2006.
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[I-D.draft-sarker-rmcat-cellular-eval-test-cases]
Sarker, Z., "Evaluation Test Cases for Interactive Real-
Time Media over Cellular Networks",
<http://www.ietf.org/id/
draft-sarker-rmcat-cellular-eval-test-cases-00.txt>.
[I-D.ietf-rmcat-app-interaction]
Zanaty, M., Singh, V., Nandakumar, S., and Z. Sarker, "RTP
Application Interaction with Congestion Control", draft-
ietf-rmcat-app-interaction-01 (work in progress), October
2014.
[I-D.ietf-tcpm-newcwv]
Fairhurst, G., Sathiaseelan, A., and R. Secchi, "Updating
TCP to support Rate-Limited Traffic", draft-ietf-tcpm-
newcwv-10 (work in progress), April 2015.
[QoS-3GPP]
TS 23.203, 3GPP., "Policy and charging control
architecture", June 2011, <http://www.3gpp.org/ftp/specs/
archive/23_series/23.203/23203-990.zip>.
[TFWC] University College London, "Fairer TCP-Friendly Congestion
Control Protocol for Multimedia Streaming", December 2007,
<http://www-dept.cs.ucl.ac.uk/staff/M.Handley/papers/
tfwc-conext.pdf>.
Appendix A. Additional features
This section describes additional features. They are not required
for the basic functionality of SCReAM but can improve performance in
certain scenarios and topologies.
A.1. Packet pacing
Packet pacing is used in order to mitigate coalescing i.e. that
packets are transmitted in bursts.
Packet pacing is enforced when owd_fraction_avg is greater than 0.1.
The time interval between consecutive packet transmissions is then
enforced to equal or higher than t_pace where t_pace is given by the
equations below.
pace_bitrate = max (50000, cwnd* 8 / s_rtt)
t_pace = rtp_size * 8 / pace_bitrate
rtp_size is the size of the last transmitted RTP packet
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A.2. Stream prioritization
As mentioned in Section 4, the prioritization between several streams
can be managed in many different ways. The most simple way is to
pick RTP packets from the RTP queues in a round-robin fashion. For
more advanced scheduling, more advanced algorithms are required.
Below is described the algorithm that is implemented in the SCReAM
code Section 9.
Suppose that we have two video streams, where stream 1 has priority
1.0 and stream 2 has priority 0.5. Each stream starts with a credit
of 0 bytes, credit is given to streams that are not given permission
to transmit at a given scheduling instant, the credit is considered
in later transmission instants.
The steps below outline how transmission and scheduling of the two
RTP streams can evolve. For simplicily it is assumed that the stream
RTP queues contain 1200 byte packets.
1. SCReAMs send window allows transmission of 1200 bytes.
* The stream with the highest priority is picked, in this case
it is stream 1. Stream 1 thus transmit 1200 bytes.
* Stream 2 gets its credit increased by 1200*0.5/1.0 = 600 byte
and thus has a credit of 600 bytes.
2. SCReAMs send window allows transmission of another 1200 bytes.
* Stream 2 has too little credit (600 bytes) to transmit a 1200
byte packet.
* Stream 1 is therefore picked again as it has the highest
priority and thus gets to transmit yet another 1200 byte
packet.
* Stream 2 gets its credit increased by 1200*0.5/1.0 = 600 byte
and thus has a credit of 1200 bytes.
3. SCReAMs send window allows transmission of another 1200bytes.
* Stream 2 now has enough credit (1200 bytes) to transmit a 1200
byte packet.
* Stream 2 thus transmits a 1200 byte packet and in the process
gets its credit reduced by 1200 byte and is then down to a
credit of 0.
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* Stream 1 gets its credit increased by 1200*1.0/0.5 = 2400 byte
and thus has a credit of 2400 bytes.
4. SCReAMs send window allows transmission of another 1200 bytes.
1. Stream 1 now has the highest credit (2400bytes).
2. Stream 1 thus transmits a 1200 byte packet and in the process
gets its credit reduced by 1200 byte and is then down to a
credit of 1200 bytes.
3. Stream 2 gets its credit increased by 1200*0.5/1.0 = 600 byte
and thus has a credit of 600 bytes.
5. SCReAMs send window allows transmission of another 1200 bytes.
1. Stream 1 still has the highest credit (1200 bytes).
2. Stream 1 thus transmits a 1200 byte packet and in the process
gets its credit reduced by 1200 byte and is then down to a
credit of 0.
3. Stream 2 gets its credit increased by 1200*0.5/1.0 = 600 byte
and thus has a credit of 1200bytes.
6. SCReAMs send window allows transmission of another 1200 bytes.
1. Stream 2 now has the highest credit (1200 bytes).
2. Stream 2 thus transmits a 1200 byte packet and in the process
gets its credit reduced by 1200 byte and is then down to a
credit of 0.
3. Stream 1 gets its credit increased by 1200*1.0/0.5 = 2400
byte and thus has a credit of 2400 bytes.
The procedure above repeats it self. In the above example it is
quite easy to see that stream 1 gets to transmit 2 RTP packets for
every 1 RTP packets that stream 2 gets to transmit. The very detais
of the algoritm is found in the C++ code (see Section 9) in the
module ScreamTx and the functions getPrioritizedStream(..),
addCredit(..) and subtractCredit(..).
The above functionality works relatively well and operates with at
the same speed as RTP packet transmission. There are however cases
where rate limited video or very large IR frames makes the
prioritization less efficient. The adjustPriorities(..) function in
ScreamTx solves this issue on a longer time scale by means of an
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additional compensation for deviations in the measured transmit
bitrate of the individual streams.
Prioritization mechanisms of sources that may be highly variant is a
relatively complicated task to achieve. The above outlined algorithm
manages it to some degree but it is quite likely that the algorithm
needs to be refined further.
A.3. Q-bit semantics (source quench)
The Q bit in the feedback is set by a receiver to signal that the
sender should reduce the bitrate. The sender will in response to
this reduce the congestion window with the consequence that the video
bitrate decreases. A typical use case for source quench is when a
receiver receives streams from sources located at different hosts and
they all share a common bottleneck, typically it is difficult to
apply any rate distribution signaling between the sending hosts. The
solution is then that the receiver sets the Q bit in the feedback to
the sender that should reduce its rate, if the streams share a common
bottleneck then the released bandwidth due to the reduction of the
congestion window for the flow that had the Q bit set in the feedback
will be grabbed by the other flows that did not have the Q bit set.
This is ensured by the opportunistic behavior of SCReAM's congestion
control. The source quench will have no or little effect if the
flows do not share the same bottleneck.
The reduction in congestion window is proportional to the amount of
SCReAM RTCP feedback with the Q bit set, the below steps outline how
the sender should react to RTCP feedback with the Q bit set. The
reduction is done once per RTT. Let :
o n = Number of received RTCP feedback messages in one RTT
o n_q = Number of received RTCP feedback messages in one RTT, with Q
bit set.
The new congestion window is then expressed as:
cwnd = max(MIN_CWND, cwnd*(1.0-0.5* n_q /n))
Note that CWND is adjusted at most once per RTT. Furthermore The
CWND increase should be inhibited for one RTT if CWND has been
decreased as a result of Q bits set in the feedback.
The required intensity of the Q-bit set in the feedback in order to
achieve a given rate distribution depends on many factors such as
RTT, video source material etc. The receiver thus need to monitor
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the change in the received video bitrate on the different streams and
adjust the intensity of the Q-bit accordingly.
A.4. Frame skipping
Frame skipping is a feature that makes it possible to reduce the size
of the RTP queue in the cases that e.g. the channel throughput drops
dramatically or even goes below the lowest possible video coder rate.
Frame skipping is optional to implement as it can sometimes be
difficult to realize e.g. due to lack of API function to support
this.
Frame skipping is controlled by a flag frame_skip which, if set to 1
dictates that the video coder should skip the next video frame. The
frame skipping intensity at the current time instant is computed
according to the steps below
The queuing delay is sampled every frame period and the last 20
samples are stored in a vector age_vec
An average queuing delay is computed as a weighted sum over the
samples in age_vec. age_avg at the current time instant is computed
as
age_avg(n) = SUM age_vec(n-k)*w(k) k = [0..20[
w(n) are weight factors arranged to give the most recent samples a
higher weight.
The change in age_avg is computed as
age_d = age_avg(n) - age_avg(n-1)
The frame skipping intensity at the current time instant n is
computed as
o If age_d > 0 and age_avg > 2*FRAME_PERIOD:
frame_skip_intensity = min(1.0, (age_vec(n)-2*FRAME_PERIOD)/(4*
FRAME_PERIOD)
o Otherwise frame skip intensity is set to zero
The skip_frame flag is set depending on three variables
o frame_skip_intensity
o since_last_frame_skip, i.e the number of consecutive frames
without frame skipping
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o consecutive_frame_skips, i.e the number of consecutive frame skips
The flag skip_frame is set to 1 if any of the conditions below is
met, otherwise it is set to 0.
o age_vec(n) > 0.2 && consecutive_frame_skips < 5
o frame_skip_intensity < 0.5 && since_last_frame_skip >= 1.0/
frame_skip_intensity
o frame_skip_intensity >= 0.5 && consecutive_frame_skips <
(frame_skip_intensity -0.5)*10
The arrangement makes sure that no more than 4 frames are skipped in
sequence, the rationale is to ensure that the input to the video
encoder does not change to much, something that may give poor
prediction gain.
Authors' Addresses
Ingemar Johansson
Ericsson AB
Laboratoriegraend 11
Luleaa 977 53
Sweden
Phone: +46 730783289
Email: ingemar.s.johansson@ericsson.com
Zaheduzzaman Sarker
Ericsson AB
Laboratoriegraend 11
Luleaa 977 53
Sweden
Phone: +46 761153743
Email: zaheduzzaman.sarker@ericsson.com
Johansson & Sarker Expires November 3, 2015 [Page 30]