--- 1/draft-ietf-rmcat-scream-cc-01.txt 2015-10-19 00:15:16.455162805 -0700 +++ 2/draft-ietf-rmcat-scream-cc-02.txt 2015-10-19 00:15:16.511164150 -0700 @@ -1,1028 +1,977 @@ RMCAT WG I. Johansson Internet-Draft Z. Sarker Intended status: Experimental Ericsson AB -Expires: January 7, 2016 July 6, 2015 +Expires: April 21, 2016 October 19, 2015 Self-Clocked Rate Adaptation for Multimedia - draft-ietf-rmcat-scream-cc-01 + draft-ietf-rmcat-scream-cc-02 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. + media services such as video. 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 January 7, 2016. + This Internet-Draft will expire on April 21, 2016. 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 the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License. Table of Contents - 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 + 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 1.1. Wireless (LTE) access properties . . . . . . . . . . . . 3 - 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3 + 1.2. Why is it a self-clocked algorithm? . . . . . . . . . . . 3 + 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4 3. Overview of SCReAM Algorithm . . . . . . . . . . . . . . . . 4 - 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 + 3.1. Network Congestion Control . . . . . . . . . . . . . . . 7 + 3.2. Sender Transmission Control . . . . . . . . . . . . . . . 7 + 3.3. Media Rate Control . . . . . . . . . . . . . . . . . . . 7 + 4. Detailed Description of SCReAM . . . . . . . . . . . . . . . 8 + 4.1. SCReAM Sender . . . . . . . . . . . . . . . . . . . . . . 8 + 4.1.1. Constants and Parameter values . . . . . . . . . . . 8 + 4.1.1.1. Constants . . . . . . . . . . . . . . . . . . . . 8 + 4.1.1.2. State variables . . . . . . . . . . . . . . . . . 10 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. Implementation status . . . . . . . . . . . . . . . . . . . . 23 - 9.1. OpenWebRTC . . . . . . . . . . . . . . . . . . . . . . . 23 - 9.2. A C++ Implementation of SCReAM . . . . . . . . . . . . . 24 - 10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 24 - 11. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 25 - 12. Security Considerations . . . . . . . . . . . . . . . . . . . 25 - 13. Change history . . . . . . . . . . . . . . . . . . . . . . . 25 - 14. References . . . . . . . . . . . . . . . . . . . . . . . . . 25 - 14.1. Normative References . . . . . . . . . . . . . . . . . . 25 - 14.2. Informative References . . . . . . . . . . . . . . . . . 26 - Appendix A. Additional features . . . . . . . . . . . . . . . . 27 - A.1. Packet pacing . . . . . . . . . . . . . . . . . . . . . . 27 - A.2. Stream prioritization . . . . . . . . . . . . . . . . . . 28 - A.3. Q-bit semantics (source quench) . . . . . . . . . . . . . 30 - A.4. Frame skipping . . . . . . . . . . . . . . . . . . . . . 31 - Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 32 + 4.1.2.1. Updating bytes_newly_acked . . . . . . . . . . . 14 + 4.1.2.2. Updating congestion window . . . . . . . . . . . 14 + 4.1.2.3. Compensation for competing flows . . . . . . . . 16 + 4.1.2.4. Send window calculation . . . . . . . . . . . . . 17 + 4.1.2.5. Resuming fast increase . . . . . . . . . . . . . 18 + 4.1.3. Media rate control . . . . . . . . . . . . . . . . . 18 + 4.1.3.1. FEC and packet overhead considerations . . . . . 22 + 4.2. SCReAM Receiver . . . . . . . . . . . . . . . . . . . . . 22 + 5. Discussion . . . . . . . . . . . . . . . . . . . . . . . . . 22 + 6. Implementation status . . . . . . . . . . . . . . . . . . . . 23 + 6.1. OpenWebRTC . . . . . . . . . . . . . . . . . . . . . . . 23 + 6.2. A C++ Implementation of SCReAM . . . . . . . . . . . . . 24 + 7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 24 + 8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 25 + 9. Security Considerations . . . . . . . . . . . . . . . . . . . 25 + 10. Change history . . . . . . . . . . . . . . . . . . . . . . . 25 + 11. References . . . . . . . . . . . . . . . . . . . . . . . . . 25 + 11.1. Normative References . . . . . . . . . . . . . . . . . . 25 + 11.2. Informative References . . . . . . . . . . . . . . . . . 26 + Appendix A. Additional features . . . . . . . . . . . . . . . . 28 + A.1. Stream prioritization . . . . . . . . . . . . . . . . . . 28 + A.2. Computation of autocorrelation function . . . . . . . . . 28 + Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 29 1. Introduction - Congestion in the internet is a reality and applications that are - deployed in the internet must have congestion control schemes in + 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 + 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 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 + 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. + adaptation algorithm, LEDBAT [RFC6817]. 1.1. Wireless (LTE) access properties - [I-D.ietf-rmcat-wireless-tests] introduces the complications that can + [I-D.ietf-rmcat-wireless-tests] describes 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 + detected with the hope that other flows will yield, this is 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). +1.2. Why is it a self-clocked algorithm? + + Self-clocked congestion control algorithm provides with a benefit + over the rate based counterparts in that the former consists of two + parts; the congestion window computation that evolves over a longer + timescale (several RTTs) especially when the congestion window + evolution is dictated by estimated delay and; the fine grained + congestion control given by the self-clocking which operates on a + shorter time scale (1 RTT). + A rate based congestion control has only one mechanism to adjust the + sending rate and that makes it more problematic to reach the goal of + prompt reaction to congestion and also high throughput when channel + conditions are good. + 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 + The core SCReAM algorithm has similarities to the concepts of self- + clocking used in TFWC [TFWC] and follows the packet conservation + principle. The packet conservation principle is described as an important key-factor behind the protection of networks from - congestion [FACK]. + congestion [PACKET_CONSERVATION]. - 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. + In case of SCReAM, the receiver of the media sends the highest + received sequence number back to the sender, the sender keeps a list + of transmitted packets and their respective sizes. This information + is then used to determine the amount of bytes can be transmitted at + any given time instant. A congestion window puts an upper limit on + how many bytes can be in flight, i.e. transmitted but not yet + acknowledged. This is how the packet conservation principle is + realized. The congestion window is determined in a way similar to + LEDBAT [RFC6817]. - 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. + LEDBAT is a congestion control algorithm that uses send and receive + timestamps to estimate the queuing delay along the transmission path. + The use of LEDBAT 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. In a few + words they are: -3.1. Congestion Control + o Congestion window validation techniques. These are similar in + action as the method described in [I-D.ietf-tcpm-newcwv]. The + allowed idle period in this draft is shorter than in the + reference, this to avoid excessive delays in the cases where e.g. + wireless throughput has decreased during a period where the output + bitrate has been low. Furthermore, this draft allows for more + relaxed rules when the congestion window is allowed to grow, this + is necessary as the variable output bitrate generally means that + the congestion window is often under-utilized. + + o Fast increase for quicker bitrate increase. It makes the media + bitrate ramp-up within 5 to 10 seconds. The behavior is similar + to TCP slowstart. The fast increase is exited when congestion is + detected. The fast increase state can be however be resumed if + the congestion level is low, this to enable a reasonably quick + rate increase in case link throughput increases. + + o A delay trend is computed for earlier detection of incipient + congestion and as a result it reduces jitter. + + o Addition of media a rate control function. + + o Use of inflection points to calculate congestion window and media + rate to achieve reduced jitter. + + o Adjustment of delay target for better performance when competing + with other loss based congestion controlled flows + + The above mentioned features will be described in more detail in + sections Section 3.1 to Section 3.3. + + +---------------------------+ + | Media encoder | + +---------------------------+ + ^ | + (3)| (1)| + | RTP + | V + | +-----------+ + +---------+ | | + | Media | (2) | Queue | + | rate |<------| | + | control | |RTP packets| + +---------+ | | + +-----------+ + | + | + (4)| + RTP + | + v + +------------+ +--------------+ + | Network | (7) | Sender | + +-->| congestion |------>| Transmission | + | | control | | Control | + | +------------+ +--------------+ + | | + | (6) |(5) + |-------------RTCP----------| RTP + | | + | v + +------------+ + | UDP | + | socket | + +------------+ + + Figure 1: SCReAM sender functional view + + The SCReAM algorithm constitutes mainly of three parts: network + congestion control, sender transmission control and media rate + adaptation. All these three parts reside at the sender side. + Figure 1 shows the functional overview of a SCReAM sender. The + receiver side algorithm is very simple in comparison as it only + generates feedback containing acknowledgements to received RTP + packets, loss count and ECN [RFC6679] count. + +3.1. Network 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. + window) and is used in the sender transmission control. 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 + measure the one-way delay (OWD). The OWD can be expressed as the + estimated queuing delay. Similar to LEDBAT, it is not necessary to + use synchronized clocks in sender and receiver in order to compute + the one way delay. It is however necessary that they use the same + clock frequency, or that the clock frequency at the receiver can be + inferred reliably by the sender. The SCReAM sender calculates the + congestion window based on the feedback from the 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 - 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. + is kept low on the average. The reaction to loss events leads to an + instant reduction of CWND. Note that the source rate limited nature + of real time media such as video, typically means that the queuing + delay will mostly be below the given delay target, this is contrary + to the case where large files are transmitted using LEDBAT congestion + control, in which case the queuing delay will stay close to the delay + target. -3.2. Transmission Scheduling +3.2. Sender Transmission Control - Transmission scheduling limits the output of data, given by the + Sender Transmission Control 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 + window. Packet pacing is used to mitigate issues with ACK + compression 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. + getting too much data queued up in the RTP packet queues at the + sender. 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 + implement additional actions, such as discarding of encoded media 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. + temporarily reduced. Which method to use is a design consideration + and outside the scope of this algorithm description. 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 + is a split between the network congestion control and the media 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 - 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 + A SCReAM sender implements media rate control and a queue for each + media type or source, where RTP packets containing encoded media + frames are temporarily stored for transmission. Figure 1 shows the + details when single media sources (a.k.a streams) are used. However, + multiple media sources are also supported in the design, in that case + the sender transmission control will include a transmission + scheduler. The transmission scheduler can then enforce the + priorities for the different streams and act like a coupled + congestion controller for multiple flows. - 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 - 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). + Media frames are encoded and forwarded to the RTP queue (1). The + media rate adaptation adapts to the size of the RTP queue (2) and + controls the media bitrate (3). The RTP packets are picked from the + RTP queue (for multiple flows from each queue based on some defined + priority order or simply in a round robin fashion) (4) by the sender + transmission controller. The sender transmission controller (in case + of multiple flows a transmission scheduler) takes care of the + transmission of RTP packets, to be written to the UDP socket (5). In + the general case all media must go through the sender transmission + controller and is allowed to be transmitted if the number of bytes in + flight is less than the congestion window. RTCP packets are received + (6) and the information about bytes in flight and congestion window + is exchanged between the network congestion control and the sender + transmission control (7). 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. + Constants and state variables are listed in this section. - An init value [] indicates an empty array. +4.1.1.1. Constants - +-------------------------------+------------------------+----------+ - | 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 | | - +-------------------------------+------------------------+----------+ + The recommended values for the constants are deduced from + experimental results. - Table 1: Constants + OWD_TARGET_LO (0.1s) + Target value for the minimum OWD - +-------------------------+--------------------+--------------------+ - | 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 | - | | 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 | | - | | 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 | | - +-------------------------+--------------------+--------------------+ + OWD_TARGET_HI (0.4s) + Target value for the maximum OWD - Table 2: State variables + OWD_WEIGHT (0.1) + Averaging factor for owd_fraction_avg - +------------------+------------------------------------------------+ - | 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 | - +------------------+------------------------------------------------+ + MAX_BYTES_IN_FLIGHT_HEAD_ROOM (1.1) + Headroom for the limitation of CWND - Table 3: Local temporary variables + GAIN (1.0) + Gain factor for congestion window adjustment -4.1.2. Network congestion control + BETA_LOSS (0.6) + CWND scale factor due to loss event - This section explains the network congestion control, it contains two - main functions + BETA_ECN (0.8) + CWND scale factor due to ECN event - 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. + BETA_R (0.9) + Target rate scale factor due to loss event - 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. + MSS (1000 byte) + Maximum segment size = Max RTP packet size - 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. + BYTES_IN_FLIGHT_SLACK (10%) + Additional slack to the congestion window -4.1.2.1. Congestion window update + RATE_ADJUST_INTERVAL (0.2s) + Interval between media bitrate adjustments - 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. + TARGET_BITRATE_MIN + Min target bitrate [bps] - 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. + TARGET_BITRATE_MAX + Max target bitrate [bps] - 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. + RAMP_UP_SPEED (200kbps/s) + Maximum allowed rate increase speed - 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. + PRE_CONGESTION_GUARD (0.0..0.2) + Guard factor against early congestion onset. A higher value gives + less jitter, possibly at the expense of a lower link utilization. - Below is described the actions when an acknowledgement from the - receiver is received. + TX_QUEUE_SIZE_FACTOR (0.0..0.2) + Guard factor against RTP queue buildup - bytes_newly_acked is updated. + OWD_TREND_LO (0.2) Threshold value for owd_trend + T_RESUME_FAST_INCREASE Time span until fast increase can be resumed, + given that the owd_trend is below OWD_TREND_LO - The OWD fraction and an average of it are computed as - owd_fraction = owd/owd_target +4.1.1.2. State variables - owd_fraction_avg = 0.9* owd_fraction_avg + 0.1* owd_fraction + owd_target (OWD_TARGET_LO) + OWD target - 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 + owd_fraction_avg (0.0) + EWMA filtered owd_fraction - 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 + owd_fraction_hist[20] ({0,..,0}) + Vector of the last 20 owd_fraction - a = R(owd_fraction_hist,1)/R(owd_fraction_hist,0) + owd_trend (0.0) + OWD trend, indicates incipient congestion - 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_mem (0.0) + Low pass filtered version of owd_trend - owd_trend = max(0.0,min(1.0,a*owd_fraction_avg)) + owd_norm_hist[100] ({0,..,0}) + Vector of the last 100 owd_norm - owd_trend_mem = max(0.99*owd_trend_mem, owd_trend) + min_cwnd (2*MSS) + Minimum congestion window - 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. + in_fast_increase (true) + True if in fast increase state - An off target value is computed as + cwnd (min_cwnd) + Congestion window - off_target = (owd_target - owd) / owd_target + cwnd_last_max (1 byte) + Congestion window inflection point, i.e. the last known highest + cwnd. Used to limit cwnd increase close to the last known + congestion point. - A temporal variable is scl_i is computed as + bytes_newly_acked (0) + The number of bytes that was acknowledged with the last received + acknowledgement i.e. bytes acknowledged since the last CWND update. + Reset after a CWND update - scl_i = max(0.2, min(1.0, (abs(cwnd-cwnd_i)/cwnd_i*4)^2)) + send_wnd (0) + Upper limit of how many bytes that can be transmitted. Updated + when CWND is updated and when RTP packet is transmitted - scl_i is used to limit the CWND increase when close to the last known - max value, before congestion was last detected. + target_bitrate (0 bps) + Media target bitrate - The congestion window update depends on whether a loss event has - occurred, and if the congestion control is if fast start or not. + target_bitrate_last_max (1 bps) + Media target bitrate inflection point i.e. the last known highest + target_bitrate. Used to limit bitrate increase close to the last + known congestion point - ____________________________________________________________________ + rate_transmit (0.0 bps) + Measured transmit bitrate - On loss event: + rate_ack (0.0 bps) + Measured throughput based on received acknowledgements - If a loss event is detected then in_fast_start is set to false and - CWND is updated according to + rate_rtp (0.0 bps) + Measured bitrate from the media encoder - cwnd_i = cwnd + rate_rtp_median (0.0 bps) + Median value of rate_rtp, computed over more than 10s - cwnd = max(min_cwnd,cwnd*BETA) + s_rtt (0.0s) + Smoothed RTT [s], computed similar to method depicted in [RFC6298] - otherwise the CWND update continues + rtp_queue_size (0 bits) + Size of RTP packets in queue - ____________________________________________________________________ + rtp_size (0 byte) + Size of the last transmitted RTP packet - in_fast_start = true: +4.1.2. Network congestion control - in_fast_start is set to false and cwnd_i=cwnd if owd_trend >= 0.2 and - otherwise CWND is updated according to + This section explains the network congestion control, it contains two + main functions - cwnd = cwnd + bytes_newly_acked*scl_i + 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. - ____________________________________________________________________ + o Calculation of send window 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. - in_fast_start = false: + Unlike TCP, SCReAM is not a byte oriented protocol, rather it is an + RTP packet oriented protocol. Thus a list of transmitted RTP packets + and their respective transmission times (wall-clock time) is kept for + further calculation. - Values of off_target > 0.0 indicates that the congestion window can - be increased. This is done according to the equations below. + The feedback from the receiver is assumed to consist of the following + elements. - gain = GAIN*(1.0 + max(0.0, 1.0 - owd_trend/ 0.2)) + o The highest received RTP sequence number. - The equation above limits the gain when near congestion is detected + o The wall clock timestamp corresponding to the received RTP packet + with he highest sequence number. - gain *= scl_i + o Accumulated number of lost RTP packets (n_loss). - This equation limits the gain when CWND is close to its last known - max value + o Accumulated number of ECN-CE marked packets (n_ECN). - cwnd += gain * off_target * bytes_newly_acked * mss / cwnd + When the sender receives RTCP feedback, the OWD is calculated as + outlined in [RFC6817] and a number of variables are updated as + illustrated by the pseudo code below. - Values of off_target <= 0.0 indicates congestion, CWND is then - updated according to the equation + update_variables(owd): + owd_fraction = owd/owd_target + #calculate moving average + owd_fraction_avg = (1-OWD_WEIGHT)*owd_fraction_avg+ + OWD_WEIGHT*owd_fraction + update_owd_fraction_hist(owd_fraction) + # R is an autocorrelation function of owd_fraction_hist + # at lag K + a = R(owd_fraction_hist,1)/R(owd_fraction_hist,0) + #calculate OWD trend + owd_trend = a*owd_fraction_avg + owd_trend_mem = max(0.99*owd_trend_mem, owd_trend) - cwnd += GAIN*off_target*bytes_newly_acked*mss/cwnd + 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 the estimated congestion level. 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 utilized in the media rate control to + indicate incipient congestion and to determine when to exit from fast + increase mode. owd_trend_mem is used to enforce a less aggressive + rate increase after congestion events. The function + update_owd_fraction_hist(..) removes the oldest element and adds the + latest owd_fraction element to the owd_fraction_hist vector. - The equations above are very similar to what is specified in - [RFC6817]. There are however a few differences. + A loss event is detected if the n_loss counter in the feedback has + increased since the previous received feedback. Once a loss event is + detected, the n_loss counter is ignored for a full smoothed round + trip time, the intention of this is to limit the congestion window + decrease to at most once per round trip. + The congestion window backoff due to loss events is deliberately a + bit less than is the case with e.g TCP NewReno. The reason is that + TCP is generally used to transmit whole files, which can be + translated to an infinite source bitrate. SCReAM on the other hand + has a source which rate is limited to a value close to the available + transmit rate and often below said value, the effect of this is that + SCReAM has less opportunity to grab free capacity than a TCP based + file transfer. To compensate for this it is necessary to let SCReAM + reduce the congestion window slightly less when loss events occur. - 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. + An ECN event is detected if the n_ECN counter in the feedback report + has increased since the previous received feedback. Once an ECN + event is detected, the n_ECN counter is ignored for a full smoothed + round trip time, the intention of this is to limit the congestion + window decrease to at most once per round trip. The congestion + window backoff due to an ECN event is deliberately smaller than if a + loss event occurs. This is inline with the idea outlined in + [Khademi_alternative_backoff_ECN] to enable ECN marking thresholds + lower than the corresponding packet drop thresholds. - 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. + The update of congestion window depends on whether a loss or ECN or + neither occurs. The pseudo code below describes actions taken in + case of different events. - ____________________________________________________________________ + on loss(owd): + in_fast_increase = false + cwnd_last_max = cwnd + cwnd = max(min_cwnd,cwnd*BETA_LOSS) + adjust_owd_target(owd)#compensating for competing flows + calculate_send_window(owd,owd_target) - A number of final steps in the congestion window update procedure are - outlined below + on ECN(owd): + in_fast_increase = false + cwnd_last_max = cwnd + cwnd = max(min_cwnd,cwnd*BETA_ECN) + adjust_owd_target(owd)#compensating for competing flows + calculate_send_window(owd, owd_target) - ____________________________________________________________________ + # when no loss or ECN event is detected + on acknowledgement(owd): + update_bytes_newly_acked() + update_cwnd(bytes_newly_acked) + adjust_owd_target(owd) #compensating for competing flows + calculate_send_window(owd, owd_target) + check_to_resume_fast_increase() - Resume fast start: + The methods are further described in detail below. - 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. +4.1.2.1. Updating bytes_newly_acked - ____________________________________________________________________ + The bytes_newly_acked is incremented with a value corresponding to + how much the highest sequence number has increased since the last + feedback. As an example: If the previous acknowledgement indicated + the highest sequence number N and the new acknowledgement indicated + N+3, then bytes_newly_acked is incremented by a value equal to the + sum of the sizes of RTP packets with sequence number N+1, N+2 and + N+3. Packets that are lost are also included, which means that even + though e.g packet N+2 was lost, its size is still included in the + update of bytes_newly_acked. - Competing flows compensation, adjustment of owd_target: +4.1.2.2. Updating congestion window - 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: + The congestion window update is based on OWD, except for the + occurrence of loss or ECN events, which was described earlier. OWD + is 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. - owd_target = min(OWD_TARGET_HI,max(OWD_TARGET_LO, owd_norm_mean_sh* - OWD_TARGET_LO*1.1)) + Pseudo code for the update of the congestion window is found below. - ____________________________________________________________________ + update_cwnd(bytes_newly_acked): + # additional scaling factor to slow down closer to target + # The min scale factor is 0.2 to avoid that the congestion window + # growth is stalled + scale = max(0.2,min(1.0,(abs(cwnd-cwnd_last_max)/cwnd_i*4)^2)) - Final CWND adjustment step: + # action depends on whether algorithm is in fast increase + if (in_fast_increase) + if(owd_trend >= 0.2) + in_fast_increase=false + cwnd_i=cwnd + else + cwnd = cwnd + bytes_newly_acked*scale + return - The congestion window is limited by the maximum number of bytes in - flight over the last 1.0 seconds according to + # not in fast increase phase + # off_target calculated as with LEDBAT + off_target = (owd_target - owd) / owd_target + + gain = GAIN + # adapt only increase based on scale + if (off_target > 0) + gain *= (1 - owd_trend/ 0.2) * scale + # increase/decrease the congestion window + # off_target can be positive or negative + cwnd += gain * off_target * bytes_newly_acked * MSS / cwnd + # Limit cwnd to the maximum number of bytes in flight cwnd = min(cwnd, max_bytes_in_flight*MAX_BYTES_IN_FLIGHT_HEAD_ROOM) - This avoids possible over-estimation of the throughput after for - example, idle periods. + cwnd = max(cwnd, MIN_CWND) - Finally cwnd is set to ensure that it is at least min_cwnd + CWND is updated differently depending on whether the congestion + control is in fast increase or not. A Boolean variable + in_fast_increase indicates if the congestion is in fast increase + state. - cwnd = max(cwnd, MIN_CWND) + In fast increase state the congestion window is increased with the + number of newly acknowledged bytes scaled by a scale factor that + depends on the relation between CWND and the last known maximum value + of CWND (cwnd_last_max). The congestion window growth when + in_fast_increase is false is dictated by the relation between owd and + owd_target, also here the scale factor scale factor is applied to + limit the congestion window growth when cwnd gets close to + cwnd_last_max. -4.1.2.2. Transmission scheduling + The scale factor as applied above makes the congestion window grow in + a similar way as is the case with the Cubic congestion control + algorithm. - 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: + SCReAM calculates the GAIN in a similar way to what is specified in + [RFC6817]. There are however a few differences. - 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 [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. + + Further the CWND is limited by max_bytes_in_flight and min_cwnd. The + limitation of the congestion window by the maximum number of bytes in + flight over the last 5 seconds (max_bytes_in_flight) avoids possible + over-estimation of the throughput after for example, idle periods. + An additional MAX_BYTES_IN_FLIGHT_HEAD_ROOM allows for a slack, to + allow for a certain amount of media coder output rate variability. + + 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. This can + be translated to the difference between the highest transmitted byte + sequence number and the highest acknowledged byte sequence number. + As an example: If RTP packet with sequence number SN is transmitted + and the last acknowledgement indicates 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, it does not matter if for instance packet with sequence number + SN-3 was lost, the size of RTP packet with sequence number SN-3 will + still be considered in the computation of bytes_in_flight. + +4.1.2.3. Compensation for competing flows + + It is likely that a flow using SCReAM algorithm will have to share + congested bottlenecks with other flows that use a more aggressive + congestion control algorithm. SCReAM takes care of such situations + by adjusting the owr_target. + + adjust_owd_target(owd) + owd_norm = owd / OWD_TARGET_LOW + update_owd_norm_history(owd_norm) + # Compute variance + owd_norm_var = VARIATION(owd_norm_history(100)) + # Compensation for competing traffic + if (owd_norm_var < 0.16) + # Compute average + owd_norm_avg = AVERAGE(owd_norm_history(20)) + # Update target OWD + owd_target = owd_norm_avg*OWD_TARGET_LO*1.1 + owd_target = min(OWD_TARGET_HI, owd_target) + owd_target = max(OWD_TARGET_LO, owd_target) + + 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_norm is the OWD divided by OWD_TARGET_LO. owd_norm_mean_sh is the + short term (last 20 samples) average of owd_norm. owd_norm_var is + the variance of owd_norm over the last 100 samples. + +4.1.2.4. Send window calculation + + The basic design principle behind packet transmission in SCReAM is to + allow transmission 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 media 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 media bitrate will have difficulties + to increase as the congestion window puts a too hard restriction + on the media frame size variation. This can lead to occasional + queuing of RTP packets in the RTP packet queue that will further + prevent bitrate increase. 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 congestion window is adjusted depending on OWD and its relation + to the OWD target. When OWD is greater than OWD target the + congestion window enforces a strict rule that helps to prevent + further queue buildup. When OWD is less than or equal to OWD target + then an additional slack is added to the congestion window 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 occurs. The adjusted congestion window + (cwnd_s) is used in the send window calculation. - The send window is computed differently depending on OWD and its - relation to the OWD target. + The send window is given by the relation between the adjusted + congestion window and the amount of bytes in flight according to the + pseudo code below. - 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. + calculate_send_window(owd, owd_target) + # compensate for backward congestion and bitrate variations + if (owd <= owd_target) + x_cwnd=1.0+BYTES_IN_FLIGHT_SLACK*(1.0-owd_trend/0.5)/100.0 + cwnd_s = max(cwnd*x_cwnd, cwnd+MSS) - 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 - 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. + send_wnd = cwnd_s-bytes_in_flight -4.1.3. Video rate control +4.1.2.5. Resuming fast increase - 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. + Fast increase can be resumed in order to speed up the bitrate + increase in case congestion abates. The condition to resume fast + increase (in_fast_increase = true) is that owd_trend is less than + OWD_TREND_LO for T_RESUME_FAST_INCREASE seconds or more. + +4.1.3. Media rate control + + The media rate control algorithm is executed at regular intervals + RATE_ADJUSTMENT_INTERVAL, with the exception of a prompt reaction to + loss events. The media rate control operates based on the size of + the RTP packet send queue and observed loss events. In addition, + owd_trend is also considered in the media rate control, this to + reduce the amount of induced network jitter. + + The role of the media rate control is to strike a reasonable balance + between a low amount of queuing in the RTP queue and a sufficient + amount of data to send in order to keep the data path busy. A too + cautious setting leads to possible under-utilization of network + capacity and that the flow is starved out by other, more + opportunistic traffic, on the other hand a too aggressive setting + leads to extra 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 + o rate_ack: 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 + The current throughput, current_rate, is computed as the maximum + value of rate_transmit and rate_ack. The rationale behind the use of + rate_ack 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. + this shortcoming; rate_ack 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 + Note that rate_ack is updated by bytes_newly_acked, which means that + even lost packets are regarded as acknowledged. The rate change behavior depends on whether a loss event has - occurred, and if the congestion control is if fast start or not. - - ____________________________________________________________________ - - 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. + occurred, and if the congestion control is in fast increase or not. - target_bitrate_i = target_bitrate + # The target_bitrate is updated at a regular interval according + # to RATE_ADJUST_INTERVAL + on loss: + target_bitrate_last_max = target_bitrate target_bitrate = max(BETA_R* target_bitrate, TARGET_BITRATE_MIN) + exit - 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)) - + if (in_fast_increase = true) + scl_i = (target_bitrate - target_bitrate_last_max)/ + target_bitrate_last_max + increment = RAMP_UP_SPEED*RATE_ADJUST_INTERVAL* + (1.0-min(1.0, owd_trend/0.2)) + # Value 0.2 as the bitrate should be allowed to increase + # at least slowly --> avoid locking the rate to + # target_bitrate_last_max 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 - + else pre_congestion = min(1.0, max(0.0, owd_fraction_avg-0.3)/0.7) - pre_congestion += owd_trend + target_bitrate=current_rate*(1.0-PRE_CONGESTION_GUARD* + pre_congestion)-TX_QUEUE_SIZE_FACTOR *rtp_queue_size + end - The target bitrate is computed as - target_bitrate=current_rate*(1.0- - PRE_CONGESTION_GUARD*pre_congestion)-TX_QUEUE_SIZE_FACTOR - *rtp_queue_size + rate_rtp_limit = max(br, max(rate_rtp,rtp_rate_median)) + rate_rtp_limit *= (2.0-1.0*owd_trend_mem) + target_bitrate = min(target_bitrate, rate_rtp_limit) + target_bitrate = min(TARGET_BITRATE_MAX, + max(TARGET_BITRATE_MIN,target_bitrate)) - ____________________________________________________________________ + In case of a loss event the target_bitrate is updated and the rate + change procedure is exited. Otherwise the rate change procedure + continues. An ECN event does not cause any action, the reason to + this is that the congestion window is reduced less due to ECN events + than loss events, the effect is thus that the expected additional RTP + queuing delay due to ECN events is so small that an additional + decrease in media rate is not warranted. - Final step: + When in fast increase state, the bitrate increase is given by the + desired ramp-up speed (RAMP_UP_SPEED) and is limited by the relation + between the current bitrate and the last known max bitrate. + Furthermore an increased OWD trend limits the bitrate increase. The + setting of RAMP_UP_SPEED depends on preferences, a high setting such + as 1000kbps/s makes it possible to quickly gain high quality media, + this is however at the expense of a higher risk of jitter, which can + manifest itself as e.g. choppy video rendering. - As a final step, the target bitrate is limited such that it is kept - within reasonable bounds. + When in_fast_increase is false, the bitrate increase is given by the + current bitrate and is also controlled by the estimated RTP queue and + the OWD trend, thus it is sufficient that an increased congestion + level is sensed by the network congestion control to limit the + bitrate. + + In the fast increase phase an allowed increment is computed based on + the congestion level and the relation to target_bitrate_last_max and + the target_bitrate is reduced further if congestion is detected. + + If in_fast_increase is false then the target_bitrate_last_max is + updated to the current value of target_bitrate if in_fast_increase + was true the last time the bitrate was updated. Additionally, a pre- + congestion indicator is computed and the rate is adjusted + accordingly. 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) + output bitrates, rate_rtp_limit. A multiplier is applied to + rate_rtp_limit, depending on congestion history. The target_bitrate + is then limited by this 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. - - +-----------------------------+-----------------------------+-------+ - | 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. - - 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. + and max values. - 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 | - +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + The vary reader may notice the dependency on the OWD in the + computation of the target bitrate, this manifests itself in the use + of the owd_trend and owd_fraction_avg. As these parameters are used + also in the network congestion control one may suspect that some odd + interaction between the media rate control and the network congestion + control, this is in fact the case if the parameter + PRE_CONGESTION_GUARD is set to a high value. The use of owd_trend + and owd_fraction_avg in the media rate control is solely to reduce + jitter, the dependency can be removed by setting + PRE_CONGESTION_GUARD=0, the effect is a somewhat faster rate increase + at the expense of more jitter. - Figure 2: Transport layer feedback message +4.1.3.1. FEC and packet overhead considerations - To make the feedback as frequent as possible, the feedback packets - are transmitted as reduced size RTCP according to [RFC5506]. + The target bitrate given by SCReAM depicts the bitrate including RTP + and FEC overhead. Therefore it is necessary that the media encoder + takes this overhead into account when the media bitrate is set. + It is not strictly necessary to make a 100% perfect compensation for + the overhead as the SCReAM algorithm will inherently compensate + moderate errors. Under-compensation for the overhead has the effect + that the jitter will increase somewhat while overcompensation will + have the effect that the bottleneck link becomes under-utilized. - 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. +4.2. SCReAM Receiver - Section 4 describes the main algorithm details and how the feedback - is used. + The simple task of the SCReAM receiver is to feedback + acknowledgements of received packets, total loss count and total ECN + count to the SCReAM sender. Upon reception of each RTP packet the + receiver will simply maintain enough information to send the + aforementioned values to the SCReAM sender via RTCP transport layer + feedback message. The frequency of the feedback message depends on + the available RTCP bandwidth. The details of this feedback is given + in another document. -6. Discussion +5. Discussion - This section covers a few open discussion points + This section covers a few 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. + reduced size RTCP [RFC5506], 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 - 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 + o Clock drift: SCReAM can suffer from the same issues with clock + drift as is the case with LEDBAT [RFC6817]. Section A.2 in said + RFC however describes ways to mitigate issues with clock drift. -9. Implementation status +6. Implementation status [Editor's note: Please remove the whole section before publication, as well reference to RFC 6982] This section records the status of known implementations of the protocol defined by this specification at the time of posting of this Internet-Draft, and is based on a proposal described in [RFC6982]. The description of implementations in this section is intended to assist the IETF in its decision processes in progressing drafts to RFCs. Please note that the listing of any individual implementation @@ -1033,21 +982,21 @@ features. Readers are advised to note that other implementations may exist. According to [RFC6982], "this will allow reviewers and working groups to assign due consideration to documents that have the benefit of running code, which may serve as evidence of valuable experimentation and feedback that have made the implemented protocols more mature. It is up to the individual working groups to use this information as they see it". -9.1. OpenWebRTC +6.1. OpenWebRTC The SCReAM algorithm has been implemented in the OpenWebRTC project [OpenWebRTC], an open source WebRTC implementation from Ericsson Research. This SCReAM implementation is usable with any WebRTC endpoint using OpenWebRTC. o Organization : Ericsson Research, Ericsson. o Name : OpenWebRTC gst plug-in. @@ -1059,162 +1007,197 @@ However, people are encouraged to have look at it and send feedback. This wiki (https://github.com/EricssonResearch/openwebrtc/wiki) contains required information for building and using OpenWebRTC. Note that to get all the SCReAM related code and build them, one has to use the cerbero fork from DanielLindstrm' s repository (https://github.com/DanielLindstrm/cerbero/tree/scream) instead of EricssonResearch fork of cerbero. o Coverage : The code implements [I-D.ietf-rmcat-scream-cc]. The - current implementation has been tuned and tested to adapt video + current implementation has been tuned and tested to adapt a video stream and does not adapt the audio streams. o Implementation experience : The implementation of the algorithm in the OpenWebRTC has given great insight into the algorithm itself and its interaction with other involved modules such as encoder, RTP queue etc. In fact it proves the usability of a self-clocked rate adaptation algorithm in the real WebRTC system. The implementation experience has led to various algorithm improvements both in terms of stability and design. For example, improved rate increase behavior and removal of the ACK vector from the feedback message. o Contact : irc://chat.freenode.net/openwebrtc -9.2. A C++ Implementation of SCReAM +6.2. A C++ Implementation of SCReAM o Organization : Ericsson Research, Ericsson. o Name : SCReAM. o Implementation link : A C++ implementation of SCreAM is also available which is aimed for doing quick experiments[SCReAM-Cplusplus_Implementation]. This repository also includes a rudimentary implementation of a simulator. This code can be included in other simulators like NS-3. o Coverage : The code implements [I-D.ietf-rmcat-scream-cc] o Contact : ingemar.s.johansson@ericsson.com, zaheduzzaman.sarker@ericsson.com -10. Acknowledgements +7. 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. + Sjoeberg, Robert Swain, Magnus Westerlund, Stefan Aalund. Many + additional thanks to Karen and Mirja for patiently reading, + suggesting improvements and also for asking all the difficult but + necessary questions. -11. IANA Considerations +8. IANA Considerations A new RFC4585 transport layer feedback message needs to be standardized. -12. Security Considerations +9. 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 +10. Change history A list of changes: + o WG-01 to WG-02: Complete restructuring of the document. Moved + feedback message to a separate draft. + o WG-00 to WG-01 : Changed the Source code section to Implementation status section. 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 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 +11. References -14.1. Normative References +11.1. Normative References [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate - Requirement Levels", BCP 14, RFC 2119, March 1997. + Requirement Levels", BCP 14, RFC 2119, + DOI 10.17487/RFC2119, 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. + Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, + 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. + Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, + DOI 10.17487/RFC4585, 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. + and Consequences", RFC 5506, DOI 10.17487/RFC5506, April + 2009, . [RFC6298] Paxson, V., Allman, M., Chu, J., and M. Sargent, - "Computing TCP's Retransmission Timer", RFC 6298, June - 2011. + "Computing TCP's Retransmission Timer", RFC 6298, + DOI 10.17487/RFC6298, 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 + DOI 10.17487/RFC6817, December 2012, + . - [FACK] "Forward Acknowledgement: Refining TCP Congestion - Control", 2006. +11.2. Informative References [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-rmcat-cc-codec-interactions] + Zanaty, M., Singh, V., Nandakumar, S., and Z. Sarker, + "Congestion Control and Codec interactions in RTP + Applications", draft-ietf-rmcat-cc-codec-interactions-01 + (work in progress), October 2015. + + [I-D.ietf-rmcat-coupled-cc] + Islam, S., Welzl, M., and S. Gjessing, "Coupled congestion + control for RTP media", draft-ietf-rmcat-coupled-cc-00 + (work in progress), September 2015. + [I-D.ietf-rmcat-scream-cc] Johansson, I. and Z. Sarker, "Self-Clocked Rate Adaptation - for Multimedia", draft-ietf-rmcat-scream-cc-00 (work in - progress), May 2015. + for Multimedia", draft-ietf-rmcat-scream-cc-01 (work in + progress), July 2015. [I-D.ietf-rmcat-wireless-tests] Sarker, Z. and I. Johansson, "Evaluation Test Cases for Interactive Real-Time Media over Wireless Networks", draft-ietf-rmcat-wireless-tests-00 (work in progress), June 2015. [I-D.ietf-tcpm-newcwv] Fairhurst, G., Sathiaseelan, A., and R. Secchi, "Updating TCP to support Rate-Limited Traffic", draft-ietf-tcpm- newcwv-13 (work in progress), June 2015. + [Khademi_alternative_backoff_ECN] + "TCP Alternative Backoff with ECN (ABE)", + . + [OpenWebRTC] "Open WebRTC project.", . + [PACKET_CONSERVATION] + "Congestion Avoidance and Control", 1988. + [QoS-3GPP] TS 23.203, 3GPP., "Policy and charging control architecture", June 2011, . + [RFC6679] Westerlund, M., Johansson, I., Perkins, C., O'Hanlon, P., + and K. Carlberg, "Explicit Congestion Notification (ECN) + for RTP over UDP", RFC 6679, DOI 10.17487/RFC6679, August + 2012, . + [RFC6982] Sheffer, Y. and A. Farrel, "Improving Awareness of Running - Code: The Implementation Status Section", RFC 6982, July - 2013. + Code: The Implementation Status Section", RFC 6982, + DOI 10.17487/RFC6982, July 2013, + . [SCReAM-Cplusplus_Implementation] "C++ Implementation of SCReAM", . [SCReAM-Implementation] "SCReAM Implementation", . @@ -1222,245 +1205,54 @@ Control Protocol for Multimedia Streaming", December 2007, . 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 - -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. - - * 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 - 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 - 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 +A.1. Stream prioritization - o frame_skip_intensity + The SCReAM algorithm makes a good distinction between network + congestion control and the media rate control, an RTP queue queues up + RTP packets pending transmission. This is easily extended to many + streams, in which case RTP packets from two or more RTP queues are + scheduled at the rate permitted by the network congestion control. - o since_last_frame_skip, i.e the number of consecutive frames - without frame skipping + The scheduling can be done by means of a few different scheduling + regimes. For example the method applied in + [I-D.ietf-rmcat-coupled-cc] can be used. The implementation of + SCReAM use something that is referred to as credit based scheduling. + Credit based scheduling is for instance implemented in IEEE 802.17. + The short description is that credit is accumulated by queues as they + wait for service and are spent while the queues are being services. - o consecutive_frame_skips, i.e the number of consecutive frame skips + For instance, if one queue is allowed to transmit 1000bytes, then a + credit of 1000bytes is allocated to the other unscheduled queues. + This principle can be extended to weighted scheduling in which case + the credit allocated to unscheduled queues depends on the weight + allocation. - The flag skip_frame is set to 1 if any of the conditions below is - met, otherwise it is set to 0. +A.2. Computation of autocorrelation function - o age_vec(n) > 0.2 && consecutive_frame_skips < 5 + The autocorrelation function is computed over a vector of values. - o frame_skip_intensity < 0.5 && since_last_frame_skip >= 1.0/ - frame_skip_intensity + Let x be a vector constituting N values, the autocorrelation function + for a given lag=k for the vector x is given by . - o frame_skip_intensity >= 0.5 && consecutive_frame_skips < - (frame_skip_intensity -0.5)*10 + n=N-k + R(x,k) = SUM x(n)*x(n+k) + n=1 - 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. + Figure 2: Autocorrelation function Authors' Addresses Ingemar Johansson Ericsson AB Laboratoriegraend 11 Luleaa 977 53 Sweden Phone: +46 730783289