Tackling Bit-Rate Variation of RTC Through Frame-Bursting Congestion Control

Interactive video applications signal widespread interest in Real-time communication (RTC), yet issues like frame delay, rebuffering, etc. remain a common complaint. We argue that this is mainly due to legacy network-oriented congestion control (CC), which assumes a continuous stream of packets is being sent. But this assumption doesn't hold in RTC since the video encoder exhibits inherent bit-rate variation: (1) Bursty spiking bit-rate leads to packets waiting in the sending buffer, which increases frame delay. (2) Low bit-rate causes insufficient packets available for sending, making current CCs hard to detect available bandwidth. In response, we propose BurstRTC, a novel paradigm for RTC transport protocol. Each frame is emitted as a whole, and the video bit-rate is directly controlled by network congestion feedback. BurstRTC uses frame-bursting to estimate available bandwidth efficiently regardless of bit-rate variation. Considering the impact of bit-rate variation on network congestion, BurstRTC models frame size as a Gaussian distribution instead of a fixed size and further derives its frame delay, preventing suboptimal performance of purely network-oriented designs. An analytic method for determining the target bit-rate replaces the trial-and-error updates of gradient-based methods, ensuring fast convergence to the available bandwidth. We evaluated the performance of BurstRTC and found that, compared with GCC, BurstRTC achieves up to <tex xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">$59.8 \%$</tex> higher bit-rate and up to <tex xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">$\mathbf{4 8. 9 \%}$</tex> lower frame delay. Further, compared with SQP and Pudica, BurstRTC can also reduce tail frame delay by up to <tex xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">$89.2 \%$</tex>, and improve average bit-rate by up to <tex xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">$15.6 \%$</tex>.

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Publication Details

Published
2024-10-28
DOI
https://doi.org/10.1109/icnp61940.2024.10858541
Citations
4
Primary Topic
Interconnection Networks and Systems
Type
article
Field-Weighted Citation Impact
1.40
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article

Tackling Bit-Rate Variation of RTC Through Frame-Bursting Congestion Control

Qingyang Li, Yihang Zhang, Zhidong Jia, Xinggong Zhang
4 citations
Interconnection Networks and Systems
1.40
article

Tackling Bit-Rate Variation of RTC Through Frame-Bursting Congestion Control

Qingyang Li, Yihang Zhang, Zhidong Jia, Xinggong Zhang
article en
4 citations

Abstract

Interactive video applications signal widespread interest in Real-time communication (RTC), yet issues like frame delay, rebuffering, etc. remain a common complaint. We argue that this is mainly due to legacy network-oriented congestion control (CC), which assumes a continuous stream of packets is being sent. But this assumption doesn't hold in RTC since the video encoder exhibits inherent bit-rate variation: (1) Bursty spiking bit-rate leads to packets waiting in the sending buffer, which increases frame delay. (2) Low bit-rate causes insufficient packets available for sending, making current CCs hard to detect available bandwidth. In response, we propose BurstRTC, a novel paradigm for RTC transport protocol. Each frame is emitted as a whole, and the video bit-rate is directly controlled by network congestion feedback. BurstRTC uses frame-bursting to estimate available bandwidth efficiently regardless of bit-rate variation. Considering the impact of bit-rate variation on network congestion, BurstRTC models frame size as a Gaussian distribution instead of a fixed size and further derives its frame delay, preventing suboptimal performance of purely network-oriented designs. An analytic method for determining the target bit-rate replaces the trial-and-error updates of gradient-based methods, ensuring fast convergence to the available bandwidth. We evaluated the performance of BurstRTC and found that, compared with GCC, BurstRTC achieves up to <tex xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">$59.8 \%$</tex> higher bit-rate and up to <tex xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">$\mathbf{4 8. 9 \%}$</tex> lower frame delay. Further, compared with SQP and Pudica, BurstRTC can also reduce tail frame delay by up to <tex xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">$89.2 \%$</tex>, and improve average bit-rate by up to <tex xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">$15.6 \%$</tex>.

Peking University (CN)
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Interconnection Networks and Systems
1.40
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