Researcher:
Kırmızıoğlu, Rıza Arda

Profile Picture
ORCID

Job Title

Master Student

First Name

Rıza Arda

Last Name

Kırmızıoğlu

Name

Name Variants

Kırmızıoğlu, Rıza Arda

Email Address

Birth Date

Filters

2017 - 20207

Advanced Search

Filter by

Settings

Researcher Of Publications: search.filters.isAuthorOfPublication.05b28317-e49b-48df-914b-d0ce13f027a0

Search Results

Now showing 1 - 7 of 7
  • Placeholder
    PublicationMetadata only
    Multi-party videoconferencing service over edge-cloud based access network infrastructure
    (Institute of Electrical and Electronics Engineers Inc., 2020) Görkemli, Burak; Department of Electrical and Electronics Engineering; N/A; Tekalp, Ahmet Murat; Kırmızıoğlu, Rıza Arda; Faculty Member; Master Student; Department of Electrical and Electronics Engineering; College of Engineering; Graduate School of Sciences and Engineering; 26207; N/A
    This article presents SEBA based edge-cloud network service architecture and proposes a premium multi-party videoconferencing service solution to provide the lowest delay and lowest total bandwidth usage by leveraging the SEBA architecture. Today all parties in a multi-party WebRTC session connect through a single selective forwarding unit (SFU). In the proposed service architecture, a separate SFU is installed at every edge cloud and each peer connects with other peers through the SFU at its own edge access network. We provide experimental results to justify our conclusions.
  • Placeholder
    PublicationMetadata only
    Multi-party webRTC services using delay and bandwidth aware SDN-assisted IP multicasting of scalable video over 5G networks
    (IEEE-Inst Electrical Electronics Engineers Inc, 2020) N/A; N/A; Department of Electrical and Electronics Engineering; Kırmızıoğlu, Rıza Arda; Tekalp, Ahmet Murat; Master Student; Faculty Member; Department of Electrical and Electronics Engineering; Graduate School of Sciences and Engineering; College of Engineering; N/A; 26207
    At present, multi-party WebRTC videoconferencing between peers with heterogenous network resources and terminals is enabled over the best-effort Internet using a central selective forwarding unit (SFU), where each peer sends a scalable encoded video stream to the SFU. This connection model avoids the upload bandwidth bottleneck associated with mesh connections; however, it increases peer delay and overall network load (resource consumption) in addition to requiring investment in servers since all video traffic must go through SFU servers. To this effect, we propose a new multi-party WebRTC service model over future 5G networks, where a video service provider (VSP) collaborates with a network service providers (NSP) to offer an NSP-managed service to stream scalable video layers using software-defined networking (SDN)-assisted Internet protocol (IP) multicasting between peers using NSP infrastructure. In the proposed service model, each peer sends a scalable coded video upstream, which is selectively duplicated and forwarded as layer streams at SDN switches in the network, instead of at a central SFU, in a multi-party WebRTC session managed by multicast trees maintained by the SDN controller. Experimental results show that the proposed SDN-assisted IP multicast service architecture is more efficient than the SFU model in terms of end-to-end service delay and overall network resource consumption, while avoiding peer upload bandwidth bottleneck and distributing traffic more evenly across the network. The proposed architecture enables efficient provisioning of premium managed WebRTC services over bandwidth-reserved SDN slices to provide videoconferencing experience with guaranteed video quality over 5G networks.
  • Placeholder
    PublicationMetadata only
    Multi-party WebRTC videoconferencing using scalable VP9 video: from best-effort over-the-top to managed value-added services
    (Institute of Electrical and Electronics Engineers (IEEE), 2018) N/A; N/A; Department of Electrical and Electronics Engineering; Department of Electrical and Electronics Engineering; Kırmızıoğlu, Rıza Arda; Kaya, Barış Can; Tekalp, Ahmet Murat; Master Student; Undergraduate Student; Faculty Member; Department of Electrical and Electronics Engineering; Graduate School of Sciences and Engineering; College of Engineering; College of Engineering; N/A; N/A; 26207
    We propose architectures and implementations for WebRTC videoconferencing services using scalable VP9 video coding with motion-adaptive rate control as either a best-effort over-the-top service or as a managed value-added service with rate reservation over software-defined networks. In the best-effort service, clients perform motion-adaptive layer selection according to the available bandwidth in order to achieve the best overall visual video quality. In the value-added managed service, the service manager reserves bandwidth between mesh-connected clients according to rates agreed by them, and clients perform motion-adaptive layer selection to adapt their send rates to the bandwidths reserved between the end points. The proposed framework has been demonstrated to yield excellent results for both point-to-point two-party and mesh-connected multi-party videoconferencing.
  • Placeholder
    PublicationMetadata only
    Motion-based rate adaptation in WebRTC videoconferencing using scalable video coding
    (IEEE-Inst Electrical Electronics Engineers Inc, 2019) N/A; N/A; Department of Electrical and Electronics Engineering; Bakar, Gonca; Kırmızıoğlu, Rıza Arda; Tekalp, Ahmet Murat; PhD Student; Master Student; Faculty Member; Department of Electrical and Electronics Engineering; Graduate School of Sciences and Engineering; Graduate School of Sciences and Engineering; College of Engineering; N/A; N/A; 26207
    This paper proposes methods for rate adaptation by motion-based spatial and temporal resolution selection in both mesh-connected and selective-forwarding-unit (SFU) connected WebRTC videoconferencing using scalable video coding. In the mesh-connected case, the proposed motion-adaptive spatial/temporal layer selection allows each peer to send video to different peers with different terminal types and network rates at different rates using a single encoder. In the SFU-connected case, motion-adaptive rate control is used both at peers to adapt to the network rate between the sending peer and SFU by spatio-temporal resolution adaptation and at the SFU by layer selection to adapt to the network rate between the SFU and receiving peer. Experimental results show that our proposed motion-based rate adaptation achieves better perceptual video quality with sufficiently high frame rates and lower quantization parameter for video with high motion; and high spatial resolution and lower quantization parameter for video with low motion compared to simple rate-distortion model-based layer selection that does not use motion complexity, at the same rate.
  • Placeholder
    PublicationMetadata only
    Multi-party WebRTC as a managed service over multi-operator SDN
    (Springer Science and Business Media Deutschland GmbH, 2020) Department of Electrical and Electronics Engineering; N/A; Tekalp, Ahmet Murat; Kırmızıoğlu, Rıza Arda; Faculty Member; Master Student; Department of Electrical and Electronics Engineering; College of Engineering; Graduate School of Sciences and Engineering; 26207; N/A
    We propose a new WebRTC premium service API and a distributed E2E WebRTC service management framework to enable provisioning premium WebRTC services across multiple network service providers (NSP) with software defined network (SDN) infrastructure. The proposed architecture supports both two-party point-to-point (P2P) and multi-party videoconferencing (via a selective forwarding unit (SFU)) with end-to-end (E2E) quality of service (QoS) as a value-added service. Each NSP has full control of its own network resources. We propose a WebRTC service manager (WSM) that is hosted by each NSP. WSMs collaborate with each other in a distributed manner to orchestrate a network slice with specified bitrate and delay over the E2E path between two clients in P2P service and between sending clients and SFU and between the SFU and receiving clients in multi-party service without a need for a central inter-operator service orchestration authority. Clients support scalable VP9 video codec, which is configured according to agreed client upload and download bitrates. Our experimental results show that the proposed managed WebRTC service with guaranteed video quality across multiple NSPs provides lower delay and consumes less network resources compared to today’s best-effort WebRTC communications.
  • Placeholder
    PublicationMetadata only
    Motion-based adaptive streaming in WebRTC using spatio-temporal scalable VP9 video coding
    (Institute of Electrical and Electronics Engineers (IEEE), 2017) N/A; N/A; Department of Electrical and Electronics Engineering; Bakar, Gonca; Kırmızıoğlu, Rıza Arda; Tekalp, Ahmet Murat; PhD Student; Master Student; Faculty Member; Department of Electrical and Electronics Engineering; Graduate School of Sciences and Engineering; Graduate School of Sciences and Engineering; College of Engineering; N/A; N/A; 26207
    WebRTC has become a popular platform for real-time communications over the best-effort Internet. It employs the Google Congestion Control algorithm to obtain an estimate of the state of the network. The default configuration employs single-layer CBR video encoding given the available network rate with rate control achieved by varying the quantization parameter and video frame rate. Recently, some open-source WebRTC platforms provided support for VP9 encoding with spatial scalable encoding option. The main contribution of this paper is to incorporate motion-based spatial resolution adaptation for adaptive streaming rate control and evaluate the use of single-layer (non-scalable) VP9 encoding vs. motion-based mixed spatio-temporal scalable VP9 encoding in point-to-point RTC between two parties in the presence of network congestion. Our results show that, during intervals of high motion activity, spatial resolution reduction with sufficiently high frame rates and reasonable quantization parameter values yield more pleasing video quality compared to the standard rate control scheme employed in open-source WebRTC implementations, which uses only quantization parameter and frame rate for rate control.
  • Placeholder
    PublicationMetadata only
    Multi-party videoconferencing service over edge-cloud based access network infrastructure
    (IEEE, 2020) Gorkemli, Burak; N/A; Department of Electrical and Electronics Engineering; Kırmızıoğlu, Rıza Arda; Tekalp, Ahmet Murat; Master Student; Faculty Member; Department of Electrical and Electronics Engineering; Graduate School of Sciences and Engineering; College of Engineering; N/A; 26207
    This article presents SEBA based edge-cloud network service architecture and proposes a premium multi-party videoconferencing service solution to provide the lowest delay and lowest total bandwidth usage by leveraging the SEBA architecture. Today all parties in a multi-party WebRTC session connect through a single selective forwarding unit (SFU). In the proposed service architecture, a separate SFU is installed at every edge cloud and each peer connects with other peers through the SFU at its own edge access network. We provide experimental results to justify our conclusions. / Bu bildiride SEBA temelli sınır-bulut ağ mimarisi anlatılmakta ve bu mimariden yararlanarak en düşük gecikme ve en düşük toplam bant genişigi kullanan üst düzey bir çoktaraflı videokonferans servisi önerilmektedir. Günümüzün çoktaraflı WebRTC servisinde tüm taraflar tek bir seçici yönlendirme birimine (SFU) baglanır. Önerilen sınır-bulut tabanlı videokonferans servisinde her sınır-bulut altyapısına bir SFU kurulur ve her WebRTC eşi kendi sınır ağındaki SFU üzerinden diğer eşlere video iletir. Önerilen çözüm, benzetimlerle dogrulanmıştır.