A systematic review on WebRTC for potential applications and challenges beyond audio video streaming

Abstract

Video conferencing and live streaming are being used in various industries, such as healthcare, gaming, telecommunication, manufacturing and others. As technology progresses, the need for real-time data transmission with minimal latency has increased. Web Real-Time Communication (WebRTC) addresses this need effectively. WebRTC is a technology designed to provide real-time communication through web and mobile browsers. Its low latency and P2P communication capabilities make it a convenient technology for secure, efficient communication in real-time applications. This paper reviews the key features of WebRTC, discusses its strengths and weaknesses and investigates a detailed analysis of 83 existing studies. Moreover, It evaluates all use cases that can be adopted by WebRTC by examining their descriptions, problem statements, and research gaps based on literature to date. Finally, It highlights the open research directions for the emerging technologies and enhancements of WebRTC. to identify their potential applications.

Full text

Multimedia Tools and Applications (2025) 84:2909–2946
https://doi.org/10.1007/s11042-024-20448-9
A systematic review on WebRTC for potential applications
and challenges beyond audio video streaming
Haitham Mahmoud1
·Raouf Abozariba1
Received: 16 June 2023 / Revised: 24 September 2024 / Accepted: 3 November 2024 /
Published online: 23 November 2024
© The Author(s) 2024
Abstract
Video conferencing and live streaming are being used in various industries, such as healthcare,
gaming, telecommunication, manufacturing and others. As technology progresses, the need
for real-time data transmission with minimal latency has increased. Web Real-Time Com-
munication (WebRTC) addresses this need effectively. WebRTC is a technology designed to
provide real-time communication through web and mobile browsers. Its low latency and P2P
communication capabilities make it a convenient technology for secure, efficient communi-
cation in real-time applications. This paper reviews the key features of WebRTC, discusses its
strengths and weaknesses and investigates a detailed analysis of 83 existingstudies. Moreover,
It evaluates all use cases that can be adopted by WebRTC by examining their descriptions,
problem statements, and research gaps based on literature to date. Finally, It highlights the
open research directions for the emerging technologies and enhancements of WebRTC. to
identify their potential applications.
Keyword
1 Introduction
The demand for high-quality, low-latency streaming increased sharply due to rise in remote
working and the need for real-time collaboration in online gaming, education, and health-
care [1]. WebRTC offers a secure, open-source, and scalable browser-based solution for
both individuals and organisations in these fields [2]. As data privacy and security become
more critical, the need for secure real-time communication tools is also growing. WebRTC
addresses this by providing a platform that protects sensitive user data [3]. Its compatibility
with other web technologies makes it easy to integrate into existing systems and applications.
The motivation to explore WebRTC beyond video conferencing stems from the increasing
need for accessible, secure, cost-effective, and real-time communication capabilities across
various applications.
BHaitham Mahmoud
haitham.mahmoud@bcu.ac.uk
Raouf Abozariba
raouf.abozariba@bcu.ac.uk
1Faculty of Computing, Engineering and Built Environment, Steamhouse, Belmont Row, Birmingham
B4 7RQ, UK
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2910 Multimedia Tools and Applications (2025) 84:2909–2946
WebRTC was first released in 2011 and has since become a key component in many appli-
cations. Initially, WebRTC was used primarily for video conferencing. With the maturation
of the technology, it became apparent that it had the potential to be used for a wide range
of real-time communication applications. Among the many applications of this technology
are gaming, health, haptic devices, and data collection (such as water sensing). It has been
shown that WebRTC can deliver low-latency, real-time communication to enhance the gam-
ing experience for users [4,5]. This marked a significant milestone in WebRTC’s evolution,
showcasing its potential beyond just video conferencing. As the healthcare industry’s need for
secure real-time communication grew, WebRTC emerged as a promising solution. In 2016,
Antunes et al. [6] evaluated the feasibility of using WebRTC in telemedicine, demonstrating
that it provides a secure platform for real-time communication, including bidirectional file
sharing and a whiteboard interface. Today, WebRTC has evolved into a versatile technology
in various real-time communication applications, from gaming and education to e-commerce
and financial services.
Despite its wide usage, there is a clear research gap in reviewing applications that utilise
WebRTC, as highlighted in existing studies. This study aims to systematically review the
key applications that use WebRTC as an enabling tool, evaluating them based on criteria
such as user experience, network connectivity, server infrastructure, interoperability and
security. Moreover. we examined the adaption requirements for supporting non-video and
non-audio data, typical attacks on WebRTC, and mitigation techniques. Furthermore, we
comprehensively reviewed the existing and potential applications that can be enabled by
WebRTC, focusing on problem statements, existing literature, and research gaps for each of
them. Finally, we addressed the future research challenges for WebRTC in terms of different
emerging technologies. The main contributions of this study are summarised below:
•We conducted a systematic literature review in four phases of identifying relevant stud-
ies and examining academic articles related to WebRTC. The method excludes papers
describing WebRTC applications without explaining their use and studies written in
languages other than English. This review also highlights the need for more scholarly
publications.
•We performed an extensive analysis of 83 WebRTC studies, evaluating them based on
criteria such as user experience and security requirements. The evaluation highlighted
several research gaps.
•Beyond existing WebRTC applications, this study identified additional applications that
had not been previously recognised. For each application, we provided a comprehensive
analysis of the utilisation of WebRTC as enabling technology, emphasising the problem
statement, current literature, research gaps, and the importance of these gaps.
•This study analysed the research gaps in WebRTC applications and reviewed survey
studies to highlight open research challenges. This study also examined challenges related
to devices and other open issues, offering a roadmap for future research.
The rest of the paper is organised as follows as shown in Fig. 1: Section 2covers the
evolution of WebRTC architecture, its adaptation to non-video data, and a review of the
typical attacks and mitigation techniques. Section 3proposes a systematic framework for
analysing WebRTC applications by addressing the key requirements for its applications
and presenting the survey methodology. Section 4discusses the results from the systematic
framework, reviews the existing survey papers, examines the advantages and disadvantages,
and provides a comprehensive review of each application in terms of the problem statement,
literature to date and research gap. Section 5investigates the open research directions within
the WebRTC domain. Finally, Section 6concludes the work.
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Multimedia Tools and Applications (2025) 84:2909–2946 2911
Fig. 1 Structure of the paper
2 WebRTC
2.1 Motivation and research questions
This survey was driven by the increasing relevance of real-time communication technologies
in our digitally connected world. WebRTC is an open-source platform, which has become
a key technology for enabling real-time communication through web browsers and mobile
devices. Despite its success, most existing studies focus on its traditional use in video confer-
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2912 Multimedia Tools and Applications (2025) 84:2909–2946
encing, leaving a gap in understanding its broader applications. As industries like healthcare,
gaming, and education continue to evolve, there’s a pressing need to explore how WebRTC
can be adapted to meet the unique demands of these sectors. This survey seeks to fill that
gap by thoroughly reviewing the literature, identifying new opportunities for WebRTC, and
addressing the challenges that must be overcome to unlock its full potential. The survey is
written based on these research questions:
•RQ1: What are the current uses of WebRTC outside of traditional video conferencing,
and how well does it serve the needs of various industries?
•RQ2: What challenges and limitations arise when implementing WebRTC in different
real-time communication settings, especially regarding security, scalability, and compat-
ibility with other systems?
•RQ3: What new applications could benefit from WebRTC, and what specific research
areas need to be addressed to enhance its use in these emerging fields?
•RQ4: How can WebRTC be improved to better handle non-video data, and what changes
are needed in its architecture and protocols to support this?
2.2 Background
WebRTC allows browsers to communicate peer-to-peer (P2P) in addition to the traditional
client-server model. The WebRTC architecture is based on a model similar to the Session
Initiation Protocol (SIP) Trapezoid (see Fig. 2).
In this model, WebRTC enables web applications downloaded from different web servers
to connect both browsers [7]. There are several other technologies available that enable real-
time communication, each offering distinct advantages. SIP is commonly used for voice
Fig. 2 Typical WebRTC Architecture of two web servers handling signalling between browsers using propri-
etary protocols over HTTP or WebSockets. After signalling, the browsers establish a direct P2P media path
through the WebRTC PeerConnection API, with the cloud representing external services like STUN/TURN
servers for secure communication
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Multimedia Tools and Applications (2025) 84:2909–2946 2913
and video calls, particularly in VoIP systems. RTMP is often used for live video broadcasts,
while HTTP Live Streaming (HLS) breaks video into chunks, making it more suitable for
large-scale streaming, although with increased latency. Proprietary systems like those used by
Zoom and Microsoft Teams provide specialized real-time communication solutions but are
generally less adaptable than the open-source WebRTC. Additional protocols such as QUIC
and Extensible Messaging and Presence Protocol (XMPP) offer secure communication but
are less widely used. Tools like Google Remote Procedure Call (gRPC) support fast, low-
latency communication between distributed systems. However, compared to WebRTC, these
alternatives often require more server resources and may not offer the same level of peer-
to-peer flexibility. Signaling messages are used to set up and end communications. These
messages are sent via HTTP or WebSocket protocols to the web servers, which can modify or
manage them as needed. WebRTC does not standardise the signalling between browsers and
servers since it is considered part of the application [8]. Data is transferred between browsers
directly via a PeerConnection without an intermediary server. The two web servers can use a
standard signalling protocol such as SIP. Alternatively, a proprietary signalling protocol may
be used.
In most WebRTC cases, both browsers run the same web application from the same
site. This simplifies the architecture, changing the Trapezoid model into a Triangle (see
Fig. 3). The transition from trapezoid SIP to triangle architecture involves tools like Net-
work Address Translation Traversal (NATS), Interactive Connectivity Establishment (ICE),
Session Traversal Utilities for NAT (STUN), and Traversal Using Relay around NAT (TURN)
to ensure secure communication [9]. NATS provides secure message delivery for real-time
communication. ICE is a technique for traversing firewalls and NATs to allow direct P2P
communication. STUN determines a device’s public IP and port, while TURN acts as a relay
server if direct communication fails. These components work together to ensure secure and
reliable communication in WebRTC architecture.
For example, in WebRTC’s triangle model, the first user’s browser sends a message to the
second user’s browser using NATS to start a video call. NATS handles the secure delivery of
this message. Once the second browser receives the message, the ICE negotiation begins,
with each browser sending STUN requests to find their public IP and port and check for NATs.
The STUN server helps determine this information. If a direct P2P connection is possible, the
video call is established. However, if network restrictions prevent direct communication, one
browser acts as a TURN server, relaying the video and audio data between the two servers.
TURN acts as a backup to ensure communication even if NATs block direct connections.
Overall, the signalling mechanism, NAT,ICE,STUN,andTURN, work together to ensure
secure and reliable communication between the two browsers involved in the video call. The
details about these components are defined in the next subsection.
2.3 Architecture
The key components of WebRTC are essential for enabling real-time communication between
browsers (See Fig. 3). First, the Media Path API allows the system to access media streams,
such as audio and video, directly from the user’s device. This is crucial for applications like
video conferencing or voice calls, where capturing and transmitting media is fundamental.
PeerConnection API sets up a P2P connection between two browsers, facilitating the transfer
of media streams and other types of data directly between them. This ensures efficient and
low-latency communication, bypassing the need for a server to handle the data exchange.
Complementing this is the RTCDataChannel API, which provides a secure and efficient
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2914 Multimedia Tools and Applications (2025) 84:2909–2946
Fig. 3 Secure WebRTC Architecture of two browsers connecting to a web server, including a STUN server,
via HTTP signalling. Media communication between the browsers is routed through a TURN server to ensure
secure and reliable data transmission, particularly in scenarios where direct P2P connections are not feasible
due to network constraints
way to transfer non-media data over the P2P connection, ensuring smooth communication
for file sharing or real-time messaging.
WebRTC also uses various Codecs for media compression. For video, codecs like VP8,
VP9, H.264, and H.265 are employed, while Opus and G.711 handle audio compression,
ensuring efficient use of bandwidth while maintaining media quality. To handle NAT Traver-
sal, WebRTC uses ICE, allowing devices behind different networks or firewalls to connect
directly. Signaling is another critical component, used to exchange connection information
between devices. It can be implemented using various protocols such as SIP, Jingle, or even a
custom protocol. ICE: a technique for establishing direct communication between browsers
by traversing firewalls and NATs. As part of the NAT traversal process, STUN helps discover
a device’s public IP and port behind a NAT, while TURN acts as a relay server when direct
communication between devices is not possible. Security is a top priority in WebRTC, which
uses Secure Real-time Transport Protocol (SRTP) and Datagram Transport Layer Security
(DTLS) to encrypt the media and data streams. It also uses ICE to ensure the connection is
secure and private.
To use WebRTC for data transfer instead of video, the architecture would need adjustments
to support this new functionality [10]. First, the data channel would need to be optimised
to efficiently transfer non-video data, potentially requiring the addition of new features or
protocols. This would ensure the data transfer is fast and reliable, without data loss during
transmission. Another aspect that would need attention is the codecs used by WebRTC. The
video codecs may not be suitable for efficient data transfer, so new codecs may need to be
developed or modified. This would allow for more efficient compression and decompression
of data, improving the overall speed and reliability of data transfer.
Second, the network protocols used by WebRTC may need to be updated or modified
to accommodate the transfer of non-video data. This could include changes to how data
is transmitted and received to ensure efficient and fast transfer. Security would also be a
key concern when transferring non-video data. The security measures for WebRTC video
conferencing may need to be adapted or extended to ensure secure data transfer. This would
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Multimedia Tools and Applications (2025) 84:2909–2946 2915
include encryption of data in transit, secure authentication of users, and protection against
hacking and other forms of cyberattacks.
Finally, interoperability would be important when using WebRTC for data transfer. To sup-
port data transfer between different systems and applications, the interoperability of WebRTC
may need to be improved, potentially including the development of new APIs or the modi-
fication of existing APIs. This would allow for seamless communication between different
systems and applications, making it easier for users to transfer data.
3 Framework for analysis of WebRTC applications
3.1 Key requirements
When implementing WebRTC technology in applications, several key factors should be
considered. By addressing these factors, the quality and suitability of a WebRTC application
can be effectively evaluated for any specific use case. These factors are as follows:
•User experience is critical, focusing on how easily users can initiate and use the applica-
tion and the overall quality of audio and video communication.
•Network connectivity is to ensure the application can handle varying network conditions
such as low bandwidth or high latency.
•Security is also essential, requiring the application to protect sensitive user data during
transmission.
•Server infrastructure must be scalable and reliable to support large volumes of users and
traffic.
•Interoperability is another consideration, ensuring the WebRTC application works seam-
lessly with other systems or communication platforms.
•Real-time communication must be supported to meet the demands of immediate data or
media exchange.
3.2 Survey methodology
To thoroughly evaluate the potential of using WebRTC in the literature, we employed a
systematic review method developed by Keele et al. [11] to analyse and examine existing lit-
erature in this research area (see Fig. 4). This method specifies each stage of our investigation
are provided in this section. At the beginning of our research, we conducted a preliminary
review of recent literature to better understand the problem and main contributions. Once we
determined the feasibility of the research, we divided our systematic review methodology
into four stages: selection, identification, screening and refinement, and compilation. These
stages allowed for a structured and unbiased collection, evaluation, and synthesis of data
pertinent to WebRTC, ensuring the inclusion of high-quality studies. These four stages are
described as follows:
3.2.1 Selection phase
In the selection phase, we first chose four scientific databases to extract relevant publications
from IEEE Xplore, Springer, Science Direct, and ACM. We used keywords like “WebRTC
applications” and “Potential applications for WebRTC” to find relevant material. Moreover,
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2916 Multimedia Tools and Applications (2025) 84:2909–2946
Fig. 4 Survey Methodology of the Systematic Literature review by Keele et al. [11] of four phases; selection
phase, Identification phase, screening and refinement phase and compilation of results phase
we used existing conference and journal publication studies to carry out our preliminary study.
We utilised the following terms in the advanced research in the databases ’Applications’ AND
’WebRTC’.
3.2.2 Identification phase
We reviewed further works on the topic using the snowballing methodology. Within the
review, we have found the studies’ repetition and duplication of a couple of studies. This led
to a deliberate effort to remove all such instances so that items appearing twice were only
counted against the publisher’s database. We found 83 unique papers across all databases.
3.2.3 Screening and refinement phase
To filter the literature, we performed a screening and refinement phase following the general
search conducted in the previous phases. The screening process involved an initial assessment
of the titles and abstracts of all studies to determine their relevance to our research focus on
WebRTC applications. studies that clearly did not meet the criteria described in Section 3.1.
Moreover, studies that focused on unrelated technologies or general communication protocols
without direct ties to WebRTC were excluded at this stage.
In the refinement stage, a more in-depth evaluation was conducted. This involved a full-text
review of the remaining articles to assess the methodological quality, relevance, and depth
of findings. Each article was scrutinised against a set of predefined inclusion and exclusion
criteria:
Inclusion Criteria: Studies were included if they specifically addressed WebRTC applica-
tions, provided empirical or theoretical evaluations of its use cases, or discussed the potential
of WebRTC in real-time communication environments. studies that demonstrated clear exper-
imental setups, validation, or case studies of WebRTC implementations were prioritised.
Exclusion Criteria: Studies were excluded if they were not written in English, lacked
a clear methodology, or if WebRTC was only mentioned superficially without substantial
analysis or discussion. Duplicate studies, opinion pieces, and publications that did not undergo
peer review were also removed. Moreover, we removed articles in languages other than
English and papers that did not mention using WebRTC as an application.
Furthermore, studies were assessed for bias, ensuring that findings were not overly influ-
enced by commercial interests or conflicts of interest. The refinement process also involved
cross-referencing the findings of the selected studies to identify consistency and gaps, pro-
viding a comprehensive view of the current state of WebRTC applications.
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Multimedia Tools and Applications (2025) 84:2909–2946 2917
3.2.4 Compilation of results phase
As a result of the revisions, 83 unique publications were selected to be included in this study;
these papers are classified based on their application. The articles identified each application
that can use WebRTC, the literature to date, and the research gap, as discussed in the following
Sections 4and 5. Using forward snowballing, additional outcomes are also extracted from
each application scenario.
3.3 Methodology analysis
Figure 5presents an analysis of the extracted publications based on publication year, type,
top 10 subject areas (keywords), and top 10 publishers. The publications per year highlight
the growing importance of real-time communication over time. Most contributions are con-
ference papers, indicating quicker, smaller-scale research with room for further development.
Key terms such as QoE, real-time communication, P2P, and web applications reflect the rel-
evance of the systematic literature review’s focus. Leading publishers, including Springer,
Elsevier, MDPI, ACM, and IEEE, emphasize the significance of this research area.
4 Comparative analysis of WebRTC applications
4.1 Results and discussion
The survey methodology followed in this study involved a thorough examination of 83
research papers on WebRTC, with a focus on the key requirements of user experience, network
connectivity, security, server infrastructure, interoperability and real-time performance. The
research papers were sourced from multiple reputable databases, including IEEE Xplore,
ACM Digital Library, and SpringerLink, spanning. studies were selected based on their
relevance, ensuring a comprehensive coverage of WebRTC advancements and challenges.
The analysis was based on the key criteria of user experience, network connectivity, security,
server infrastructure, interoperability, and real-time performance (see Tables 1,2,3). The
inclusion criteria for the papers involved studies that explicitly discussed the technical and
implementation aspects of WebRTC. Exclusion criteria were set to eliminate papers that
focused solely on theoretical or speculative analysis without providing empirical data or
prototype implementation. Each paper was rigorously reviewed to extract key insights related
to the six criteria, with findings documented in a structured format to facilitate comparison
and synthesis.
Most studies in the literature review have focused on enhancing video-conferencing appli-
cations and how they can be applied in different industries such as healthcare, gaming, and
virtual reality. The primary focus of most studies has been to improve the quality of transmit-
ted data to enhance the user experience, especially given that WebRTC works with multimedia
data and any data loss can significantly impact performance.
The network connectivity of WebRTC systems has been a key area of focus, emphasising
the parameters of reliability and scalability. Research highlights that connectivity issues
often arise from NAT traversal and firewall complications, which significantly impact the
quality of service (QoS) in real-time applications. Solutions such as STUN, TURN, and
ICE protocols have been explored to mitigate these issues but remain suboptimal under high
network congestion scenarios [12]. While the transmitted data is of high quality, it must
also be stable and reliable to meet the requirements of real-time applications. Scalability
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2918 Multimedia Tools and Applications (2025) 84:2909–2946
(a) Publications per year
(b) Types of Publications
(c) Top 10 Subject area
(d) Top 10 Publishers
Fig. 5 Data Visualisation of the Extracted papers
remains a challenge in the field, and further research is needed to address this issue. The
performance of WebRTC systems has not been fully assessed, which may result in poor
real-time performance and high latency when the system cannot accommodate many users.
Concerning security, only a few studies have discussed using cryptography to encrypt
transmitted data to maintain its confidentiality [13]. The Secure Real-time Transport Pro-
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Multimedia Tools and Applications (2025) 84:2909–2946 2919
Table 1 Framework Results from Systematic review on Video and Speech Systems
Ref Description QoE/QoS Network connectivity Security Server Infrastructure Interoper- ability Real-time
Video and Speech Systems
[23–26] Video Streaming in 5G Networks High Low Latency No Reliable No Yes
[14,27,28] browser-based P2P video streaming in
resource constrained devices
High Low BW and Latency No Reliable and Scalable No Yes
[29] WebRTC signaling mechanisms and
build video conferencing
High Low latency No Reliable No Yes
[30] An implementation of Audio trans-
mission using WebRTC
High Low BW and Latency No Reliable No Yes
[31] Real-Time Analysis of Video Streams
on edge
High Low Latency No Reliable and Scalable Yes Yes
[32] Comparing real-time screen-sharing
technologies
High Low Latency No Reliable No Yes
[33,34] performance analysis of client-server
applications by SIP and WebRTC.
High Low Latency No Reliable No Yes
[35] Implementing a novel WebRTC sig-
nalling mechanism
High Low BW and Latency Yes Reliable No Yes
[36] Investigating the privacy for WebRTC
using symmetric encryption algorithm
High Low Latency Yes Reliable and Scalable No Yes
[37] Conceptualising privacy-preserving
and maintaining security between
peers.
High Low Latency Yes Reliable and Scalable No Yes
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Table 1 continued
Ref Description QoE/QoS Network connectivity Security Server Infrastructure Interoper- ability Real-time
Video and Speech Systems
[38] Full Reference (FR) analysis of video
and audio models using WebRTC
High Low Latency No Reliable No Yes
[39] Synchronised mixing of audio/video
streams from multiple peers.
High Low Latency No Reliable No Yes
[40] A server-less framework for edge-
based live-video streaming.
High Low Latency No Reliable Yes Yes
[41] A WebRTC-compliant framework for
scalability.
High Low Latency No Reliable and Scalable No Yes
[42] Utilising a Kalman filter for conges-
tion control
High Low Latency No Reliable No Yes
[43] Web based video and audio streaming
challenges
High Low Latency No Reliable No Yes
[44] Securing web real-time communica-
tion using encryption
High Low Latency Yes Reliable No Yes
[45] Videoconferencing security models
based on simple access control lists
and capabilities.
High Low Latency Yes Reliable No Yes
[46] An adjusting bitrate algorithm for
speech applications
High Low BW and Latency No Reliable No Yes
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Table 2 Framework Results from Systematic review on Streaming Applications, agaming, AR, and VR applications and P2P video and file sharing applcations
Ref Description QoE/QoS Network connectivity Security Server Infrastructure Interoper- ability Real-time
Streaming applications
[47] A load testing WebRTC services High Low Latency No Reliable No Yes
[48] Introduced WebRTC for the open web platform. – – – Reliable Yes Yes
[49,50] Live streaming application High Low Latency No Reliable No Yes
Gaming, AR and VR applications
[48,51,52] Introduced WebRTC for open web platform. – – – Reliable Yes Yes
[53,54] Displaying Videos on computer screens using AR. High Low BW and Latency Yes Reliable Yes Yes
[55,56] Interactive Video Conferencing High Low BW and Latency Yes Reliable Yes Yes
[57] 360 degree VR High Low BW and Latency Yes Reliable Yes Yes
[58] Interactive Educational Platform High Low BW and Latency Yes Reliable Yes Yes
[59,60] Utilising WebRTC for game streaming High Low BW and Latency Yes Reliable Yes Yes
[61,62] Developed to support multiplayer gaming High Low BW and Latency Yes Reliable Yes Yes
[63] Cloud gaming using WebRTC and Nvidia GeForce High Low bandwidth Yes Reliable Yes Yes
P2P Video and file sharing applications
[64–66] file sharing decentralised system High Low BW and Latency Yes Reliable Yes Yes
[67] DICOM sharing system High Low BW and Latency Yes Reliable Yes Yes
[68] Selenium automation for WebRTC testing High Low latency No Reliable Yes Yes
[69] QUIC-based signaling for WebRTC in impaired networks High Low latency No Reliable Yes Yes
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Table 3 Framework Results from Systematic review on Healthcare Systems, Websites, Mobile real-time analyses applications, Censorship Circumventing, Educatoinal Systems,
IT applications and Vehicular Systems
Ref Description QoE/QoS Network connectivity Security Server Infrastructure Interoper- ability Real-time
Healthcare systems
[70] Behavioural health clinical care High Low BW and Latency Yes Reliable Yes Yes
[71] A Medicine file sharing High Low Latency Yes Reliable Yes Yes
[72] Telepresence wheelchair with 360-degree vision for
healthcare applications
High Low Latency Yes Reliable Yes Yes
[73] Medical Image presentation High Low Latency No Reliable Yes Yes
[74–82], Telehealth consulting System High Low Latency No Reliable Yes Yes
[83] Interactive Monitoring system using AR High Low Latency No Reliable Yes Yes
[84] Extended Reality for telemedicine using WebRTC High Low latency Yes Reliable Yes Yes
Websites, Mobile real-time analytics applications and machine learning processing
[85 ?] Mobile based analytics system High Low BW and Latency No Reliable Yes Yes
[86–88] Automated machine learning High Low BW and Latency No Reliable Yes Yes
Censorship circumventing
[89,90] Censorship circumventing High Low BW and Latency No Reliable Yes Yes
Educational Systems
[91–94] An online Educational system High Low BW and Latency No Reliable Yes Yes
IoT applications
[95] A heterogeneous network for web D2D on the devices High Low Latency No Reliable and Efficient Yes Yes
[96] An auditory impacted assistive technology High Low Latency No Reliable Yes Yes
[97] Smart water supply systems using WebRTC Medium Low bandwidth No Reliable Yes Yes
Vehicular Systems
[12] Connected and autonomous vehicle discovery High Low Latency No Reliable Yes Yes
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Multimedia Tools and Applications (2025) 84:2909–2946 2923
tocol (SRTP) is commonly used; however, vulnerabilities still exist, particularly concerning
man-in-the-middle attacks and Denial of Service (DoS) threats. Recent advances suggest
incorporating more robust cryptographic algorithms, but implementation remains limited in
practical applications.
However, the interoperability of WebRTC systems with other technologies has not been
widely considered in the studies conducted to date [14]. Interoperability issues particularly
arise when integrating WebRTC with legacy systems or non-browser-based applications,
hindering seamless communication and data exchange across platforms.
In WebRTC development, the software and tools used in the literature are leveraged
to establish real-time peer-to-peer communication, ensuring seamless audio, video, and
data transfer. The core WebRTC APIs, such as RTCPeerConnection, RTCDataChannel,
and getUserMedia(), are central to capturing media streams and handling peer connections
directly within web browsers [14]. RTCPeerConnection is primarily responsible for estab-
lishing and maintaining direct connections between peers, and facilitating the exchange of
audio, video, and other data streams without the need for an intermediary server. RTCDat-
aChannel allows developers to transfer arbitrary data, including files and game states, over
low-latency, bidirectional connections, which is critical for applications beyond traditional
video conferencing [15]. Adapter.js is often implemented to resolve cross-browser compati-
bility issues, providing a JavaScript shim that normalizes WebRTC behaviour across different
browsers such as Chrome, Firefox, and Safari, thereby enhancing the reliability of WebRTC
applications in production environments [16].
Libraries like PeerJS and SimpleWebRTC simplify the process of peer communication
and data transfer [17]. PeerJS abstracts the complexities involved in peer-to-peer networking,
providing an easy-to-use API that automates signalling and peer connection management,
while SimpleWebRTC builds on top of the core APIs, adding features like room management,
video grids, and a higher-level approach to peer discovery and communication. In more com-
plex use cases, media servers like Kurento, Janus, and Jitsi are utilized to manage large-scale,
multi-peer communication, enabling features like media transcoding and video conferencing.
Kurento Media Server, for instance, allows for real-time processing of multimedia streams,
offering advanced capabilities like augmented reality filters and computer vision analysis.
Janus, on the other hand, is known for its modularity, supporting multiple plugins that cater
to specific needs such as streaming, recording, and SIP gateway functionality. Jitsi, widely
recognized for its ease of deployment, is often used in scalable video conferencing solutions
due to its robust support for simulcast and SVC (Scalable Video Coding).
For signalling, developers often rely on WebSockets or Socket.IO to manage connection
setups and exchange signalling data, such as session descriptions and ICE candidates [18].
WebSockets provide a full-duplex communication channel over a single, long-lived con-
nection, essential for real-time bidirectional data exchange between the client and server,
whereas Socket.IO adds additional layers of abstraction, including automatic reconnection,
disconnection detection, and multiplexing capabilities. STUN and TURN servers assist with
NAT traversal, helping peers discover their public IP addresses and facilitating connections
when direct communication is hindered by network restrictions. TURN servers, in particular,
relay media between peers in cases where direct peer-to-peer connections fail, albeit with an
added latency overhead [19]. Moreover, QUIC is gaining popularity for its reduced latency
and network resilience, offering improved performance over traditional TCP connections by
incorporating congestion control, forward error correction, and multiplexing within a single
connection [20]. SRTP ensures the security of media streams, providing encryption, mes-
sage authentication, and integrity for the RTP packets, safeguarding media content against
eavesdropping and tampering [21].
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2924 Multimedia Tools and Applications (2025) 84:2909–2946
On the hardware side, WebRTC implementations typically utilize standard device cameras
and microphones, with optional integration into specialized hardware setups depending on
the application, such as conferencing systems. For professional environments, WebRTC can
be integrated into high-end audio-visual systems that include dedicated Pan-Tilt-Zoom (PTZ)
cameras, beamforming microphones, and custom Digital Signal Processing (DSP) units to
enhance audio and video quality, especially in large conference rooms or live event streaming
setups [22].
4.2 Existing review studies
Tabl e 4demonstrates the existing review papers that touched WebRTC in the literature review.
Izima et al. [98] reviewed the machine learning techniques for predicting video quality,
including WebRTC applications. Two key findings are extracted from this study: (a) The
authors of [99] proposed a method to monitor streaming quality for WebRTC-based audio
and visual communication services. They used machine learning models to identify the
root causes of video quality problems by analysing various application-layer performance
statistics. They used decision trees, random forests, Naive Bayes, SMO, KNN, and bagging
models to classify impairments such as video blockiness and audio distortion. They found
that decision tree models with high accuracy detecting all three targets. In a similar study, the
authors of [100] evaluated WebRTC measurements on a Wi-Fi network and used machine
learning models to predict video freezing events. They found that the random forest model had
the best performance. These techniques aim to create self-healing systems that can adjust their
strategies to improve video quality. Future research could improve these techniques by using
multi-dimensional models and incorporating a broader range of audio-visual impairments
into evaluations.
There has been a lot of discussion about the potential technologies and challenges of AR
in Qiao et al., [101], where a discussion of WebRTC has been made as the communica-
tion channel for AR. The authors of Zoran et al. [102] have reviewed the media streaming
technology for live streaming events (i.e., concerts). Within that context, WebRTC has been
mentioned as one of the potential technologies since it can provide low latency and real-time
streaming.
A variety of attacks on WebRTC can compromise security and privacy. MITM attacks,
fake STUN/TURN servers, leaks of WebRTC IP addresses, and WebRTC data leaks are
common. Media and signalling data are encrypted with DTLS-SRTP and WSS to mitigate
the risk of MITM attacks. By using shared secrets or certificates, WebRTC protects against
fake STUN/TURN server attacks and authenticates TURN servers using certificate-based
authentication. WebRTC also allows users to disable WebRTC or use browser extensions
that protect data and IP addresses from leaks and uses STUN servers to determine the public
IP address. Users are encouraged to be cautious and take additional steps to protect their
privacy and security when using WebRTC despite the current secure architecture, which
provides several security features. Moreover, users can utilise VPNs to enhance their privacy
and security further by disabling WebRTC, using browser extensions that block IP and data
leaks, and using browser extensions that block IP leaks.
No review paper explores WebRTC and addresses its potential applications and signif-
icance to the best of our knowledge. Therefore, We examined the technology’s benefits,
shortcomings, and potential applications to learn about its significance in this paper.
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Multimedia Tools and Applications (2025) 84:2909–2946 2925
Table 4 Existing Review studies about WebRTC
Ref Description App. studies Pros Cons
Izma et. al., 2021 [98] A review of ML techniques for video
quality prediction
Video conferencing 2 Best ML for video quality prediction did not focus on WebRTC
Qiao et al., 2019 [101] Review advantages challenges of AR
including WebRTC
AR 1 Presented the motivation did not focus on WebRTC
Zoran et. al., 2022 [102] reviews the media streaming tech-
nologies
Streaming 3 reviews WebRTC as potential approach did not focus on WebRTC
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2926 Multimedia Tools and Applications (2025) 84:2909–2946
4.3 Advantages and challenges
4.3.1 Advantages
There are several advantages brought about by WebRTC that can be summarised as follows:
•The technology is an open-source system that can be easily integrated into other appli-
cations, contributing to its accessibility to various industries and organisations.
•Because WebRTC provides low-latency communication, it is ideal for real-time applica-
tions such as video conferencing and live streaming, enabling users to enjoy high-quality
and responsive communication experiences.
•It enables P2P communication, which reduces server dependency and enhances scalabil-
ity, making it an attractive solution for large-scale applications.
•Secure communication is ensured by WebRTC’s integrated security features, such as
encryption and certificate management.
4.3.2 Challenges
In contrast, WebRTC has some challenges such as system complexity, interoperability, secu-
rity risks, firewall constraints, ambitious requirements, and standardisation. These challenges
can be summarised as follows:
•The complexity of WebRTC’s architecture and the high level of technical expertise
required to implement it may make it difficult for users without technical expertise to
utilise it.
•The majority of browsers currently support WebRTC, including Chrome, Firefox, and
Opera. Users who rely on Internet Explorer or Safari may be unable to use it due to this
limitation. It cannot be used by users using different browsers or platforms due to the
requirement that both the sender and the receiver use compatible browsers.
•Communication via WebRTC relies on User Datagram Protocol (UDP), which may lead
to communication issues with some networks and firewalls that block or restrict UDP
packet transmissions.
•There are potential security risks associated with using WebRTC, despite the technology’s
built-in security features, such as encryption and certificate management. The privacy
and security of WebRTC users may be compromised due to WebRTC leaks, for example.
•In low-bandwidth or high-latency network conditions, WebRTC may be resource-
intensive and may not perform well. This can affect the quality of communication.
4.4 WebRTC enabled applications
In this section, we undertake a thorough examination of both existing and potential applica-
tions facilitated by WebRTC technology (See Table5). Our focus is to analyse each application
in terms of its description, problem statement, literature to date, and the gaps that remain
unexplored. This review aims to highlight the broad potential of WebRTC across various
sectors while identifying key areas where further research and development are required to
fully realize its capabilities.
4.4.1 Voice/video chatting and conferencing
Description WebRTC is commonly used for voice and video chatting and conferencing,
allowing teams to hold real-time virtual meetings with audio and video streams, under low
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Multimedia Tools and Applications (2025) 84:2909–2946 2927
Table 5 Summary of WebRTC Enabled Applications
Application Description Problem Statement Literature to Date Research Gap
Video Conferencing Real-time video and audio
communication between
users.
Ensuring high-quality,
low-latency communica-
tion across devices.
Extensive research on enhanc-
ing video and audio quality.
Interoperability with
legacy systems and
improving QoS.
Gaming Platforms Enables real-time communi-
cation and streaming in online
games.
Reducing latency for a
seamless gaming experi-
ence.
Studies focus on performance
and latency improvements.
Enhancing haptic feed-
back and standardizing
data transmission.
Telemedicine Facilitates remote patient con-
sultations and monitoring.
Secure and reliable data
transmission for health-
care applications.
Research on video consulta-
tions and remote monitoring
tools.
Integration with health-
care systems and improv-
ing security.
Live Streaming Real-time streaming of events,
classes, and performances.
Delivering low-latency,
high-quality streams to
large audiences.
Demonstrated suitability for
various live events.
Optimizing for different
network conditions and
privacy concerns.
P2P File Sharing Direct file transfer between
users without intermediaries.
Ensuring fast, secure, and
reliable data transfers.
Proof-of-concept studies on
secure file sharing.
Scalability and compati-
bility with existing file-
sharing systems.
Virtual & Augmented Reality (VR/AR) Enables immersive experi-
ences through real-time inter-
action and updates.
Low-latency communica-
tion and high bandwidth
for VR/AR applications.
Initial studies on VR con-
ferencing and interactive AR
tools.
Developing standards for
VR/AR data transmission.
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2928 Multimedia Tools and Applications (2025) 84:2909–2946
Table 5 continued
Application Description Problem Statement Literature to Date Research Gap
Online Marketplaces Enhances buyer-seller interac-
tions through real-time com-
munication.
Facilitating secure and
efficient transactions in
real-time.
Few initiatives, with focus on
enhancing user experience.
Scaling WebRTC for large
platforms and ensuring
transaction security.
Educational Platforms Supports live lectures, group
discussions, and virtual labs.
Providing low-latency,
high-quality educational
experiences online.
Explored in online classrooms
and interactive learning tools.
Further development of
educational tools and
improving scalability.
Weather Sensing Collects and communicates
real-time weather data in
urban environments.
high accuracy and timely
weather data collection
and analysis.
Limited research on urban
weather data collection using
Web RT C.
Enhancing data accuracy
and integrating with exist-
ing weather systems.
Censorship Circumvention Bypasses government restric-
tions on information access.
Secure, real-time commu-
nication without detection.
Some studies on using
WebRTC for secure, private
communication.
Addressing vulnerabilities
and improving scalability.
Machine Learning Processing Facilitates decentralized and
real-time data processing for
machine learning.
Ensuring privacy and effi-
ciency in data handling for
ML tasks.
Early-stage research on inte-
grating WebRTC with ML
processes.
Improving scalability,
standardization, and pri-
vacy measures.
Haptic - Tactile Internet Transmits touch sensations
over the internet for interactive
experiences.
Low-latency, high accu-
racy transmission of haptic
feedback.
Initial studies on teleoperation
and medical applications.
Developing advanced
hardware and standardiz-
ing haptic communication.
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Multimedia Tools and Applications (2025) 84:2909–2946 2929
latency and high-quality communication. Platforms like Google Meet and Zoom are exam-
ples of WebRTC-powered video conferencing solutions. Another use is voice chat, where
WebRTC supports applications such as Discord, often used in gaming or social networking,
allowing users to engage in seamless voice communication.
Problem statement WebRTC’s implementation for voice and video communication focuses
on enabling real-time connections between two or more users via a web browser or a mobile
app, without the need for additional software or plugins. Key challenges include ensuring
device and browser compatibility, achieving optimal network performance, and maintaining
secure communication channels. In addition, the basic functionalities of video conferencing
should be augmented with vertical tools to enable more access, control and independent
customization by users.
Literature to date WebRTC literature has focused primarily on voice and video confer-
encing, including system enhancements, performance optimisations, and user experiences.
Researchers introduced new techniques for managing network congestion [42], processing
multimedia in real-time, and P2P communication to improve the performance and scalability
of WebRTC systems [31,36]. Authentication and encryption mechanisms have been inte-
grated into WebRTC systems to enhance security, and privacy [35–37]. Further research has
been conducted to improve WebRTC user experiences, including ease of use and interface
design. The research community also explored whether WebRTC can be integrated with
existing communication systems, such as telephones, to improve interoperability and expand
its application scope. WebRTC has also been studied in emerging applications, including
virtual reality and the Internet of Things (IoT), to provide new dimensions for real-time
communication. For a deeper explanation of such potential applications, we separated the
IoT applications from this one.
Research gap Aforementioned the literature, the development of WebRTC for voice and
video chatting and conference is still in its early stages. Several research gaps need to be
addressed. One major challenge is how to interoperate with legacy communication systems
such as telephones; another issue is managing network congestion and reducing latency, so
users have a positive experience; meanwhile, protecting user data from man-in-the-middle
attacks remains an important concern [103,104]. Additionally, current technology does not
have reliable methods to ensure Quality of Service (QoS) for consistent and high-quality
audio and video experiences [24]. Multi-view (MV) learning that allows for improving gen-
eralization efficiency by learning from multiple viewpoints is one of the future optimizing
techniques being explored by the research community [105]. Optimizing Deep neural net-
works (DNN) parameters such as the number of layers and nodes, which can be leveraged to
estimate QoE and QoS of WebRTC streaming remains an open research question [24].
4.4.2 Gaming platforms
Description WebRTC technology is used in the gaming industry to enable real-time com-
munication within web browsers. Without dedicated servers, P2P communication between
players provides low latency and high-quality audio and video streams. The result is a more
immersive and interactive gaming experience. Game developers can use WebRTC technol-
ogy to enhance player interaction, including real-time voice and video communication, live
gameplay streaming, and transmitting haptic information. WebRTC technology has already
been implemented on some gaming platforms such as Twitch, with many more exploring its
potential in this sector[106,107].
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Problem statement To support gaming applications effectively, WebRTC technology must
meet several requirements. A seamless gaming experience requires low latency in real-time
communication. User experience can be negatively affected by latency or the delay between
sending and receiving information between gamers. As a result, it is essential to minimise
latency when large volumes of data are exchanged. Users must be able to enjoy an interactive
and engaging gaming experience with WebRTC as an enabler by ensuring that audio and
video are transmitted with high quality. To ensure the smooth and effective operation of
gaming platforms, WebRTC must also be able to handle high data volumes and bandwidth
requirements.
Literature to date Several studies have been conducted to assess the use of WebRTC in
online gaming, focusing on topics such as performance, latency, user experience, and secu-
rity. For example, study by Sabesan Muralikrishnan found that WebRTC provided low latency
and high performance in online gaming compared to traditional gaming solutions [59,60].
Another study by the National Institute of Standards and Technology (NIST) found that
WebRTC was suitable for low-latency multiplayer gaming, providing high-quality perfor-
mance even in network-constrained environments [61,62].
Research gap Several research gaps exist when it comes to WebRTC use in gaming. Gaming
data must be transmitted with low latency, network congestion must be efficiently managed
to minimise lag, haptic hardware and software must be developed to create a more immersive
gaming experience, and gaming data must be transmitted in a standard manner to ensure
compatibility across all platforms and devices. Despite some initiatives exploring this appli-
cation, further improvements are needed for reliable streaming with high quality. Leveraging
network traffic data and arrangements with service providers, WebRTC can further reduce
latency by utilizing application-level metrics, particularly between gamers separated by large
physical distances [108].
4.4.3 Streaming platforms
Description Live streaming is also possible using WebRTC, such as concerts, classes, and
events. Using the technology, developers can create low-latency, high-quality streaming solu-
tions. YouTube Live is widely recognised as a live streaming service that utilises WebRTC. In
addition to a seamless streaming experience, WebRTC allows users to view online broadcasts
of live events with minimal delay. Remote audiences can participate in real-time at virtual
events, facilitated by this feature.
Problem statement Real-time streaming imposes heavy load ensuring high quality video
communication on the network. As the transmitter has to encode video sequences in real-time
by deciding the pacing rate adjusting Congestion Control and Adaptive Bit rate considering
the network condition. In addition, as video streaming consumes a significant portion of
bandwidth, there is a need to allocate bandwidth more efficiently to improve overall user
experience. WebRTC is crucial for live streaming applications due to its low latency, (P2P)
network capabilities, and built-in security features. Low latency ensures seamless, near-
instant communication for a high-quality user experience. P2P network reduces server burden
and cost, allowing for a scalable solution. Built-in security protects live streaming data from
unauthorised access or tampering.
Literature to date WebRTC enables users to stream live events smoothly and with little
to no delay, with little to no delay between the live event and the broadcast [49,50,109].
Additionally, WebRTC’s P2P architecture reduces the load on server infrastructure, resulting
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Multimedia Tools and Applications (2025) 84:2909–2946 2931
in cost savings for streaming services [47,48]. It has been demonstrated that WebRTC is
suitable for streaming various applications, including live sports events, concerts, and other
large-scale gatherings. WebRTC has been demonstrated to be effective in providing high-
quality, low-latency streaming to many users, despite less-than-ideal network conditions.
Research gap Although WebRTC for streaming can potentially improve video and audio
quality and reliability, research gaps remain. In addition to addressing potential security
and privacy concerns, one gap is finding effective solutions for WebRTC streaming with
high bandwidth and low latency. In addition, WebRTC needs to be optimised for different
network conditions and improved to be compatible with existing streaming platforms and
technologies.
4.4.4 P2P video and file sharing platforms
Description File sharing is popular tool for web real-time file transfer applications. The
common methodology of implementing file-sharing is uploading contents to a file server by
a sender, then receivers can download the contents after passing access control and authentica-
tion mechanisms [110]. WebRTC enables efficient, secure, and scalable P2P file sharing with
improved speed, reliability, and robust security features like end-to-end encryption, without
the need for a middle server. When a sender initiates a transfer process, the application gen-
erates a link recipients can use to download the file over the WebRTC data channel. This
makes it a valuable technology for developers creating file-sharing apps, as demonstrated by
Facebook Messenger’s P2P file-sharing feature.
Problem statement WebRTC uses four types of servers to establish P2P connection includ-
ing web hosting, signalling servers, STUN server, and Traversal Using Relays around NAT
(TURN) server. WebRTC uses the TURN server in case the STUN server fails. However, the
study shows in [111] that 92% connections are established on STUN while 8% connections
are established using TURN. Therefore, webRTC has to cope with firewall and NAT devices
to establish a real-time P2P connection between two browsers. Efficient and fast data trans-
fer, robust security, and scalability are crucial for WebRTC in P2P file sharing. WebRTC
enables direct communication between browsers for faster transfer speeds, offers end-to-end
encryption for secure transfers, and has a decentralised architecture for scalability.
Literature to date In some existing studies, WebRTC has been examined as a potential
tool for P2P file sharing, including faster transfer speeds, increased reliability, and increased
security through end-to-end encryption [64,65]. A proof-of-concept of developing a DICOM
file exchange using WebRTC was introduced in [67]. In addition to highlighting some chal-
lenges, these studies have highlighted the need for improved scalability and compatibility
with the existing file-sharing protocols [66]. The security and reliability aspects of shared
data through a combination of blockchain and WebRTC for decentralized file control was
investigated in [112].
Research gap Optimising WebRTC for file sharing, especially regarding speed and relia-
bility, requires further research. As well as security and privacy concerns, WebRTC must be
compatible with various file-sharing devices and platforms to prove its potential. Blockchain
integration with WebRTC requires further investigation in terms of relay node selection meth-
ods and compensation mechanisms to improve the reliability and availability of relays in the
future.
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4.4.5 Virtual reality and augmented reality platforms
Description WebRTC enhances Extended Reality (XR) app experience with low-latency
real-time communication and P2P capabilities. It enables Virtual Reality (VR) users to interact
with each other in near real-time and AR users to receive real-time updates, resulting in a
more immersive experience. It also provides a scalable solution, reducing the burden on
centralised servers.
Overall, WebRTC improves the user experience in VR/AR applications.
Problem statement To achieve low latency in AR and VR applications is challenging as the
user experience is influenced if the delay is more than 200ms. The high-end supercomputers
and clusters are needed for immersive and virtual reality environments. There is still a need
to deal with timing dependencies for distributed applications running on these clusters while
implementing virtual reality environments.. WebRTC meets the essential requirements for
VR and AR applications, with low latency communication and P2P capabilities for scalability.
The technology ensures real-time interactions and updates for an immersive experience, and
its built-in security features protect sensitive information from unauthorised access.
Literature to date It appears that the key idea is to apply the concept of augmented reality
to video conferencing in general, [55,56]. There are many ways to engage and interact
with users in 360-degree VR [57]. There has been a growing interest in the potential of fully
immersive and virtual reality of 3D, and WebRTC has been mentioned as one of the techniques
used in this study [52]. As a result of the application of augmented reality in healthcare,
telehealth experiences have been enhanced [83]. This is another application considered for
online education to provide interaction using virtual reality [58]. WebRTC was also proposed
to update the position of the virtual camera to receive the Free Viewpoint Video (FVV) live
view encoded as a video. In this context, a WebRTC server is placed between a remote user
and a view renderer in the local user to enable a WebRTC connection [113].
Research gap VR and AR applications require WebRTC, which has several requirements.
These include low latency for real-time interaction, high bandwidth for transmitting large
volumes of VR/AR data, reliable data transmission that ensures a smooth user experience,
and compatibility across different VR/AR platforms and devices. Standards and protocols
for VR/AR data transmission must also be developed to ensure seamless communication
between different devices and applications. In addition, adapting the deployment and the
WebRTC Server to handle several simultaneous remote users receiving the same FVV Live
transmission remains an open research problem.
4.4.6 Online marketplaces
Description The online marketplace connects buyers and sellers online, creating a new
relationship between companies and customers giving a personalized shopping experience.
Through real-time communication, WebRTC can enhance online marketplace users’ experi-
ence. WebRTC enables real-time voice and video calls between sellers and buyers in online
marketplaces. As a result, buyers can ask questions and receive live demonstrations from
sellers, which allows them to make more informed purchases. Customers and customer ser-
vice representatives can also communicate in real-time with WebRTC. As another use-case
of how WebRTC can be used in online marketplaces, buyers and sellers can collaborate in
real-time during the transaction process. WebRTC can, for example, be used by buyers and
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Multimedia Tools and Applications (2025) 84:2909–2946 2933
sellers to digitally sign contracts, which will reduce the time and effort involved in completing
the transaction.
Problem statement More than 4 billion users are connected today to internet where 85%
of these are online and 93% of these are connected through mobile devices. E-commerce is
growing exponentially, especially with the increase in digital advertisement by companies
like Google Ads, Display Ads, Youtube Ads, and Gmail Ads [114]. To manage contents on
online marketplaces, existing approaches relied on cloud computing and content distribution
networks which require large monetary investment and operation costs. To offload some of
the distribution costs onto end users, more recent solutions used client-side software or web
browser plug-ins but studies reported poor user incentives, which limited their implementation
in the industry. WebRTC promises to provide real-time content sharing among companies
and customers, end-to-end encryption, and a convenient user experience, enhancing online
marketplaces.
Literature to date Several websites and articles have mentioned the idea of utilizing
WebRTC in the online marketplace model [115–117], but the concept has not been extensively
explored in the literature. For example, [115] proposed the use of WebRTC to enable con-
tent distribution using online marketplace scenarios in two modes: (i) client-to-coordinator
and (ii) client-to-client. the results show the proposed solution reduces the 95th percentile
bandwidth due to image content at the operator by over 75%.
By reducing the time and effort required for buyers and sellers to exchange information,
these initiatives demonstrated that WebRTC could enhance the user experience. Conse-
quently, further research is required to evaluate the significance of this application.
Research gap To improve the effectiveness of WebRTC and overall marketplace security,
additional research is needed to address its scalability, integration, user experience, and
security in online marketplaces. The objective is to ensure the technology that supports the
growing demands and enhances the user experience.
4.4.7 Healthcare monitoring systems
Description WebRTC enables virtual healthcare services like virtual consultations, remote
patient monitoring and telehealth education. It allows doctors to diagnose patients and provide
treatment recommendations through real-time virtual consultations. Moreover, Healthcare
can benefit from WebRTC, making care delivery more efficient and improving efficiency in
the system. For example, real-time medical information can be exchanged between doctors
and patients using telemedicine. As a result, patients may receive better care and travel to
medical appointments may be reduced. Furthermore, WebRTC allows healthcare providers
to communicate and coordinate more efficiently and effectively, improving patient care. By
securely sharing medical information and images, healthcare providers can collaborate more
effectively and reduce the possibility of medical mistakes. Additionally, WebRTC can also
improve the delivery of remote healthcare services, such as remote patient monitoring and
telerehabilitation. As a result, individuals living in rural or remote areas can have easier access
to healthcare services and improved quality of care for those unable to travel to a healthcare
facility.
Problem statement To ensure the success of WebRTC in telemedicine, reliable and secure
data transmission, seamless integration with healthcare systems, standardised regulation,
and sufficient connectivity and device compatibility are crucial. Healthcare providers must
address data security issues with encryption, secure servers, and firewalls while ensuring
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2934 Multimedia Tools and Applications (2025) 84:2909–2946
compatibility with existing healthcare systems. Standardisation and regulation are necessary
to guarantee the quality and reliability of telemedicine services, and adequate connectivity
and device compatibility are essential to reach patients in need.
Literature to date Telehealth video monitoring has been explored in a few references [74–
79,81]. The system should be able to connect a doctor with another doctor or patient remotely
for diagnosing and follow-ups. Similarly, another contextual health information system from
the connected medical sensors is developed. This paper reviewed the characteristics, context
and use cases of telehealth applications on the web [80]. An extension to this idea, Augmented
reality has been applied to provide more interactive features within the telehealth video
conference [83]. This can enable the patients to benefit from medical assistance by calling
remote medical support [82].
Research gap Research is still needed to unlock the potential of WebRTC in healthcare. The
focus is on the impact of technology on patient outcomes, security and privacy, integration
with healthcare systems, and cost-effectiveness. More research is required to determine how
WebRTC can improve patient outcomes, secure P2P communication, integrate seamlessly
with healthcare systems, and analyse its long-term financial impact.
4.4.8 Websites and mobile real-time analytics
Description With WebRTC, data collection and analysis can be performed in real-time for
mobile devices and websites. Users benefit from a seamless user experience, and businesses
gain valuable insights from real-time data analytics. Similarly, WebRTC can enable crowd-
sensing. A crowdsensing technology gathers real-time data on many users through their
mobile phones. Data can be collected from mobile devices and transmitted to a central server
using WebRTC, a real-time communication technology. By combining data from multiple
sources, crowdsensing applications can provide a wide range of real-time insights into various
aspects of the environment, including traffic, air quality, and weather.
Problem statement The website or mobile app must communicate with their end users in
real-time for real-time analytics. Data can be collected in real-time using WebRTC technology
so that companies can get immediate feedback and insights into their users’ behaviour.
To facilitate real-time communication, WebRTC eliminates the need for users to download
additional software or plugins and provides a seamless user experience. As a result, websites
and mobile apps have become more engaging and user-friendly.
Literature to date In real-time analytics for websites and mobile devices, WebRTC technol-
ogy has been adopted since it allows real-time communication within the web browser [85].
Another application develops a mobile-based over-the-cloud solution that takes advantage of
the many features of the YOLO algorithm [118]. Studies have been conducted on WebRTC
for real-time analytics on websites and mobile devices. WebRTC has also been shown to
enhance the user experience by providing real-time support and interaction, which leads to
higher customer satisfaction and engagement. Other studies have demonstrated that WebRTC
can enhance online marketplaces’ functionality, allowing buyers and sellers to communicate
and collaborate in real-time.
Research gap Additionally, there is still a lot of research to be done to determine whether
WebRTC is reliable and scalable for large-scale data collection, whether data security and
privacymeasures can be improved, and how WebRTC can be integrated with existing analytics
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Multimedia Tools and Applications (2025) 84:2909–2946 2935
platforms. Moreover, WebRTC-based analytics need to be visualised and communicated most
effectively.
4.4.9 Censorship circumventing
Description Censorship circumvention refers to the efforts to bypass or overcome govern-
ment restrictions or censorship of information or communication technologies. WebRTC
technology can be used as a tool to help circumvent censorship in countries where the
free flow of information and open communication is restricted. WebRTC provides secure,
real-time communication directly from browsers, making monitoring easy for censorship.
It offers end-to-end encryption for added security and privacy, making it useful for jour-
nalists, activists, and others in censored countries. However, it may still be vulnerable to
interception or blocking, and more research is needed to improve its security for censorship
circumvention.
Problem statement In censored countries, WebRTC can meet three key needs: privacy,
real-time communication, and ease of use. A user-friendly solution for fast and effective
communication is offered, which includes end-to-end encryption to ensure secure communi-
cations, real-time communication directly from a web browser, and end-to-end encryption.
Literature to date The use of WebRTC for circumventing censorship has been explored
in various studies to date. Several studies have explored how individuals and organisations
perceive and use WebRTC for censorship circumvention, and their challenges in accessing
and using it [89,90]. In these studies, it was demonstrated that WebRTC does provide a
user-friendly solution for circumventing censorship. However, there are still challenges to
be overcome, including technical expertise and compatibility with existing circumvention
tools.
Research gap There’s a need for research on WebRTC scalability, security, user experience
and compatibility for censorship circumvention. To ensure WebRTC can support increased
demands, address potential vulnerabilities, be accessible and user-friendly, and integrate
effectively with existing censorship circumvention tools.
4.4.10 Machine learning and deep learning processing
Description With WebRTC, data can be decentralised and distributed across devices and
models, enabling real-time communication and collaboration. As a result, machine learning
tasks can be performed more quickly and efficiently, while data transfer and storage costs can
also be reduced. Furthermore, WebRTC can allow machine learning to process data without
transferring it to a central server, preserving privacy. Various fields, including predictive
maintenance, image and speech recognition, and autonomous systems, can benefit from this
application of WebRTC.
Problem statement It is imperative to ensure the privacy and security of sensitive data
when using WebRTC for machine learning, to manage large amounts of data, and to achieve
real-time performance. Additionally, WebRTC devices and machine learning models must be
interoperable and standard. A simulated eavesdropping attack on WebRTC was presented in
this paper [119]. The privacy concerns within the system must also be considered in addition
to the security risk.
Literature to date There has been a growing interest in WebRTC technology’s benefits for
automating machine-learning processes in real-time. Several studies have been conducted in
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2936 Multimedia Tools and Applications (2025) 84:2909–2946
the past few years to explore the potential of WebRTC in this area [86,87]. Machine learning
algorithms benefit from WebRTC’s ability to transfer and process data in real-time, enhancing
their speed and accuracy. An analysis of WebRTC’s potential for distributed machine learning
processing can be found in [88]. Using WebRTC for distributed machine learning is an ideal
solution due to its ability to transfer data efficiently between multiple nodes in a network. In
this manner, machine learning algorithms can be trained and executed in parallel, resulting
in an increase in accuracy and a reduction in processing time.
Research gap A relatively new area of research utilising WebRTC for machine learning is
WebRTC-based machine learning. There are a limited number of studies available on this
topic. There are several research gaps in the use of WebRTC for machine learning; WebRTC
cannot handle (1) Large amounts of data due to its limited scalability; (2) Standardisation
of WebRTC implementation for machine learning is lacking; (3) WebRTC has not yet been
studied for its performance and efficiency with machine learning; and (4) security and privacy
should be considered.
4.4.11 Haptic - tactile internet
Description Haptic technology uses touch sensations and forces feedback to enhance users’
experience with digital content and devices. This type of technology can be used to improve
navigation through websites or provide more immersive gaming experiences. The haptic
internet is a technology that can transmit touch sensations over the Internet. WebRTC enables
support for haptic technologies, which allows for more interactive and immersive digital
experiences. These benefits may be valuable in gaming and telemedicine contexts, where
enhanced experiences could have positive effects.
Problem statement To fully realise the potential of WebRTC for haptic applications, it is
necessary to provide low latency, high bandwidth, improved hardware/software, and standard-
isation. A real-time, seamless haptic transmission is essential to fully exploit the potential of
WebRTC for haptic applications. To ensure immediate response and feedback, a low latency
connection and high bandwidth are required to accommodate the large volume of haptic data.
Furthermore, hardware and software technology advancements are necessary to ensure high
accuracy and precise transmission of haptic information. Establishing industry standards to
ensure interoperability between different haptic devices and applications is also imperative.
Literature to date Some studies have explored the use of WebRTC for haptic communication
in teleoperation, telemedicine, and other applications. Kenm et al. (2021) [120] investigated
the use of WebRTC for haptic communication in real-time teleoperation systems, in which
remote users can control a robot or other equipment in real-time. Teleoperation applica-
tions can benefit from WebRTC’s low latency and high reliability when transmitting haptic
information. Another study by Kurillo et al. (2016) [121] explored the use of WebRTC to
facilitate remote diagnosis and treatment of patients through real-time haptic communication.
This study found that WebRTC provided an effective way of transmitting haptic information
in real-time, enabling medical professionals to diagnose and treat patients remotely in high
accuracy. In this study, WebRTC was demonstrated to have the potential to provide haptic
feedback and communication in real-time applications, and further research is needed to
improve and refine this technology.
Research gap A full understanding of the potential of WebRTC for haptic applications will
require ongoing research. There are four main areas of focus: low latency transmission, opti-
mal bandwidth utilisation, development of advanced haptic hardware and software, and the
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Multimedia Tools and Applications (2025) 84:2909–2946 2937
standardisation of haptic information transmission. There is a need for research to minimise
latency, reduce bandwidth for large haptic data, improve haptic hardware and software, and
develop widely adopted haptic communication standards.
4.4.12 Weather sensing in urban environments
Description In an urban environment, WebRTC can collect and communicate weather data
in real-time. Weather forecasting accuracy can be improved by gaining insight into the current
weather. Forinstance, a central database can be connected to weather sensors throughout a city
using WebRTC. In addition to collecting data about temperature, humidity, and wind speed,
these sensors can also measure other important variables related to the weather. WebRTC
can transfer data from the weather station to the central database for analysis and display in
real-time.
Problem statement Weather sensing has various challenges in an urban environment when
using webRTC for data collection from various points. These include a lack of high accuracy
weather data, effective methods of processing and analysing large amounts of data, and
seamless integration with existing weather monitoring systems. These challenges must be
addressed to obtain high accuracy and timely information about local weather conditions and
formulate an interoperable and comprehensive system for weather sensing.
Literature to date A few amount of literature has been published on using WebRTC for
weather sensing in urban environments. It has been demonstrated, however, that WebRTC
technology has the potential to improve forecast accuracy and provide real-time weather
information. The use of WebRTC in an urban environment can be observed by the collection
and transmission of weather data from multiple devices and sensors [122]. UAVs were also
used in some studies for data collection [123,124]. As a result, it is possible to monitor and
analyse weather patterns and conditions in real-time, allowing for high accuracy forecasts
and early warnings.
Research gap Several gaps must be addressed to fully leverage WebRTC for weather sensing
in an urban environment: (1) WebRTC data transmission must be improved for accuracy
and reliability; (2) need to develop effective methods for processing and analysing large
quantities of weather data collected from multiple sources, (3) review the integration of
WebRTC technology with existing weather monitoring systems and networks to provide
weather sensing systems that are seamless and interoperable. As cited in [123], more research
is needed to improve the accuracy and reliability of collected weather data.
4.4.13 Educational platforms
Description Students and teachers can communicate in real-time online using WebRTC in
educational settings, enabling immersive and interactive learning. As a result of its low latency
and high-quality audio and video capabilities, it allows for live lectures, group discussions,
and remote collaboration, thus improving access to education and lowering geographical bar-
riers. Using WebRTC, educational tools and applications such as virtual labs and simulations
can be developed, further enhancing student learning.
Problem statement For WebRTC to be effective in education, audio and video transmis-
sion must be of low latency and high quality for real-time interaction and collaboration, data
transmission must be reliable to ensure a seamless learning experience, it must be compatible
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2938 Multimedia Tools and Applications (2025) 84:2909–2946
across devices and platforms to reach a broader audience, and educational tools and applica-
tions must be developed to enhance the learning experience. Protecting sensitive information
and personal data is also essential for educational usage.
Literature to date WebRTC has been used in several studies and research projects for
educational technology. Some of these studies have focused on using WebRTC for online
classrooms, distance learning, and collaboration between students and teachers [92,125–
129]. A recent study noted that WebRTC offers the advantage of using limited internet
connections due to its low bandwidth requirements [126]. In another study, Augmented
reality (AR) was used to provide interactivity in the process [58]. Among the study’s findings
was that WebRTC provided a high level of service and could support real-time multimedia
communication between students and teachers in real-time [127,130]. Studies show that
WebRTC can be used for educational technology, providing a flexible and effective way for
students and teachers to collaborate and communicate in real-time. Despite this, additional
research is required to fully understand the potential and limitations of WebRTC in the context
of education. Similarly, videoconferencing for a mathematics class in which the whiteboard
can be shown is considered [131].
Research gap WebRTC’s potential for education needs to be realised by addressing several
research gaps: For effective online collaboration and interaction between students and teach-
ers, low latency is critical for reducing latency in real-time communication. Despite many
studies about this, these studies are not mature because of the need for reliability, stability,
security and privacy. Another research gap concerns the development of educational tools and
applications that use WebRTC technology effectively. These include virtual and augmented
reality simulations and collaborative online learning environments.
5 Open research directions and limitations
5.1 Open research directions
In addition to the addressed research gaps for each application, as mentioned previously,
some general open research shall be considered for enhancing WebRTC. These challenges
include:
•Improved Quality of Service (QoS): WebRTC currently relies on the underlying net-
work conditions to determine the quality of the call. To improve QoS, researchers
are developing algorithms like Adaptive Bitrate (ABR) and Forward Error Correction
(FEC)[132–136]. ABR adjusts the video quality based on available bandwidth, ensur-
ing smoother streaming [137]. FEC adds redundant data to help recover lost packets,
improving call reliability. For example, the ABR algorithm can dynamically adjust video
resolution to maintain a stable experience even if network conditions fluctuate.
•Network Adaptation: WebRTC is currently limited to working on a single network, but
struggles with seamless operation across different networks such as 3G, 4G, and WiFi.
Researchers are working on algorithms and protocols that can handle transitions between
networks without interrupting the communication.
•Security and Privacy: As WebRTC becomes more widely adopted, there is a growing
need to ensure that the technology is secure and that users’ privacy is protected. Algo-
rithms for end-to-end encryption, such as SRTP, are used to encrypt media streams.
Additionally, secure key exchange mechanisms like Diffie-Hellman are implemented to
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Multimedia Tools and Applications (2025) 84:2909–2946 2939
protect encryption keys. For example, SRTP encrypts audio and video streams, while
Diffie-Hellman ensures that encryption keys are securely exchanged between users.
•Interoperability: WebRTC primarily works with other WebRTC clients, but making it
compatible with other communication technologies like SIP and Extensible Messaging
and Presence Protocol (XMPP) is a focus area. Researchers are developing bridging
solutions and gateways that convert WebRTC signals to formats compatible with these
protocols [138].
•Machine Learning: WebRTC is increasingly used in XR applications where machine
learning algorithms can enhance the user experience [24]. For example, machine learning
can be used to improve video quality through real-time image processing or to predict
and reduce latency by analysing network patterns. Algorithms like Convolutional Neural
Networks (CNNs) can enhance video clarity and reduce artifacts during streaming.
•WebRTC on IoT: WebRTC hasthe potential to provide a low-latency, secure, and reliable
communication channel between IoT devices and servers. Researchers are exploring
how WebRTC can be used to transmit real-time data from sensors and control devices
efficiently. For instance, WebRTC could enable real-time monitoring and control of smart
home devices with minimal delay.
•Edge Computing: With the increasing adoption of WebRTC, edge computing is becom-
ing more important in reducing latency and bandwidth consumption. Edge computing
involves processing data closer to the source rather than in a central data centre [139].
This can reduce latency and improve performance for WebRTC applications. For exam-
ple, deploying WebRTC servers on edge nodes can decrease the time it takes for data to
travel between users and servers.
•Adaptation of non-video data: To make WebRTC architecture capable of accepting non-
video data, improving its efficiency and reliability is necessary. To facilitate the transfer
of non-video data over WebRTC, modifying the underlying protocol and network archi-
tecture may be necessary. New tools and technologies may also be developed to facilitate
this data transfer. To ensure seamless integration with existing data transfer solutions,
it is also necessary to ensure compatibility with existing data transfer technologies and
systems.
5.2 Study limitations
While this review provides a comprehensive overview of WebRTC’s capabilities and applica-
tions, several limitations must be acknowledged. First, the scope of the review was constrained
by the exclusion of non-English studies and publications that did not provide explicit expla-
nations of WebRTC’s use. This may have led to the omission of significant contributions
from non-English-speaking research communities or studies addressing relevant applica-
tions outside the reviewed context. Furthermore, although this study analysed numerous
applications of WebRTC, there remains a lack of large-scale, real-world deployment data to
validate the findings of many reviewed papers, particularly in emerging areas such as virtual
reality and telemedicine. Furthermore, while the systematic review methodology was robust,
the rapidly evolving nature of WebRTC and its integration into various technologies means
that the findings may quickly become outdated as new advancements emerge. Finally, the
study primarily focused on the technical aspects of WebRTC and did not extensively explore
the socio-economic or legal challenges surrounding its widespread adoption, which could
provide important insights into its future development.
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2940 Multimedia Tools and Applications (2025) 84:2909–2946
6 Conclusions
This review highlights its versatile capabilities and potential for real-time communication
applications beyond traditional video conferencing. Through an extensive evaluation of 83
studies, it was demonstrated that WebRTC has established itself as a crucial tool across
various sectors, including telemedicine, online gaming, education, and live streaming. The
low-latency, peer-to-peer communication feature, combined with robust security protocols,
provides unique advantages for high-quality, real-time data transmission. However, despite its
adoption across numerous industries, several critical areas of improvement remain. This study
identifies major research gaps, such as the need for improved scalability, enhanced security,
and better handling of non-video data. Furthermore, WebRTC’s integration with emerging
technologies, like machine learning and edge computing, presents new opportunities for its
advancement. Ultimately, the paper concludes that WebRTC is positioned to significantly
influence future developments in real-time communication applications, but there is a need
for focused research to address its current limitations and explore new applications.
List of Abbreviations WebRTC: Web Real-time Communication SIP: Session Initiation Protocol NATS: Net-
work Address Translation Traversal ICE: Interactive Connectivity Establishment STUN: Session Traversal
Utilities for NAT TURN: Traversal Using Relay around NAT SRTP: Secure Real-time Transport Protocol
DTLS: Datagram Transport Layer Security UDP: User Datagram Protocol QoS: Quality of Service FEC:
Forward Error Correction IoT: Internet of Things XR: Extended Reality VR: Virtual Reality AR: Augmented
Reality ABR: Adaptive Bitrate P2P: Peer-to-Peer W3C: World Wide Web Consortium IETF: The Internet
Engineering Task Force XMPP: Extensible Messaging and Presence Protocol CNN: Convolutional Neural
Networks
Data Availability The data is available upon request.
Open Access This article is licensed under a Creative Commons Attribution 4.0 International License, which
permits use, sharing, adaptation, distribution and reproduction in any medium or format, as long as you give
appropriate credit to the original author(s) and the source, provide a link to the Creative Commons licence,
and indicate if changes were made. The images or other third party material in this article are included in the
article’s Creative Commons licence, unless indicated otherwise in a credit line to the material. If material is
not included in the article’s Creative Commons licence and your intended use is not permitted by statutory
regulation or exceeds the permitted use, you will need to obtain permission directly from the copyright holder.
To view a copy of this licence, visit http://creativecommons.org/licenses/by/4.0/.
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