International Journal of Innovative Technology and Exploring Engineering (IJITEE)

ISSN: 2278-3075, Volume-8, Issue-9S, July 2019

Performance Analysis of WebRTC and SIP for

Video Conferencing
Navrattan Parmar and Virender Ranga

 section 6, which is followed by Conclusion and Future Work

in Section 7.
Abstract:With the advancement in communication and
development of technologies like VoIP and Video Conferencing,
Web Real-Time Communication (WebRTC) is developed to II. INTRODUCTION OF PROTOCOLS
communicate without plugins and stream the videos on a real
time. It was initially developed by Web Consortium(W3C) and Previously, we needed external support for implementation of
Internet Engineering Task Force (IETF). It allows to transfer media transfer using IP. Protocols like WebRTC and SIP
videos and audios between different browsers. This research make our task easier by using multiple standards, protocols
paper, analyse the parameters during the call in different and rich APIs. We firstly discuss the architecture of these
browsers and conditions (number of end points). The concept of protocols i.e. SIP and WebRTC which proceeds as below.
WebRTC is inspired from Session Initiation Protocol(SIP). It
helps in the establishment of sessions and maintain it. It also
supports data and message transmissions. It also works on remote
location and different network transmission protocols. It also
allows peer to peer communication. In this research work, we
examine the behaviour of WebRTC and SIP during the call from
different browsers. We examine the different parameters like
packets sent, jitter, VO-Width and bandwidth during the call and
call supported on cloud during our experimental work.
Index Terms: WebRTC , SIP , SDP , UDP , Codec , VoIP , Session
Management , Internet Engineering Task Force (IETF), TLS,
Channel Bitrate, Inter-Process Communication (IPC).

I. INTRODUCTION
WebRTC is a project that was started by google. It is a
collection of frame-work and libraries. It is an open source Fig.1 SIP Network Components
and provides real-time communication between various web
browsers and mobile applications. It uses simple application
programming interfaces (APIs). It permits audio and video A. Session Initiation Protocol
communication (VoIP). It does not use any third party Session Initiation Protocol (SIP) is a signaling protocol for
software or plug-in. The session Initiation Protocol (SIP) is a video conferencing, VoIP, multiplayer games and real
signalling protocol. It is used for establishment, invite, time messaging application. It lays set of rules how two
maintaining, and terminating of media sessions between user systems communicate with one another by initiating
agents or the end points. SIP is an application layer protocol sessions. It is rich repository of methods defined for
used for delivery of voice and multimedia of internet different locations end points using different media
telephony for voice and video calls, over the networks. interactions and capabilities. SIP is loose coupled and
The main aim of this research work is to analyse the interactive protocol. It is developed and being looked
performance of these two frameworks in various scenarios. upon by an IETF, which is society for developing Internet
The flow of the paper is organized as follows: Section 2 shows standards. It works on application layer and is also
the brief introduction of the proposed protocols, Section 3 independent of network layers protocol. SIP network
describes Related Work, and Section 4 describes motivation architecture can be understood as in Fig. 1. End-user may
of research work. In section 5, Experimental Setup is shown be client or server. It interacts with the proxy server, which
with the help of different scenarios. Results are discussed in functions similar to the router and forwards request to
registrar server. Proxy server can be stateful or stateless
having information of the network and other without any
network information. Registrar server authenticates the
Revised Manuscript Received on June 15, 2019. end user and responds 200 Ok for success after which
Navrattan Parmar, parmarnavrattan@[Link], Department of
Computer Engineering, National Institute of Technology, Kurukshetra, request goes to Location server as in Fig. 2.
Haryana.
Virender Ranga, [Link]@[Link], Department of Computer
Engineering, National Institute of Technology, Kurukshetra, Haryana.

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DOI: 10.35940/ijitee.I1109.0789S19 679 & Sciences Publication
 Performance Analysis of WebRTC and SIP for Video Conferencing

SIP calling using SSIP is illustrated in Fig. 4, it can be seen

SIP number are similar to DNS as explained.

B. WebRTC

WebRTC is an open source technology that allows

peer-to-peer connection to different web applications. It is
used for Real Time Communication (RTC) for conferencing,
social networking, online medical consultations, live games
etc. It provides RTCDataChannel, RTCPeerConnection, and
MediaStream objects for the development. WebRTC
architecture is defined as in Fig. 5.
Fig. 1 Registration in SIP

location server replies 3xx (redirect response) from the  RTCPeerConnection: It provides connection to
location database created by the registrar server and peers and attach audio/video's
session is established for communication. this whole streams(onAddStream event). It use Java-Script
process is depicted in Fig. 3. Session Establishment Protocol (JSEP) for
connection establishment. Secure Real-Time
Transport Protocol(SRTP) is used for the making the
media delivery secure and safe. It also provides
Secure Real-Time Control Transport
Protocol(SRTCP) for reliability[22].

 MediaStream(getUserMedia api): It request for the

media stream and adds it to RTCPeerConnection. it
establishes the connection using the signaling server
firstly and there after configures for direct
communication.
Fig. 2 SIP Session Establishment

SIP performs various function on session like Call drop, Hold,  RTCDataChannel: It is used to send data along with
Video Conferencing, moderator meetings (controlled by the media streams. It uses Stream Control
leader), moderator-less. SIP allows Peer-to-Peer(P2P) as well
Client-Server communication in a network. In Client-Server
communication, unlike the P2P end-users any can have FIG. 4 SIP CALL
different capabilities. In P2P end users can swap their roles
and if one is unavailable other can communicate. However,
this allows different users to connect and disconnect during
the call multiple times. SIP have different response codes
which are important to remember for communication which
can categorized as follows:

 1** is class of response for End to End responses

codes like 100 TRYING.
 2** is class of response are for the response
accepted.
 3** It is a redirect response class by redirect server
in reply to INVITE.
 4** class of response are for Client failure similar Transmission Protocol (SCTP) for secured data transfer.
to HTTP.
It adds TCP like features like multiplexing, flow control
 5** It is the response used for Server Failure.
 6** It is class for Global failure Response. Server and reliability[23].
denies such request to be forwarded as it will fail
at other location too.

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 International Journal of Innovative Technology and Exploring Engineering (IJITEE)
ISSN: 2278-3075, Volume-8, Issue-9S, July 2019

Cağatay Vildiz et. al[6] show the Real Time Sip Network
Simulation and Monitoring System. In Simulation System is
installed at a SIP server, contribute services for assembling
network data and server statistics. It also gives a framework
for developing SIP Network Applications.

Jin Zhou et. al[7] propose an approach of automated SIP

network discovery based on message probing. This approach
gives minimal number of SIP request messages that will be
sent out to probe SIP entities. It could then wrap and parse the
received responses and required information could be
obtained. while costs have been limited probing messages,
their results show that the approach can effectively obtain sip
information for management.

Fig. 3 WebRTC Architecture Gao Zhiguo et. al[8] presents SIP accelerator (SIP Offload
Engine (SOE)) to enhance server performance. It helps
From the End User Media Stream is split into codecs like offloading parsing, processing, security and transport.
H.264, ISAC, OPUS and VP8. Different browsers may use Research results shows improvement in the SIP Server
different codecs for audio and video. WebRTC transport use throughput.
UDP because loss of packets is not that significant. We can
still maintain a video quality even after the packets loss to a Abhishek Bansal et. al[9] focuses on DOS attack by SIP
certain value. It also establishes and maintains the sessions. messages and analyze server performance . Calculates CPU
Utilization and memory usage during VoIP calls by using the
III. BACKGROUND STUDY performance metric. Results shows when SIP server is
overloaded by call requests so that quality of call degraded.
Tomokatsu Mizukusa et. al [2] propose an environment for
SIP products based on Feed-Forward design. Its environment Victoria Beltran [Link][10] focuses on different IDM Models.
gives performance value that includes Signal Processing and This research is on cloud-based services. It manages user
Thermal Dispersion. This research results in reduction of utilities in Unified Communication as Service(UCaaS). This
design period. research evaluates requirements on WebRTC-based UC
services and propose modifications of WebRTC to meet the
Dirk Hoffstadt et. al[3] gave the architecture, features and requirements on IDM. Analysis of different models to identify
usage of a sip trace recorder(str) used to parse and store information storage. It accelerates the information recast from
important sip data in database. str plug-ins provides threat the enterprise to the cloud.
analyses also with privacy option.
Alfonso Sandoval Rosas et. al[11] define the meeting(video
Helmut Hlavacs et. al[4] describe an approach Babel-SIP session), i.e. system oriented and gives users higher
for increasing the rate of acceptance for SIP messages and understandability to communicate by conve standard
concludes that, Babel-SIP can exceedingly enhance the telephony and multimedia in real-time on web browsers. This
message acceptance Rate. The main motto of Babel-SIP is to research proposes a cooperative interaction scheme that
act as intermediary for a SIP proxy and analyzes its messages differs from others in the context of telephone network
sent to its proxy. It studies which register messages were communication straight from a web browser during an active
accepted by this proxy, and the ones rejected. video conference, which also enables Real-Time Media
Streams between these two techniques to be exchanged.

Demir Y. Yavas et. al[5] present a Fluid-Flow model to

analyze the Priority-Based Request Scheduling Mechanism Basar Daldal et. al[12] propose an approach to make
(PRSM) in the overloaded SIP server having an infinite communication between
buffer. The model justify the performance of the PRSM WebRTC-to-WebRTC, and
Fluid-Flow Model. WebRTC-to-legacy VoIP by

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DOI: 10.35940/ijitee.I1109.0789S19 681 & Sciences Publication
 Performance Analysis of WebRTC and SIP for Video Conferencing

the use of Restful Web Wervices. Its approach is to carry "a  It will useful for Testing and Automation for SIP and
pointer" that contains a value to define a Restful Service link WebRTC protocols on cloud.
to actual media description of endpoints instead of media
description itself.  This research provides a lot of useful observation's
on QOS parameters.
Ahmadreza Montazerolghaem et. al[13] paper implements
OpenSIP on a real testbed which includes open VSwitch and  It will be boon to the student's and researchers as it
the floodlight controller. The results show that the proposed will give them a direction and quick start in world
architecture has a low overhead and satisfactory performance of IoT. They will understand the architecture and
and also take benefit of a flexible scale-out design during QOS parameters for SIP and WebRTC.
application deployment.

 This work will not only help in understanding, but

katrien De Moor et. al[14] conducts an experiment with 22 also make the us aware of broad scope of
observations where 2-party WebRTC based audiovisual WebRTC.
conversations took place under varying technical conditions.
It is observed that the data and unity of quality of experience
 WebRTC and SIP are having high cohesion and low
of WebRTC is affected by the annoyance leads to lowest
coupling. It makes the development easy. There
quality.
are tremendous number of classes available we just
need to implement them. These protocols are Open
Huaying Xue et. al[15] paper propose a framework based on source and also reliable, so development is trusted.
WebRTC, video conferencing system with augment features
of screen sharing. It explains architecture with its
 It is possible for an individual to design a video
components. This approach gives a premium Quality video
calling application on top of WebRTC.
even in the low bandwidth networks and ensures wonderful
user experience.
V. EXPERIMENTAL SETUP

Alexandre Gouaillard et. al[16] provide a thorough Here we analyze, the call through WebRTC[17][18] and
overview of the various Testing Problems encountered when SIP. Our proposed approach uses different browsers,
the WebRTC was first released. Testing is carried out on the different number of end points at different locations as in
shown in Fig.6, Call through WebRTC. It has been deduced
grounds of compliance with W3C. It tests [Link] API,
that the capacity of the channel is directly proportional to
Stand-Alone Web Application Testing, WebRTC Safety the power of the signal when the bandwidth remains
testing, P2P Network and ICE testing, Synchronous and constant. Chrome browser use codec vp-8 for video and
Asymmetric Testing and complete worldwide interoperability opus codec for audio. Similarly different browsers use
Testing. different codec for call quality[19][20]. It may be noted
different versions of same browsers can also use different
codecs.
IV. MOTIVATION
A. Scenario 1

 [Link] et. al[1] propose a detailed study of

We join the call using two end points at remote location via
various issues in IoT like scalability,
SIP[21] protocol. This experiment use chrome and Firefox
interoperability, heterogeneity, quality of service
web-browser. In this Scenario, Call did not connect only one
and security. the proposed research work focuses
end point was there. We analyse the value of different
on interoperability, and QOS.
parameter and found bitrate is quite low, close to [Link]
Bitrate and Video Bitrate decreases to zero(since,failure).

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 International Journal of Innovative Technology and Exploring Engineering (IJITEE)
ISSN: 2278-3075, Volume-8, Issue-9S, July 2019

Fig. 4 Call through WebRTC

B. Scenario 2

We join the call using two end points at local location via SIP
protocol. We use chrome and firefox web-browser. In this
scenario call connects . We can show it in Fig.7, Fig.8, Fig. 9,
and Fig. 10.

Fig.8 Audio in Scenario-2 &

Fig. 7. End-Point in Scenario-2 Fig.9 Video in Scenario-2

I. Audio Bitrate and Video Bitrate remains constant.

II. VI Width =1280 and VO Width=1200 means quality
is 720p.
III. RTT is 275 for video and 247 for audio. No Network
Loss.
IV. No Network loss.
V. Call Quality is constant.
Fig.10 Channel Rate in Scenario-2
VI. Channel Rate is 92000.

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DOI: 10.35940/ijitee.I1109.0789S19 683 & Sciences Publication
 Performance Analysis of WebRTC and SIP for Video Conferencing

Fig.11 Audio in Scenario-3

D. Scenario- 4
C. Scenario-3
We join the call using two end points at local location via
We join the call using 22 end points at local location via SIP WebRTC[22][23] protocol. We use chrome and firefox
protocol. We use chrome and Firefox web-browser. In this web-browser. In this scenario call connects to both end-points
scenario call connects to all . We observe as in Fig.11, Fig.12, .We observe the results as in Fig.14.
Fig.13. I. Audio Bit-rate and Video Bit-rate remains constant
I. Audio Bitrate and Video Bitrate remains constant. II. VI Width =1920 and VO Width=1920 means quality is
II. VI Width =1920 and VO Width=1280 means quality is 1080p(HD) for both send and receive.
1080p(HD). III. RTT is 267 (average).
III. RTT is 269 for video and 244 for audio(average). IV. NO Network Loss.
IV. NO Network Loss. V. Call Quality is constant and is equal to call quality
V. Call Quality is constant. received.

Fig.12 An End-point in Scenario-3

Fig.14 Call Quality in WebRTC in Scenario-4

E. Scenario-5
We join the call using two end points at remote location via
SIP protocol and WebRTC protocol. We use chrome and
Firefox web-browser. In this scenario call connects. At the
starting of the call, video quality is low and audio remained
almost constant during the call[24][25]. We observe
following observations in Fig.15, Fig.16 and Fig.17.
I. Audio Bitrate and Video Bitrate remains constant
II. VI Width =1200 and VO Width=1200 means quality
Fig.13 Video in Scenario-3
is 720p. for both
send and receive.
III. At SIP, AO Round
Trip Delay is

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 International Journal of Innovative Technology and Exploring Engineering (IJITEE)
ISSN: 2278-3075, Volume-8, Issue-9S, July 2019

33.94ms, VO Round Trip Delay is 50.61 and round VI. RESULT DISCUSSION
trip delay average is 46.83. After the repeated observations, some conclusions may be
IV. At webRTC, AO Round Trip Delay is 30.67ms, VO drawn:
Round Trip Delay is 20.75 and round trip delay  There is no significant network loss (AI,AO,VI,VO or
average is 28.74. any other) or error loss. So, it not a major problem be
V. No Network Loss. looked in.
VI. Call Quality is constant and is equal to call quality
 Even though some packets are dropped, but the call
received.
quality remains almost constant. We can use UDP as
transport protocol in Real Time
Communication(RTC's).
 Call quality is better in WebRTC protocol on the same
network conditions. RTT is independent on Bitrate.
 There is little or no difference is calling from remote
location than local locations.
 Bandwidth effects the call quality.
 Channel rate is directly proportional to SNR, thus
channel rate is dependent on Power of Signal and
Bandwidth.
 Latency is low which is the advantage and why we use
these protocols for RTC.

VII. CONCLUSION

I. There are several other protocols like DDS which

can built on top of UDP and can provide RTC. In the
future, we will understand its architecture and
Fig.15 SIP end Point in Scenario-5
implement for different scenarios.

II. WebRTC is being developed by open source

community and it can be extended support to
android and even mobile browsers.

III. As we observed, there is a difference in RTT in

audio stream and video stream it can be combined in
a single stream.

Fig. 16 WebRTC end point in Scenario-5 diag-1

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AUTHORS PROFILE
[Link], A., Moghaddam,M.H.Y., &
Leon-Garcia,A.(2018), OpenSIP: Toward Software-Defined SIP Navrattan Pamar is currently pursuing [Link],
Networking , in IEEE Transactions on Network and Service Computer Engineering from National Institute of
Management, vol. 15,no.1, 184-199. Technology, Kurukshetra, Haryana, India. He has
graduated from Guru Nanak Dev University, Amritsar
in 2017. He has been published one book chapter in
[Link],K.D., Arndt,S. & Ammar,D.(2017), Exploring diverse Springer Lecture Notes in Networks and Systems,
measures for evaluating QoE in the context of WebRTC, Ninth (Scopus Indexed). He is currently looking forward in research in protocols
International Conference on Quality of Multimedia Experience and currently working as an Intern in a software company in Bangalore.
(QoMEX), Erfurt,, 1-3.

15. Xue,H. & Zhang,Y. (2016) A WebRTC-Based Video Conferencing

System with Screen Sharing, 2nd IEEE International Conference on
Computer and Communications (ICCC), Chengdu, 485-489.
Virender Ranga received his PhD degree in 2016
from Computer Engineering Department of National
16. Gouaillard,A. & Roux,L.(2017), Real-Time Communication Institute of Technology, Kurukshetra, Haryana, India.
Testing Evolution with WebRTC 1.0, Principles, Systems and He has published more than 50 research papers in
Applications of IP Telecommunications (IPTComm), Chicago, IL, 1-8. various International SCI Journals and reputed International Conferences in
the area of Computer Communications. Presently, he is Assistant Professor
in the Computer Engineering Department since 2008. He has been conferred
17. Jian.C. & Lin,Z.(2015), Research and Implementation of WebRTC by Young Faculty Award in 2016 for his excellent contributions in the field
Signaling via WebSocket-based for Real-time Multimedia of Computer Communications. He has been acted as member of TPC in
Communications, ,5th International Conference on Computer Sciences various International conferences of repute. He is a member of editorial
and Automation Engineering ICCSAE, 374-380. board various reputed journals like Journal of Applied Computer Science &
Artificial Intelligence, International Journal of Advances in Computer
Science and Information Technology(IJACSIT), Circulation in Computer
18. Edan,N.M., Al-Sherbaz,A. & Turner,S.(2017), Design and Science (CCS), International Journal of Bio Based and Modern Engineering
Evaluation of Browser-to-Browser Video Conferencing in (IJBBME) and International Journal of Wireless Networks and Broadband
WebRTC",Global Information Infrastructure and Networking Technologies. He is an active reviewer of many reputed journals of IEEE,
Symposium (GIIS), St. Pierre, 75-78. Springer, Elsevier, Taylor & Francis, Wiley and InderScience. His research
area includes Wireless Sensor & Ad-hoc Networks, SDN, IoT security,
FANET security.
19. Haensge,K., & Maruschke,M.(2015), QoS-based WebRTC Access
to an EPS Network Infrastructure", 18th International Conference on
Intelligence in Next Generation Networks, Paris, 9-15.

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