We started in winters on 2012 with Webrtc . At time time it just looked like a new tech jargon that might fade away when new ones comes . In many many WebRTC’s buzz has died down since its massive adoption. But i nevertheless still see a lot of potential and development around it.
What really is WebRTC ? I made an entry on it here .
Around nov – dec 2012 , team and I spend the time learning the nitty-grities of HTML5 based media operation and Javascript sip stack of SIPML. I remember toward the end of the year ie before Christmas , We were done with the explanation and education aspects of WebRTC , a technology that will revolutionise communication in ages to come , at-least so says the numerous other blogs , and documents i read so far .
Usecases for WebRTC range across a wide variety , of them the most revenue generating ones are around video conferencing with realtime HD audio-video-data streams ,
To bridge the flow between a webrtc client to a PSTN endpoint via IMS , interworking between webrtc media standards and codecs with that of gateways in IMS is critical . For instance WebRTC mandates secure RTP ( SRTP) the media engine / gateway should be able to support and connect with RTP from PSTN endpoints.
client BOB -> webrtc2sip Gateway -> SIP server -> client Alice
can be understood with the callflow of a simple SIP Invite initiated from one html page towards another which passes through the configuration of gateway to IMS world , SIP Telecom Application server , Database , nodes of IMS environment etc.
For the purpose of a simple Explanation a simplified call flow ca be depicted as ,
A very high level architecture of solution deployment in IMS world could be
As the solution matures into a full fleshed project . The alpha version has been released with the following feature set . The WebRTC platform Suite offers a easily deploy-able solution to enable communication
Alpha Release WebRTC platform Suite
Single Sign On
Login with id and password to access all services
Audio / Video Call
Call Hold / Call Transfer
Messaging:
SIP Instant Messaging
Message to Facebook Messenger
Message delivered as Email
Chatroom
group chat between multiple users . Room is created for set of users .
Video Conferencing
video chat between multiple parties . Room is created for set of users .
File Transfer
Sharing of files from local to remote , in peer-to-peer and broadcasting fashion .
Third party Webservices
Widgets like calendar , weather , stocks , twitter are embedded.
Visual Voice Mail
Record and deliver voice message to recipients voice mail inbox which can be accessed/ played from web client .
Phonebook
cloud integration
add new entries
add photos to contacts identity
import contacts from google account
Click to Call :
Drop down list of contacts form mail call console
2 step Click to call from Phonebook
Presence :
Publish online / offline status
Use Subscribe / notify requests of SIP
Web Ssocket to SIP Gateway
Conversion between the signal coming from the WebRTC and SIP client to the IMS core
Conversion of “voice/video ” media between sRTP and RTP
Conversion of other media (data channel) towards MSRP and Transcoding.
Support of ICE procedure
Implementation of a STUN server
QoS Support
Beta Release WEBRTC PLATFORM SUITE
Logs
calls logs
Message logs
User Profile
user details like address , email and social networking accounts
Phonenumber for GSM integration through SMS
User’s Media storage like Pictures , profile picture , Audio , video
File sharing documents storage for future access in the same format
Real Time and Offline Analytics
service usage with graphical and tabular history trends
Session Management
Single Sign-on
Forgot password regeneration using secure question
Registration of new user account
Logout and clearance of session parameters
Security
No redirection to any page through url entry without valid session
No going back to home page after logout by back button on browser
No data vulnerability
Multiple login through different devices handled
OAuth
Login via IMAP / token through facebook and Google
Phonebook with Presence functionality inbuilt
Directory Service based on country / region
Geolocation of approximate location detection of device logged in and visibility to others
WebRTC client deployment view , accessible devices , network elementsWebRTC deploymenet overview and inetraction with other network elemets such as gateway , cloud storage , sipserver , IMS
Commercial release features specs for WebRTC over IMS
Integration with new age CSP deployments like VoLTE, ViLTE, VoWiFi
MOS , R-factor ( derived from latency , jitter , packet loss )
CDR (Call detail records ) and accounting
Lawful interception
Updating this article 2019
There was a long journey from traditional telecom architectures to NFV cloud based architectures ( like openstack). supported over web , 4G , LTE or other upcoming networks. Many OTT providers prefer using the public cloud over a NFV data centre.
Multinode / Multiedge computing platforms like Media Resource Function are expected to meet the need for quick delivery with additional features like hardware accelerated media , algorithms for optimised data flow (packetization, decongesting , security ) etc . With th decomposed architecture they can better utilise the
CPU – contains couple of cores optimised for sequential serial processing such as graphics or video processing
GPU – contains many smaller cores to accelerate creation of images for computer display . Can include texture mapping, image rotation, translation, shading or more enhanced features like motion compensation, calculation of inverse DCT, etc. for accelerated video decoding.
DSP- processing data representing analog signals
Although IMS based solutions are more suited to telephony applications and CSPs ( Communication service providers like telecom companies ) but similar or same architectures are widely finding their into newer developed cloud communications solutions supporting tens of millions of subscribers and hyper scale deployment . It could be around applications such as
Demand these says is for a decentralised system of pool of servers ( media and signalling ) that can scale independently to match up to peak traffic at any moment , with ofcourse carrier class performance . Not only these flexible solutions reduce complexity but also OpEX .
Unified Communicator and Collaborator for Enterprise
Modular enterprise communicator solution for enterprise based communication and collaboration . Use sipml5 client side library to provide webRTC based media stream capture and propagation from client side without external plugins.
WebRTC (Web Real-Time Communication) is an API definition drafted by the World Wide Web Consortium (W3C) that supports browser-to-browser applications for voice calling, video chat, and P2P file sharing without the need for either internal or external plugins.
Enables browser to browser media streaming over secure RTP profile
Standardization, on an API level at the W3C and at the protocol level at the IETF.
Enables web browsers with Real-Time Communications (RTC) capabilities
written in c++ and javascript
BSD style license
free, open project available in all major browsers
VideoEngine is a framework video media chain for video, from camera to the network, and from network to the screen.
VP8 : Video codec from the WebM Project. Designed for low latency Real time Comm.
Video Jitter Buffer: conceal the effects of jitter and packet loss on overall video quality.
Image enhancements : removes video noise
Transport
Transport / Session Layer of WebRTC stack provide Session Management for WebRTC media streams .
It consists of network stack for Secure RTP, the Real Time Protocol.
STUN/ICE for NAT , Network Address Traversal across various types of networks.
Session Management which is an abstracted session layer for call setup.
Standardization by IETF and W3C
As of the 2019 update the W3C defines it as
a set of ECMAScript APIs in WebIDL to allow media to be sent to and received from another browser or device implementing the appropriate set of real-time protocols. The specification being developed in conjunction with a protocol specification developed by the IETF RTCWEB group and an API specification to get access to local media devices.
W3C contribution to WebRTC standardization
Media Stream Functions : API for connecting processing functions to media devices and network connections, including media manipulation functions.
Audio Stream Functions : An extension of the Media Stream Functions to process audio streams (e.g. automatic gain control, mute functions and echo cancellation).
Video Stream Functions : An extension of the Media Stream Functions to process video streams (e.g. bandwidth limiting, image manipulation or “video mute“).
Functional Component : API to query presence of WebRTC components in an implementation, instantiate them and connect them to media streams.
P2P Connection Functions : API functions to support establishing signalling protocol-agnostic peer-to-peer connections between Web browsers
API specification Availability
WebRTC 1.0: Real-time Communication Between Browsers – Draft 3 June 2013 available
Implementation Library: WebRTC Native APIs
Media Capture and Streams – Draft 16 May 2013
Supported by Chrome , Firefox, Opera in desktop of all OS ( Linux, Windows , Mac )
Supported by Chrome , Firefox in Mobile browsers ( android )
IETF contribution to to WebRTC standardization
Communication model
Security model
Firewall and NAT traversal
Media functions
Functionality such as media codecs, security algorithms, etc.,
Media formats
Transport of non media data between clients
Input to W3C for APIs development
Interworking with legacy VoIP equipment
Open and Free Codecs
Codecs signifies the media stream’s compession and decompression. For peers to have suceesfull excchange of media, they need a common set of codecs to agree upon for the session . The list codecs are sent between each other as part of offeer and answer or SDP in SIP.
WebRTC uses bare MediaStreamTrack objects for each track being shared from one peer to another. Codecs associated in those tracks is not mandated by webrtc soecification.
For video as per RFC 7742 WebRTC Video Processing and Codec Requirements , the manadatory codesc to be supported by webrtc clients are : VP8 and H.264‘s Constrained Baseline profile.
For Audio as per RFC 7874 WebRTC Audio Codec and Processing Requirements, browser must support Opus codec as well as G.711‘s PCMA and PCMU formats.
Unless the SDP specifically signals otherwise, the web browser receiving a WebRTC video stream must be able to handle video at at least 20 FPS at a minimum resolution of 320 pixels wide by 240 pixels tall.
In the best scenarios ( avaible bandwidth and media devices ) VP8 had no upper mark set on resolution of vdieo stream hence the stream can even go asfar as maximum resolution of 16384×16384 pixels.
Webrtc does not specify any signalling / telecommunication protocl and it is upto the adoptor to perform ofeer/answer exchaneg in any way deemed fit for the usecase . For ex maple for a web only application on may use only plain websockets, whereas for a teelcom endpoints compatible app one should SIP as the protocol.
NAT-traversal ( ICE, STUN, and TURN)
The post describe ICE (Interactive Connectivity Establishment ) framework which is mandatory by WebRTC standards. It is find network interfaces and ports in Offer / Answer Model to exchange network based information with participating communication clients. ICE makes use of the Session Traversal Utilities for NAT (STUN) protocol and its extension, Traversal Using Relay NAT (TURN). I have written in detail about TURN based WebRTC flow diagrams in post below.
Learn about hosting / integrating different TURN servers for WebRTC in the article on “TURN server for WebRTC – RFC5766-TURN-Server , Coturn , Xirsys “.
WebRTC video quality is noticeably better than Flash.
(+) Up to 6x faster connection times
Using JavaScript WebSockets, also an HTML5 standard, improves session connection times and accelerates delivery of other OpenTok events.
(+) Reduced audio/video latency
WebRTC offers significant improvements in latency through WebRTC, enabling more natural and effortless conversations.
(+) Freedom from plugins like Flash
With WebRTC and JavaScript WebSockets, you no longer need to rely on Flash for browser-based RTC.
(+) Native HTML5 elements
Customize the look and feel and work with video like you would any other element on a web page with the new video tag in HTML5.
The major players behind the conception and advancement of WebRTC standards and libraries are IETF, W3C, Java community, GSMA. The idea is to develop a Lightweight browser-based call console, to make SIP calls from a Web page. This was successfully achieved using fundamental technologies – Javascript, html5, web-sockets and TCP /UDP, open-source sip server. It is good to note that there is no extra extension, plugin or gateway required, such as flash support. Also, it bears cross-platform support, including Mozilla, chrome so on.
Bottlnecks
Although WebRTC is a great technology and holds very good potential it is not devoid of problems
(-) Secure networks and Firewalls block RTP (-) Security in VPN and topology hiding (-) Cross-platform concerns and codecs incompatible (-) Late adopters like Microsoft and Apple
WebRTC forms a p2p communication channel between all the peers . that means as the participant count grows , it converts to a mesh networking topology with incoming and outgoing stream towards direction of each of its peers .
Two party call p2p
Peer to peer calling
p2p call
Multiparty Call and mesh network
Mesh based arrangement .
Mesh based webrtc video confeerncing
In special case of broadcasting or large number of viewers ( without outgoing media stream ) it is recommended to setup a Media Control Unit ( MCU) which will replay the incoming stream to large number of users without putting traffic load on the clients from where the stream is actually originating . Important note :
It should be noted that these diagrams do not depict the ICE and NAT traversal and have been simplified for better understanding. In real-world scenarios, almost all the time STUN and TURN servers are involved.
Also, the webrtc mandates the use of secure origin ( HTTPS ) on the webpage which invoke getusermedia to capture user media devices like audio, video and location.
navigator.mediaDevices.getUserMedia({ audio: true, video: true })
.then(function(stream) {
var video = document.querySelector('video');
// Older browsers may not have srcObject
if ("srcObject" in video) {
video.srcObject = stream;
} else {
// Avoid using this in new browsers, as it is going away.
video.src = window.URL.createObjectURL(stream);
}
video.onloadedmetadata = function(e) {
video.play();
};
})
.catch(function(err) {
console.log(err.name + ": " + err.message);
});
DOMException Error on getusermedia
Rejections of the returned promise are made by passing a DOMException error object to the promise’s failure handler. Possible errors are:
AbortError : Although the user and operating system both granted access to the hardware device, problem occurred which prevented the device from being used.
NotAllowedError : One or more of the requested source devices cannot be used at this time. This will happen if the browsing context is insecure( http instead of https) or if the user has specified that the current browsing instance /sessionis not permitted access to the device or has denied all access to user media devices globally.
NotFoundError : No media tracks of the type specified were found that satisfy the given constraints.
NotReadableError : Although the user granted permission to use the matching devices, a hardware error occurred at the operating system, browser, or Web page level which prevented access to the device.
OverconstrainedError : no candidate devices which met the criteria requested. String value is the name of a constraint which was not meet, and a message property containing a human-readable string explaining the problem. Exmaple conatraints :
SecurityError : User media support is disabled on the Document on which getUserMedia() was called.
TypeError : The list of constraints specified is empty, or has all constraints set to false.
Pan/Tilt/Zoom camera controls
RTCPeerConnection
enables audio and video communication between peers. It performs signal processing, codec handling, peer-to-peer communication, security, and bandwidth management.
stable : There is no offer/answer exchange in progress. This is also the initial state, in which case the local and remote descriptions are empty.
have-local-offer : Local description, of type “offer”, has been successfully applied.
have-remote-offer : Remote description, of type “offer”, has been successfully applied.
have-local-pranswer : Remote description of type “offer” has been successfully applied and a local description of type “pranswer” has been successfully applied.
have-remote-pranswer : Local description of type “offer” has been successfully applied and a remote description of type “pranswer” has been successfully applied. closed The RTCPeerConnection has been closed; its [[IsClosed]] slot is true.
RTCSDPType
offer : SDP offer.
pranswer : RTCSdpType of pranswer indicates that a description MUST be treated as an [SDP] answer, but not a final answer.
answer : treated as an [SDP] final answer, and the offer-answer exchange MUST be considered complete. A description used as an SDP answer may be applied as a response to an SDP offer or as an update to a previously sent SDP pranswer.
rollback : canceling the current SDP negotiation and moving the SDP [SDP] offer back to what it was in the previous stable state.
RTCPeerConfiguration
Defines a set of parameters to configure how the peer-to-peer communication established via RTCPeerConnection
iceServers of type sequence : array of objects describing servers available to be used by ICE, such as STUN and TURN servers.
iceTransportPolicy of type RTCIceTransportPolicy : bundle policy affects which media tracks are negotiated if the remote endpoint is not bundle-aware, and what ICE candidates are gathered. If the remote endpoint is bundle-aware, all media tracks and data channels are bundled onto the same transport.
relay : ICE Agent uses only media relay candidates such as candidates passing through a TURN server.
all : The ICE Agent can use any type of candidate when this value is specified.
bundlePolicy of type RTCBundlePolicy. media-bundling policy to use when gathering ICE candidates. Types :
balanced : Gather ICE candidates for each media type in use (audio, video, and data). If the remote endpoint is not bundle-aware, negotiate only one audio and video track on separate transports.
max-compat : Gather ICE candidates for each track. If the remote endpoint is not bundle-aware, negotiate all media tracks on separate transports.
max-bundle : Gather ICE candidates for only one track. If the remote endpoint is not bundle-aware, negotiate only one media track.
rtcpMuxPolicy of type RTCRtcpMuxPolicy. rtcp-mux policy to use when gathering ICE candidates.
certificates of type sequence A set of certificates that the RTCPeerConnection uses to authenticate.
iceCandidatePoolSize of type octet, defaulting to 0 Size of the prefetched ICE pool as defined in [JSEP]
RTCDataChannel
Allows bidirectional communication of arbitrary data between peers. It uses the same API as WebSockets and has very low latency.
(+) DataChannel is p2p and is also ened to end encrypted leader to higher privacy
(+) build in security due to p2p transfer
(+) high throughput than text transfer via a messaging server
(+) lower latency as p2p transfer takes shortest route
getStats
allows the web application to retrieve a set of statistics about WebRTC sessions. These statistics data are being described in a separate W3C document.
Basics for building a WebRTC based communication solution :-
Websockets for signalling / Offer Answer
TURN server like xirsys(paid), CoTURN(opensource , self hosted)
Js library for WebRTC wrappers
Https served webpage
WebRTC enabled Browser
Approaches to develop webrtc unified communication system
1. Pluggable module or npm
Source code for the WebRTC project is shipped as a pluggable library or npm module.
2. collaboration as a Service ie CaaS
Clients redirect users to our WebRTC platform for communication.
3. Communication Platform
We provider all communication and related Services as a standalone platform
Updates in W3C 13 Dec , 2019
Over the years since its adoption many of the associated tech were depricated from the Webrtc based platforms and enviornments , some of which are: OAuth as a credential method for ICE servers Negotiated RTCRtcpMuxPolicy (previously marked at risk) voiceActivityDetection RTCCertificate.getSupportedAlgorithms() RTCRtpEncodingParameters: ptime, maxFrameRate, codecPayloadType, dtx, degradationPreference RTCRtpDecodingParameters: encodings RTCDatachannel.priority
Some of the newly added features include:
restartIce() method added to RTCPeerConnection Introduced the concept of “perfect negotiation”, with an example to solve signalling races. Implicit rollback in setRemoteDescription to solve races. Implicit offer/answer creation in setLocalDescription to solve races.
References :
[1] WebRTC 1.0: Real-time Communication Between Browsers – W3C Candidate Recommendation 13 December 2019 https://www.w3.org/TR/webrtc/
WebRTC stands for Web Real-Time Communications and introduces a real-time media framework in the browser core alongside associated JavaScript APIs for controlling the media frame and HTML5 tags for displaying. If you are new to WebRTC, read “What is WebRTC?” From a technical point of view, WebRTC will hide all the complexity of real-time media behind a very simple JavaScript API.
Webrtc is a media framework which is independant of signalling protocol which means that we can plug any form of signalling to support session establishment using offer-answer handshake and SDP. Some of the popular options
Polling
XHR ( XML over HTTP Request)
Websocket ( HTTP upgraded )
SSE ( Server Sent Events )
socket.io ( use set of protocols for best compatibility and fallback)
HTTP/2
Other form of less used signalling options
FTP
HTTP
long poll
XMPP
MQTT
One may also send the SDP for local and remote over any other means of communication mechanism such as email, REST API or any custom propriatory protocol.
SSL is the secure session layer which adds encryption capability to an otherwise readable packet.
DTLS (Datagram TLS) adds Security on UDP packets which is used by Media stream and Data Channel messages.
TLS ( Tansport Layer Security) adds security to TCP messahes used in signalling such as SDP based offer answer handshake which enables setup, modification or breakdown of the session.
WebRTC offers web application developers the ability to write rich, realtime multimedia applications (think video chat) on the web, without requiring plugins, downloads or installs. It’s purpose is to help build a strong RTC platform that works across multiple web browsers, across multiple platforms.
Web API – An API to be used by third-party developers for developing web-based video chat-like applications.
WebRTC Native C++ API – An API layer that enables browser makers to easily implement the Web API proposal
Transport / Session – The session components are built by re-using components from libjingle, without using or requiring the XMPP/jingle protocol.
RTP Stack – A network stack for RTP, the Real-Time Protocol.
Session Management – An abstracted session layer, allowing for call setup and management layer. This leaves the protocol implementation decision to the application developer.
VoiceEngine is a framework for the audio media chain, from sound card to the network.
NetEQ for Voice– A dynamic jitter buffer and error concealment algorithm used for concealing the negative effects of network jitter and packet loss. Keeps latency as low as possible while maintaining the highest voice quality.
Acoustic Echo Canceler (AEC) – The Acoustic Echo Canceler is a software-based signal processing component that removes, in real-time, the acoustic echo resulting from the voice being played out coming into the active microphone.
Noise Reduction (NR) -The Noise Reduction component is a software-based signal processing component that removes certain types of background noise usually associated with VoIP. (Hiss, fan noise, etc…)
REMB (receiver-side bandwidth estimation) is more common and transport-wide-cc (sender-side bandwidth estimation) is the more modern and future looking approach
BWE (Bandwidth Estimation )
FEC (Forward Error Correction) and ULPFEC (Uneven Level Protection Forward Error Correction) RED (Redundant coding) FIR (Full Intra Request) PLI (Picture Loss Indication) for video
RTCRtpEncodingParameters dictionary describes a single configuration of a codec for an RTCRtpSender.
active : flag to set if encoding is currently actively being used. codecPayloadType : single 8-bit byte (or octet) specifying the codec to use for sending the stream.
dtx : used for audio to indicate if discontinuous transmission (a feature by which a phone is turned off or the microphone muted automatically in the absence of voice activity)
maxBitrate : (unsigned long integer) maximum number of bits per second to allow for this encoding.
maxFramerate : (double-precision floating-point) maximum number of frames per second to allow for this encoding.
ptime: (unsigned long integer) preferred duration of a media packet in milliseconds used in audio encodings.
rid : (DOMString) if set, specifies an RTP stream ID (RID) to be sent using the RID header extension.
scaleResolutionDownBy :(double-precision floating-point) specifying a factor by which to scale down the video during encoding.
default value, 1.0 if video’s size will be the same as the original.
2.0 scales the video frames down by a factor of 2 in each dimension, resulting in a video 1/4 the size of the original.
update 2020 – This article was written very early in 2013 while WebRTC was being standardised and not as widely adopted since the inception of WebRTC began in 2012.
There are many more articles written after that to explain and emphasize the detailing and application of WebRTC. List of these is below :
webrtcdevelopment Open Source WebRTC SDK and its implementation steps 
Telecom R & D 
