WebRTC SIP / IMS solution

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 ,

webrtccallflow

A very high level architecture of solution deployment in IMS world could be

solution arch2

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 solution
WebRTC client deployment view , accessible devices , network elements
WebRTC deploymenet overview and inetraction with other network elemets such as gateway , cloud storage ,  sipserver , IMS
WebRTC 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
  • Multi vendor support
  • Interactive webrtc services
  • Media Services
    • Automated Natural language Speech recognition
    • Semantic processing via ML
    • Enhanced incall services replacing IVR ( touch -tone)
    • VQE (voice Quality Enhancements)
    • Encoding and Decoding – Multiple Codec Support
    • Transcoding
    • Silence Suppression
  • Security via TLS, encryption and AAA
  • Http, NFS caching
  • NAT using Xirsys TURN
  • Recording, playback and media file compression
  • active frame selection
  • DTMF (Dual Tone Multi Frequency)
    • SIP info messages (out-of-band)
    • SIP notify messages (out-of-band)
    • Inband DTMF not supported yet
  • Audio
    • mixing
    • announcements ( VXML, MSML )
    • filters
    • gain control ( AGC using webrtc stack)
    • noise suppresesion ( webrtc stack)
    • speakers notification
    • Narrowband, Wideband, and Super Wideband
    • dynamic sample rate
  • Video
    • continuous presence ( Face detetion )
    • floor control
    • video lipsync (sync)
    • speaker tile selection
  • VQE (Voice Quality Enhancement )
    • Acoustic Echo Cancelation
    • noise reduction
    • noise line detection
    • noise gating
    • Packet Loss concealment
  • Call analyics
    • progress analysis
    • 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

  • HD (High Definition ) calls
  • UCC ( conf , draw-board, speech recognition , realtime streaming)
  • immersive experiences ( Augmented reality , virtual reality , face recognition , tracking )
  • contextual communication ( transcription etc)
  • video content delivery with deep media analytics

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 .

Ref:

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.

Github Repo – https://github.com/altanai/unifiedCommunicator

Unified Communications and Collaborations ( UC&C ) – https://telecom.altanai.com/2013/07/12/unified-communication/

What is WebRTC?

webrtc draft
 

WebRTC 1.0: Real-time Communication Between Browsers – W3C Candidate Recommendation 13 December 2019 https://www.w3.org/TR/webrtc/

Read more in the layers of webrtc  and their functionalities here :  WebRTC layers

webrtc_development_logowebrtcdevelopment
Open Source WebRTC SDK and its implementation steps https://github.com/altanai/webrtc

What is WebRTC ?

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 of either internal or external plugins.

  • Enables browser to browser media streaming over secure RTP profile
  • Standardization , on a 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
  • BSDD style license
  • free, open project avaiable in all major borwsers 

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.

 The following is the browser side stack for webrtc media .  

WebRTC media stack Solution Architecture
WebRTC Media Stack

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.

Video Resolution handling

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.

Independant of Signalling 

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 . 

Read more about WebRTC handshakes :

NAT-traversal technologies such as ICE, STUN, and TURN

Have written in detail about TURN based WebRTC flow diagrams .

https://telecom.altanai.com/2015/03/11/nat-traversal-using-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) 

NAT and TURN Relay

Learn about hosting / integrating different TURN servers for WebRTC

TURN server for WebRTC – RFC5766-TURN-Server , Coturn , Xirsys – https://telecom.altanai.com/2015/03/28/turn-server-for-webrtc-rfc5766-turn-server-coturn-xirsys/

Why is WebRTC importatnt ?

Significantly better video qualityWebRTC video quality is noticeably better than Flash.
Up to 6x faster connection timesUsing JavaScript WebSockets, also an HTML5 standard, improves session connection times and accelerates delivery of other OpenTok events.
Reduced audio/video latencyWebRTC offers significant improvements in latency through WebRTC, enabling more natural and effortless conversations.
Freedom from FlashWith WebRTC and JavaScript WebSockets, you no longer need to rely on Flash for browser-based RTC.
Native HTML5 elementsCustomize 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 conception and advancement of WebRTC standards and libraries are  :

IETF , W3C , Java community , GSMA .   The idea is to develop a Light -weight browser based call console , to make SIP calls from Web page .This was successfully achieved using fundamental technologies as Javascript , html5 , web-sockts  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 .

 Peer to peer Communication

 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

two party call
p2p call

Multiparty Call and mesh network

Mesh based arrangement .

Multiparty party call
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 :     1.It should be notes that these diagrams do not depict the ICE and NAT traversal and have been simplifies for better understanding. In real world scenarios there is almost all the time a STUN and TURN server involved .  

More on TURN Servers is given here : NAT traversal using STUN and TURN

2.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 .

Browser Adoption

As of March 2020 , webrtc is supported on following client’s browsers

  • Desktop PC
    Microsoft Edge 12+[25]
    Google Chrome 28+
    Mozilla Firefox 22+[26]
    Safari 11+[27]
    Opera 18+[28]
    Vivaldi 1.9+
  • Android
    Google Chrome 28+ (enabled by default since 29)
    Mozilla Firefox 24+[29]
    Opera Mobile 12+
  • Chrome OS
  • Firefox OS
  • BlackBerry 10
  • iOS
    MobileSafari/WebKit (iOS 11+)
  • Tizen 3.0

Furthermore , read about the Steps for building and deploying WebRTC solution – https://telecom.altanai.com/2014/12/04/steps-for-building-and-deploying-webrtc-solution/

TURN based media Relay

WebRTC APIs

Javascript functions  to access and process the browser media stack

getUserMedia

acquires the audio and video media (e.g., by accessing a device’s camera and microphone)

Properties

ondevicechange

Methods

enumerateDevices()
getDisplayMedia()
getSupportedConstraints()
getUserMedia()

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 :

var constraints = { video: { facingMode: (front? "user" : "environment") } };

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.

RTCPeerConnection

enables audio and video communication between peers. It performs signal processing, codec handling, peer-to-peer communication, security, and bandwidth management.

Properties

canTrickleIceCandidates
connectionState
currentLocalDescription
currentRemoteDescription
getDefaultIceServers()
iceConnectionState
iceGatheringState
localDescription
onaddstream
onconnectionstatechange
ondatachannel
onicecandidate
oniceconnectionstatechange
onicegatheringstatechange
onidentityresult
onnegotiationneeded
onremovestream
onsignalingstatechange
ontrack
peerIdentity
pendingLocalDescription
pendingRemoteDescription
remoteDescription
sctp
signalingState

Methods

addIceCandidate()
addStream()
addTrack()
close()
createAnswer()
createDataChannel()
createOffer()
generateCertificate()
getConfiguration()
getIdentityAssertion()
getReceivers()
getSenders()
getStats()
getStreamById()
getTransceivers()
removeStream()
removeTrack()
restartIce()
setConfiguration()
setIdentityProvider()
setLocalDescription()
setRemoteDescription()

 signalling state transitions diagram , source W3C

RTC Signalling states

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
An 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.

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

Peer to Peer DTMF

-tbd

Call Setup betweeb WebRTC Endpoints

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 featufres 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 :

WebRTC 1.0: Real-time Communication Between Browsers – W3C Candidate Recommendation 13 December 2019https://www.w3.org/TR/webrtc/

WebRTC Stack Architecture and layers

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. 

Codec Confusion :

Video Codecs

Currently VP8 is the codec of choice since it is royalty-free. In mobility today, the codec of choice is h264. H264 is not royalty-free. But it is native in most mobile handsets due to its high performance.

Audio Codecs

Opus is a lossy audio compression format developed by the Internet Engineering Task Force (IETF) targeting a broad range of interactive real-time applications over the Internet, from speech to music. As an open format standardized through RFC 6716, a reference implementation is provided under the 3-clause BSD license. All known software patents Which cover Opus are licensed under royalty-free terms.

G.711 is an ITU (International Telecommunications Union) standard for  audio compression. It is primarily used in telephony. The standard was released in 1972. It is the required standard in many voice-based systems  and technologies, for example in H.320 and H.323 specifications.
Speex is a patent-free audio compression format designed for speech and also  a free software speech codec that is used in VoIP applications and podcasts. Some consider Speex obsolete, with Opus as its official successor, but since
significant content is out there using Speex, it will not disappear anytime soon.

G.722 is an ITU standard 7 kHz Wideband audio codec operating at 48, 56 and 64 kbit/s. It was approved by ITU-T in 1988. G722 provides improved speech quality due to a wider speech bandwidth of up to 50-7000 Hz compared to G.711 of 300–3400 Hz.

AMR-WB Adaptive Multi-rate Wideband is a patented wideband speech coding standard that provides improved speech quality due to a wider speech bandwidth of 50–7000 Hz. Its data rate is between 6-12 kbit/s, and the codec is generally available on mobile phones.

Architecture :

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.

WebRTCpublicdiagramforwebsite

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.

STUN/ICE – A component allowing calls to use the STUN and ICE mechanisms to establish connections across various types of networks.

Session Management – An abstracted session layer, allowing for call setup and management layer. This leaves the protocol implementation decision to the application developer.

VoiceEngine – VoiceEngine is a framework for the audio media chain, from sound card to the network.

iSAC / iLBC / Opus

iSAC: A wideband and super wideband audio codec for VoIP and streaming audio. iSAC uses 16 kHz or 32 kHz sampling frequency with an adaptive and variable bit rate of 12 to 52 kbps.

iLBC: A narrowband speech codec for VoIP and streaming audio. Uses 8 kHz sampling frequency with a bitrate of 15.2 kbps for 20ms frames and 13.33 kbps for 30ms frames. Defined by IETF RFCs 3951 and 3952.

Opus: Supports constant and variable bitrate encoding from 6 kbit/s to 510 kbit/s, frame sizes from 2.5 ms to 60 ms, and various sampling rates from 8 kHz (with 4 kHz bandwidth) to 48 kHz (with 20 kHz bandwidth, where the entire hearing range of the human auditory system can be reproduced). Defined by IETF RFC 6176.

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…)

Video Engine – VideoEngine is a framework video media chain for video, from the camera to the network, and from network to the screen.

VP8  – Video codec from the WebM Project. Well suited for RTC as it is designed for low latency.

Video Jitter Buffer – Dynamic Jitter Buffer for video. Helps conceal the effects of jitter and packet loss on overall video quality.
Image enhancements -For example, removes video noise from the image capture by the webcam.

W3C contribution


w3c

  • 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

ietf

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

WG RFC   Date

  • draft-ietf-rtcweb-audio-02      2013-08-02
  • draft-ietf-rtcweb-data-channel-05      2013-07-15
  • draft-ietf-rtcweb-data-protocol-00      2013-07-15
  • draft-ietf-rtcweb-jsep-03      2013-02-27
  • draft-ietf-rtcweb-overview-07      2013-08-14
  • draft-ietf-rtcweb-rtp-usage-07     2013-07-15
  • draft-ietf-rtcweb-security-05      2013-07-15
  • draft-ietf-rtcweb-security-arch-07      2013-07-15
  • draft-ietf-rtcweb-transports-00      2013-08-19
  • draft-ietf-rtcweb-use-cases-and-reqs-11      2013-06-27
  • Plus over 20 discussion RFC drafts

What will be the outcome of WebRTC Adoption?

In simple words, it’s a phenomenal change in decentralizing communication platforms from proprietary vendors who heavily depended on patented and royalty bound technologies and protocols.  It will revolutionize internet telephony.  Also it will emerge to be platform-independent ( ie any browser, any desktop operating system any mobile Operating system ).

WebRTC allows anybody to introduce real-time communication to their web page as simple as introducing a table.

Read More about webRTC business benefits


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 :

For SIP IMS and WebRTC

Read about STUN and TURN which form a crtical part of any webrtc based communication system

Security of WebRTC based CaaS and CPaaS

WebRTC APIs