
WebRTC Audio/Video 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. As WebRTC provides containerless bare mediastreamgtrackobjects. Codecs … Continue reading WebRTC Audio/Video Codecs
JavaScript Session Establishment Protocol (JSEP) in WebRTC handshake
This article is aimed at explaning the intricacies and detailed offer answer flow in webrtc handshake and JSEP . You can read the following artciles on WebRTC as prereq before reading through this one WebRTC API – Peerconnection , getUserMedia , Datachannel , DataStaats JSEP (JavaScript Session Establishment Protocol) JSEP (JavaScript Session Establishment Protocol) is … Continue reading JavaScript Session Establishment Protocol (JSEP) in WebRTC handshake
WebRTC CPaaS ( Communication Platform as a Service )
CPasS ( communication platform as a service ) is cloud based communication platform that provides real time communication capabilities. This should be easily integrable with any given external environment or application of the customer, without him worrying about building backend infrastructure or interfaces .

Session Border controller for WebRTC
SBC became important part of comm systems developed over SIP and MGCP. SBC offer B2BUA ( Back to Back user agent) behavior to control both signalling and media traffic.

Setting up ubuntu ec2 t2 micro for webrtc and socketio
Setting up a ec2 instance on AWS for web real time communication platform over nodejs and socket.io using WebRTC . Primarily a Web Call , Chat and conference platform uses WebRTC for the media stream and socketio for the signalling . Additionally used technologies are nosql for session information storage , REST Apis for getting sessions details to third parties.

WebRTC Security
WebRTC Security
Identity Management ,
Browser Security ,
Authentication and
Media encryption.
Browser Threat Model
Best practices for the Webrtc comm agents
ICE TURN challenges
DTLS
SRTP
continue : Streaming / broadcasting Live Video call to non webrtc supported browsers and media players
This blog is in continuation to the attempts / outcomes and problems in building a WebRTC to RTP media framework that successfully stream / broadcast WebRTC content to non webrtc supported browsers ( safari / IE ) / media players ( VLC )
Streaming / broadcasting Live Video call to non webrtc supported browsers and media players
As the title of this article suggests I am going to pen my attempts of streaming / broadcasting Live Video WebRTC call to non WebRTC supported browsers and media players such as VLC , ffplay , default video player in Linux etc . I am currently attempting to do this by making my own MP4 … Continue reading Streaming / broadcasting Live Video call to non webrtc supported browsers and media players
Steps for building and deploying WebRTC solution
Error in connectivity , errors in console , blank video are the problems that might appear . So well err things begin to get a bit complicated from here . To bypass network firewalls , corporate net policies , UDP blocks and filters we require a TURN server .
APPRTC , Talky.io , TokBox
The fundamental holes in WebRTC specification are still the same with less being done to fulfill them . Ofcourse now there are abundance of interactive WebRTC API each using a new masking method to call the same old WebRTC API function of getusermedia and peer-connection .
Call Continuity from Mobile GSM network to WebRTC
In the present age of IP telephony when telecom convergence is the big thing all around the world , need of the hours is to enable fixed and mobile Service Providers ( SP ) to monetize the subscriber’s phone number by extending it to new web based services.SPs can offer a WebRTC Communicator endpoint that … Continue reading Call Continuity from Mobile GSM network to WebRTC
WebRTC communication over Web Services
This post is about communication from any application to WebRTC using Web Services. HTTP and XML is the basis for Web services WSDL WSDL stands for Web Services Description Language It specifies the location of the service and the operations (or methods) the service exposes. XML-based language for describing Web services. SOAP SOAP stands for … Continue reading WebRTC communication over Web Services
WebRTC Media Streams
SDP signaling and negotiation for media plane Read more on SDP and its attributes : https://telecom.altanai.com/2014/01/03/sip-in-depth/(opens in a new tab) Media plane adaptation is done at the SBC for network carried media, it should be done for all network hosted media services which face peer-to-peer media. The high-level architecture elements of WebRTC media streams can be divided … Continue reading WebRTC Media Streams

Performance of WebRTC sites and apps
As security is a broad topic touching on many sections of WebRTC this section is not meant to address all topics but instead to focus on specific “hot spots”, areas that require special attention due to the unique properties of the WebRTC service. There are several security related topics that are of particular interest with respect to WebRTC. They can be grouped into the following areas: Identity Management
Browser Security
Authentication
Media encryption
Syntax checks using regex
Regulatory/Legal Considerations and CALEA with WebRTC development
This post is deals with some less known real world implication of developing and integrating WebRTC with telecom service providers network and bring the solution in action . The regulatory and legal constrains are bought to light after the product is in action and are mostly result of short sightedness . The following is a … Continue reading Regulatory/Legal Considerations and CALEA with WebRTC development
WebRTC compatible android client
This post describes the requirement of creating a SIP phone application on android over the same codecs as WebRTC ( PCMA , PCMU , VP8) . In my project concerning the demonstration of WebRTC inter operability ( presence , audio / video call , message ) with a native android client , I had to … Continue reading WebRTC compatible android client
Difference between WebRTC and plugin based communication
A lot of service providers ie telecom operators had deduced their own ways to provide Web based communication even before WebRTC was born . With time , as WebRTC has become stronger , more secure , resilient to failure they have come around to migrate their existing system from previous closed box native APIs to … Continue reading Difference between WebRTC and plugin based communication
E-Learning
e-learning platform which harness the power of Internet for the purpose of distance education and where students around the world volunteer to teach each other any subject they wish to. This will be made possible through a combination of real time communication technologies like WebRTC and plethora of knowledge repositories.

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 … Continue reading WebRTC SIP / IMS solution

WebRTC business benefits to OTT and telecom carriers
Historically, RTC has been corporate and complex, requiring expensive audio and video technologies to be licensed or developed in house. Integrating RTC technology with existing content, data and services has been difficult and time consuming, particularly on the web.Now with WebRTC the operator finally gets a chance to take the shift the focus from OTT … Continue reading WebRTC business benefits to OTT and telecom carriers

What is WebRTC?
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 webrtcdevelopment 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 … Continue reading What is 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 … Continue reading WebRTC Stack Architecture and layers
WEBRTC CALL BETWEEN BROWSER AND SIP PHONE
HTML5 and WebRTC enabled Web Client : We are using open source HTML5 SIP client entirely written in javascript to make it light and to have easy integration with the SIP server. No extension, plugin or gateway is needed to initiate the call from the web Client. The media stack rely on WebRTC. The client … Continue reading WEBRTC CALL BETWEEN BROWSER AND SIP PHONE