There is a strong evolution being seen in CSP space. Now operators are looking forward to implement the open standard for intelligent networks. It reduces their dependency on proprietary platforms and on vendor’s road maps. Open –source platform gives operator flexibility to develop their own applications without being dependent on vendor. An open, standards based, service logic execution environment (SLEE) that integrates with current and future networks is the key to providing innovative and revenue generating services. Providing one (standards based) carrier grade execution environment that integrates SS7, SIP, OSA/Parlay, OSS/BSS and J2EE environments offers significant benefits to operator.
Business benefits of SIP JAINSLEE based platform
Network Independence: The JAIN SLEE framework is independent of any particular network protocol, API or network topology. This is supported through the resource adaptor architecture
Portable Services: Application components can be developed and then deployed on JAIN SLEE compliant platforms from different vendors without recompilation or source code modification.
Supports Complex Applications: JAIN SLEE application components can have state, can be composed from other components, can create and destroy other application components, can invoke other application components both synchronously and asynchronously, and can invoke resource adaptors.
Industry Standard: JAIN SLEE is specified via the Java Community Process which allows multiple companies and individuals to collaborate in developing Java technology specifications.
In order to reduce the operating cost of legacy infrastructure more and more operators are investing and implementing open source platform. These new platforms bring agility and new service delivery capability to CSP.
The JAINSLEE based platform can be used to develop and deploy carrier-grade applications that use SS7-based protocols such as INAP and CAP, IP protocols such as SIP and Diameter, and IT / Web protocols, such as HTTP Servlet, XML and Service Orientated Architectures (SOA).
Fundamental Concepts :
Application can be written once and run on many different implementations of JAIN SLEE.
Applications can access resources and protocols across multiple networks from within the JAIN SLEE environment.
Follows the ACID transaction .
component model for structuring the application logic of communications applications as a collection of reusable
object-orientated components, and for composing these components into higher level and more sophisticated services.
SLEE specification also defines the management interfaces used to administer the application environment and also
defines set of standard Facilities (such as the Timer Facility, Trace Facility, and Alarm Facility so on )
Extension framework to allow new external protocols and systems (such as MSCs, MMSCs, SMSCs, Softswitchs, CSCFs, HLRs) to be integrated.
Characteristics of SLEE specification
• Event based model, asynchronous, support for composition
• Container manages component state
• Container manages garbage collection of components
• Transaction boundaries for demarcation and semantics of state replication
• Management of lifecycle of Server, Services, Provisioned state
• Versioned services, upgrade of services, existing activities stay on existing service instances, new activities are directed to instances of upgraded services
• Independent of network technology/ protocols/elements through resource adaptor architecture
Entities :
Service
A service in JAIN SLEE terminology is a managed field replaceable unit.
The system administrator of a JAIN SLEE controls the life cycle (including deployment, undeployment and on-line upgrade) of a service. The program code can include Java classes Profiles, and Service Building Blocks.
Profile
A JAIN SLEE Profi le contains provisioned service or subscriber data.
Service Building Blocks running inside the JAINSLEE may access profiles as part of their application logic.
Service Building Block
The element of re-use defined by JAINSLEE is the Service Building Block (SBB).
An SBB is a software component that sends and receives events and performs computational logic based on the receipt of events and its current state. SBBs are stateful.
The program code for an SBB is comprised of Java classes.
Event
An event represents an occurrence that may require application processing.
An event may originate from a number of different sources, for example, an external resource such as a communications protocol stack, from the SLEE itself, or from application components within the SLEE.
Resources and Resource ADAPTERS
Resources are external entities that interact with other systems outside of the SLEE, such as network elements (HLR, MSC, etc), protocol stacks, directories and databases.
A Resource Adaptor implements the interfacing of a Resource into the JAINSLEE environment.
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 ( Over The Top service providers like SKype , Google chat WebEx etc that were otherwise eating away the Operators revenue ) to its very own WebRTC client Server solution , hence making the VOIP calls chargeable , while at the same time being available from any client ( web or softphone based )
WebRTC has now implemented open standards for real-time, plugin-free video, audio and data communication.
Many web services already use RTC, but need downloads, native apps or plugins. These includes Skype, Facebook (which uses Skype Flash ) and Google Hangouts (which use the Google Talk plugin). Downloading, installing and updating plugins can be complex, error prone and annoying , such as Flash , Java .,etc
Plugins can be difficult to deploy, debug, troubleshoot, test and maintain—and may require licensing and integration with complex, expensive technology. It’s often difficult to persuade people to install plugins in the first place/ bookmark it or keep it activated at all times.
WebRTC support across various browsers , pic source : caniuse.com
API support from browser
PeerConnection API
getUserMedia
WebAudio Integration
dataChannels
TURN support
Echo cancellation
MediaStream API
Multiple Streams
Simulcast
Screen Sharing
mediaConstraints
Stream re-broadcasting
getStats API
ORTC API
H.264 video
VP8 video
Solid interoperability
srcObject in media element
Promise based getUserMedia
Promise based PeerConnection API
WebRTC trends
ic source : Disruptiveanalysis
The APIs and standards of WebRTC can democratize and decentralize tools for content creation and communication—for telephony, gaming, video production, music making, news gathering and many other applications.
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
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
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 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 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
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 : 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 .
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+
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.
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.
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 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.
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
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
Communication model
Security model
Firewall and NAT traversal
Media functions
Functionality such as media codecs, security algorithms, etc.,
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.
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 :
– signalling at the protocol level (such as SIP, MGCP and SS7)
•For telephony, data and wireless communications networks, the Java APIs defined through.
– service portability
– network independence
– open development
•A Service Logic Execution Environment (SLEE) is high-throughput, low-latency, event-processing application environment.
•JAIN SLEE is designed specifically to allow implementations of a standard to meet the stringent requirements of communications applications (such as network-signaling applications).
Goals of JAIN SLEE are:
– Portable services and network independence.
– Hosting on an extensible platform.
– services and SLEE platform available from many vendors.
Key Features are :
•Industry standard :- JSLEE is the industry-agreed standard for an application server that meets the specific needs of telecommunications networks.
•Network independence:-The JSLEE programming model enables network independence for the application developer. The model is independent of any particular network protocol, API or network topology.
•Converged services:- JSLEE provides the means to create genuinely converged services, which can run across multiple network technologies.
•Network migrations :-As JSLEE provides a generic, horizontal platform across many protocols, independent of the network technology, it provides the ideal enabler technology for smooth transition between networks.
•Global market—global services:-JSLEE-compliant applications, hosted on a JSLEE application server are network agnostic. A single platform can be used across disparate networks
•Robust and reliable:- As with the enterprise application server space, deploying applications on a standard application server that has been tested and deployed in many other networks reduces logic errors, and produces more reliable applications
webrtcdevelopment
