What is this fuss about RCS ? For the past few weeks I’ve been trying to find the answer to this one .
After much information gathering i made myself understand that :-
The Rich Communication Services programme is a global initiative to deploy inter-operator services within an industry ecosystem.
Marketed by the GSMA under the brand name joyn, RCS is an upgrade that marks the transition of messaging and voice capabilities from Circuit Switched technology to an all-IP world.
Wider and large scale IMS deployment, interoperability between different terminal vendor RCS clients and RCS service interworking between operators are the key aims of the RCS Initiative.
Whats special about RCS ?
Enhanced Phonebook: service capabilities and enhanced contacts information such as presence and service discovery.
Enhanced Messaging: enables a large variety of messaging options including chat, emoticons, location share and file sharing.
Enriched Calls: enables multimedia content sharing during a voice call, video call and video sharing (see what I see).
Five releases of the RCS specifications have been made to date. Each release expanded the scope of its predecessor.
lRelease 1 : Offered the first definitions for the enrichment of voice and chat with content sharing, driven from an RCS enhanced address book.
lRelease 2 : Added broadband access to RCS features: enhancing the messaging and enabling sharing of files.
lRelease 3: Focused on the broadband device as a primary device.
lRelease 4: Included support for LTE.
lRelease 5: The most recent release, global interoperability is a key aspect of these specifications.
As the team developed a web client for making and receiving SIP calls over websockets through a proxy SIP server , I felt its an achievement big enough. To integrate it with RCS ( Rich Communication Suite ) stack appeared as a very complicated job .
I began by adding RCS specific standards modules one by one instead of importing the whole stack / library all together .
So the modules for XCAP for buddylist , MSRP for file transfer , geolocation mapping , cloud sync of phonebook and message book have begun taking shape . In essence following features set are expected out of a RCS enabled Client ( short outline )
· OAUTH integration with operator customer portal
· RCS HTTP Auto-Configuration
· Manual IMS credentials, typically reserved for testing / troubleshooting RCS Services
· Service Discovery
o OPTIONS and Presence supported
o Address book polling
· 1-1 Chat
o Store & forward / deferred message and notification delivery
o Hotfixes compliant
· Group Chat
o Hotfixes compliant
o Geo-location Push
· File Transfer
o MSRP based
o HTTP based
o Store & forward
o Geo-location push via FT
o vCard sharing
o Thumbnail support
· Voice & video
o Best effort voice
· Network Address Book Support
o Synchronization to SyncML based network address book
o Contact import from Google, Facebook etc.
At this stage i will also put in a bit more about RCS e
What is the difference between RCS-e and RCS?
RCS-enhanced (RCS-e) is the currently available version of RCS, developed to speed time to market. It offers enhanced features such as instant messaging, live video sharing, and file transfer across any device on any network operator.
RCS e Benefits :
Focus on advanced communication
– chat , file transfer , video sharing
Easy to Use
-zero touch from end user perspective
-minimal setup for subscribers
-Interoperability across devices , infrastructure components and service providers
Low barrier to entry / simplify networks
– Capability discovery using SIP OPTIONS
– Less impact on network elements and handset battery
– Lack of presence server reduces cost and time to market
– Allows implementation in lower range devices
– One common device specification
RCS-e Customer Value Proposition
New IP Communication Services , Profile Sharing , Native Device Integration
RCS e Characteristics
Dynamic Capability Discovery
network detects when user attaches with RCs e device
detection triggers network provisioning and client configuration
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 .
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 WS 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 inbuild . Directory Service based on country / region Geolocation of approximate location detection of device logged in and visibility to others
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 Recording , playback and media file compression active frame selection DTMF Audio mixing , announcements , filters , gain control , speakers notification , Narrowband, Wideband, and Super Wideband , dynamic sample rate Video continuous presence , floor control , video lipsync , speaker tile selection VQE 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
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 .
To make a model that separates the services offered by
fixed-line (traditional telcos), mobile (traditional cellular), and converged service providers (cable companies and others who provide triple-play — voice, video, and data — services) from the access networks used to receive those services.
IMS architecture is broken into distinct layers:
Revenue streams for plain vanilla voice services are sharply falling and the need of the hour is to propose smart intuitive and creative service to kep up the Telecom market alive .
Both radio and core nework evolution all-IP packet-switched architecture standardized by 3GPP lower CAPEX ans OPEX involved Evolved from Universal Mobile Telecommunication System (UMTS), which in turn evolved from the Global Also aligned with 4G (fourth-generation mobile) LTE is backward … Continue reading →
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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 . Continue reading →
SBC ( Session Borde Controllers ) are basically gateways that provide interconnectivity between the hosted IP-PBX of the enterprise to the outside world endpoints such as telco service provider, PSTN/ TDM , SIP trunking providers or even third party OTT provider apps like skype for business etc. Continue reading →
In a VOIP system, where SIP is a signaling protocol , a SIP proxy never participates in the media flow, thus it is media agnostic. SDP packets describing a session with codecs , open ports , media formats etc are … Continue reading →
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The purpose of this article if to demo the process of using Kamailio + RTP Engine to enable SIP based WebRTC call to a traditional SIP UA like Xlite. Kamailio Will thus provide not only call routing but also NATing … Continue reading →
Dispatcher Module This modules implements a dispatcher for destination addresses. It computes hashes over parts of the request and selects an address from a destination set. The selected address is used then as outbound proxy. The module can be used … Continue reading →
Solution design and architecture, developed many custom WebRTC and SIP based solutions such as telecom applications, surveillance, IOT, Unified communication-collaboration , signalling gateways , SBC , soft turrets
Developed use cases on Machine Learning and Computer vision for VoIP and Media streaming platforms including - NLP , Image processing and Real Time Video Analytics etc
Core engineer for Plivo SIP Trunking “Zentrunk”
Core member to architecture and build IPC Unigy 360
Media broadcasting and transcoding for scalable Live Streaming Solutions
IMS integration( SIP, RTP, RTCP and related 3GPP IMS protocols )
Video Telephony on LTE/VOLTE , BLE and WiFi radio technologies
Build framework and platform for Jiyo
Invented RamuDroid , a IOT Road-Cleaning robot
Ardent contributor to Open Source Software like TFX , blockchain VoIP and among many others .
Author of the book “ WebRTC Integrator's Guide“ by Packt Publishing
Frequent techblogger and contributor to forums and communities , have many white papers and conferences presentations to my credit including an IEEE paper on Hybrid micro Grid .
Applied 2 parents, one for an elearning platform and another for image processing and mechanical design of a specific purpose robot
Speaker at many tech conferences/meetups (Jquery Conf , GraceHopper , kranky Geek , Hackaday , Barcamp , blockchain summit , plivo Tech conference etc) and won many hackathons . Jury member for smart india Hackathon .
Working on my second book titled “Innovations in Voice Over IP and its practical applications”.
“Multimedia Conferencing“ US patent - US20180284957A1