RCS ( Rich Communication Suite )

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 understood that majority of communicatio platform provider’s mostly OTT such as imessage from apple , RBM already support this features. And it is partially a term coined by Google to bring smart communication festures in Android and other smart phones. In essence RCSe is a part of broader IMS archietcture pulished by GSMA

Update : after 2019 plenty of carriers also already provide RCSe as a replacement of SMS . OEM providers like Samsung also have RCS feature inbuild in phones.

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).

RCS releases

Five releases of the RCS specifications have been made to date. Each release expanded the scope of its predecessor.

Release 1 : Offered the first definitions for the enrichment of voice and chat with content sharing, driven from an RCS enhanced address book.

Release 2 : Added broadband access to RCS features: enhancing the messaging and enabling sharing of files.

Release 3: Focused on the broadband device as a primary device.

Release 4: Included support for LTE.

Release 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 ) Provisioning

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
  • Delivery & display notifications o Message composing indication
  • 1-1 Chat
    o IMDN
    o Is-Composing
    o Store & forward / deferred message and notification delivery
    o Hotfixes compliant
  • Group Chat
    • Hotfixes compliant
    • Is-Composing
    • IMDN
    • Geo-location Push
    • Start and end a group chat session
    • Add or remove participants
    • Broadcast / Send messages to all participants
  • File Transfer
    • Send file via URL or as MIME attachment o Accept or Reject file transfer
    • MSRP based
    • HTTP based
    • Store & forward
    • Geo-location push via FT
    • vCard sharing
    • Thumbnail support
  • Voice & video
    o Best effort voice
    o Transcode
  • Network Address Book Support
    o Synchronization to SyncML based network address book
    o Contact import from Google, Facebook etc.

RCS APIs

( published under Joyn in GSMS RDC Devloepr documents in http://rcs.oneapi-gw.gsma.com)

There are many individual RCS APIs available. Brief details are as follows.

  1. Log user in and out of the RCS server Allows address book contacts to receive periodic updates of user availability for chat, file transfer and other services. This is essential for use of the RCS services.

2. Subscribe application to server-originated notifications This is one of the most important aspects of RCS as events are asynchronous; the notifications channel is used by your application to know how an API call is proceeding and to receive updates from other users such as messages, files and status updates. You can subscribe to certain types of notifications if your application is focused on particular RCS services.

3. Network address book access and management RCS provides each registered user with a server managed address book. Though the other APIs are not restricted to contacts in the network address book this makes it much more convenient to connect with other users. The network address book can have contacts added, removed, attributes changed, and is available to any application the user signs in to.

4. Individual chat services Unlike SMS, which is broadly a ‘fire and forget’ service, the RCS chat services allow much more sophisticated messaging services to be built. Depending on user configuration you can check out whether the user has received the message, and displayed it. RCS chat also allows feedback to other parties when a user is composing a message. There are also group chat services allowing multiple parties to collaboratively message. RCS provides API features for setting up a group chat session, adding or removing chat participants.

5. File Transfer These APIs allow applications to send content between users – whether this is a picture or a document, and whether it is stored originally on the user’s device or elsewhere located on the Internet. They allow the recipient to receive information about the file separately from the file itself so the recipient can choose whether or not to accept the file.

6. User capabilities API Enables the application to retrieve and amend system managed common capabilities of the user.

At this stage I will also put in a bit more about RCS e

RCS e ( enhanced )

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
  • Universal
    Allows implementation in lower range devices
    One common device specification

RCS-e Customer Value Proposition

New IP Communication Services , Profile Sharing , Native Device Integration

rcs5
rcs6

RCS e Characteristics

Dynamic Capability Discovery
User Perspective
network detects when user attaches with RCs e device
detection triggers network provisioning and client configuration
authentication by network
SSO / GIBA in 3G coverage
SIP Digest in Wifi
Encryption for Wifi access
TLS for SIP and TCP media or IPsec
SRTP for UDP or IPsec
NAT traversal and Keep-alives

Realted Terms

MSRP

File Transfer using   MSRP (or Message Session Relay Protocol) is an instant messaging or chat protocol, defined by the IETF in RFC4975. MSRP is a text-based, connection-oriented protocol for exchanging  arbitrary (binary) MIME content, especially instant messages.

References

Wikipedia – https://en.wikipedia.org/wiki/Rich_Communication_Services

GSMA Rich Communication Services APIs Developer Guide vs1.2

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/