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 .
People working at different locations need a fast, secure and reliable way to share information across computer networks . This is were a way to connect private networks over and top of public network becomes necessary and Virtual Private Network comes into picture .
SIP ( Session Initiation Protocol ) for VPN
VOIP across an SSL-based VPN is achieved in good quality by encapsulating the UDP VOIP packets ( SIP and RTP ) in TCP/IP .
Data used for defining a VPN like its Groups, its Members and the associated profiles is organized hierarchically.It includes information like who is the operator, subscriber of VPN, group ID and member ID.
Groups created to implement policies and restrictions common to a set of users.These include:
Apply permissions to call between the Groups and to the outside world
Apply pricing between distinct types of of PNP (Mobile, Fixed, Privileged list)
Some numbers assigned a preferential tariff plan. These numbers are not part of the VPN ( Virtual On-Net) .
privileged list within a VPN across multiple groups
VPN has no negative influence on latency, jitter and packet loss
With enabling authentication, encryption, HMAC, anti-replay attack, and initialization vector, and use small RTP size for Codec, the vpn overhead is high
For developing a VPN application counters are employed , some of which could be as follows
* Number of calls On-Net and Off-Net
* Numbers of Calls VPN
* Number of calls with Forced On-Net
Calls between endpoints like
* MS to MS Normal (mobile)
* MS to MS Privilege
* MS toward PABX
Success Fail rate
Number of calls successful without rerouting
Number of calls with successful rerouting
Number of calls with Failure (Failed = No answer, Busy, Not reachable, Congestion)
Number of calls on non-response (No Answer)
Number of calls on Not Reachable
Number of calls Route Select Failure
Number of calls on busy
Number of calls barred by VPN service.
Total number of queries
Number of States created/modified
Number of change in the rights of calls
Number of issuance of observation Reports
Lets see how would a SIP based VPN services over telecom application server with Service Broker works .
Leveraging the Service Broker to offer voice VPN service to existing Subscribers is an arduous task. The Subscriber shall benefit from reduced charging rates for VPN calls (ON-Net), improved employee connectivity (within the VPN scope) and a consistent user experience across fixed and mobile phones.
VPN services shall be integrated with the R-IM-SSF component of the service broker. R-IM-SSF shall provide mediation as well as session and state management capabilities that shall make VPN service available over multiple networks including SS7 and IMS networks.
note : R-IM-SSF = reverse IMS gateway to IN
The subscriber base can be interfaces via a SMP that might also be used to add groups and assign right and privilege to member
note : SMP is the Provisioning interface for VPN service subscriber
Features of VPN application
1.Private numbering plan for both mobile and fixed subscribers (Short number dialing).
2.Distribution of subscriber under a hierarchical Data Model :
Subscriber VPN( Enterprise Level)
Group of Users ( Group level. Can be either of type Mobile or PABX )
State (End user of service)
3.Grouping of a short number on the basis of following types:
Member of mobile VPN
4. Forced On-Net call handling, which shall allow user to dial the public number of another On-Net user with On-Net call Features.
5.Virtual On-Net Call Handling which allocates On-Net extension to non VPN users( Privileged list)
6.Off-Net call Handling via exhaust code which shall allow vpn users to access non-vpn public numbers
7. Prohibit the call based on a set of rules like ( all off-net calls barred).
8.Allow calls based on destination numbers. For example allow off-net calls for numbers provisioned in the white list(allowed list)
9.Outgoing call screening on the basis of time( Time based barring)
WebRTC video quality is noticeably better than Flash.
Up to 6x faster connection times
Reduced audio/video latency
WebRTC offers significant improvements in latency through WebRTC, enabling more natural and effortless conversations.
Freedom from Flash
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 .
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
Multiparty Call and mesh network
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 .
IMS is a an architectural framework for IP based multimedia rich communications. It was standardized by a group called 3GPP formed in 1999.
It started as an enabler for 3rd generation mobile networks in European market and later spread to wirelne networks too . IMS became the key to Fixed Mobile Convergence (FMC).
Based on IETF Protocols (such as SIP, RTP, RTSP, COPS, DIAMETER, etc.) , IMS is now crucial for controlling conmmunication in a IP based Next Genration Network ( NGN ).
Communication service providers and telecom operators are migrating from circuit-switched networks to IMS technology with the increasing bandwidth (5G) and user expectations.
Why IMS ?
Early days TDM networks were not robust enough to support emerging technologies and data networking. There was a need to migrate from voic eonly network to Triple play network ( voice , video and data ).
Other factors included :
rapid service development
service availiability in both home and roaming network
wireline and wireless convergence
Due to these above mentioned reasons TDM was outdated and IMS gained support .
What benefits does IMS bring ?
It offers counteless applications around rich multimedia services on wireless , packet swtched and even tradional circuit switched networks.
•Easier to Create and Deploy New Applications and Services
Enhanced applications are easier to develop due to open APIs and common network services.
Third-party developers can offer their own applications and use common network services, sharing profits with minimal risk
New services involving concurrent sessions of multimedia (voice, video, and data) during the same call are now possible
Reduced time-to-market for new services is possible because service providers are not tied to the timescales and functions of their primary NEPs
•Capture New Subscribers, Retain Current Subscribers
Better voice quality for business applications, such as conferencing, is possible
Wireless applications (like SMS, and so on) can be offered to wire line or broadband subscribers.
Service providers can more easily offer bundled services.
•Lower Operating and Capital Costs
Cost-effective implementation of services is possible across multiple transports, such as Push-To-Talk (PTT), presence and Location-Based Services (LBS), Fixed-Mobile Convergence (FMC), mobile video services, and so on.
Common provisioning, management, and billing systems are supported for all networks.
Significantly lower transport costs result when moving from time-switched to packet-switched channels.
Service providers can take advantage of competitive offerings from multiple NEPs for most network elements.
IMS results in reduced expenses for delivering licensed content to subscribers of different types of devices, encodings, or networks.
The reason for widespread adoption of IMS is also that it follows standards and open interfaces from 3GPP and ETSI, also is flexible for policy control , OSS/BSS , Value Added Services etc .
1. Abstraction from Underlying Network :
IMS is essentially leading towards an open and standardized network and interface , irrespective of underlay network.
2. Fixed /Mobile Convergence
Inter operability with Circuit Switched (CS) Mobile application Part (MAP)
Location awareness between home and visiting network.
4. Application layer Call Control
IMS application layer has the provision for defining proxy or B2BUA based call flow completion . This leads to operator being able to introduce business logic into call sessions.
IMS is supplemented by SIP (IETF ) , Diameter ( IETF) and H248(ITU-T). The release cycle of IMS is as follows :
2002-03-14 Rel-5 : IMS was introduced with SIP. Qos voice over MGW.
2004-12-16 Rel-6 : Services like emergency , voice call continuity , IPCAN ( IP connectivity Access Network )
2005-09-28 Rel-7 : Single Radio Voice Call Continuity , multimedia telephony,eCall ,ICS
2008-12-11 Rel-8 : IMS centralized services , supplementary services and internetworking between IMS and Circuit Switched Networks,charging , QoS
2009-12-10 Rel-9 : IMS emergency numbers on GPRS , EPS(Enhanced packet system) , Custom alert tone , MM broadcast/Multicast
2011-3-23 Rel-10 : home NodeB, M2M, Roaming and Inter UE transfer
2012-09-12 Rel-11 :-tbd
2014-09-17 Rel-12 :- tbd
2015-12-11 Rel-13 :- tbd
Majorly IMS is divided into 3 horizontal layers given below :
•Transport / MediaEndpoint Layer
Unifies transports and media from analog, digital, or broadband formats to Real-time Transport Protocol (RTP) and SIP protocols. This is accomplished by media gateways and signaling gateways.
It also includes media servers with media processing elements to allow for announcements, in-band signaling, and conferencing. These media servers are shared across all applications (voicemail, interactive response systems, push-to-talk, and so on), maximizing statistical use of the equipment and creating a common base of media services without “hard-coding” these services into the applications.
•Session and Control Layer
This layer arranges logical connections between various other network elements. It provides registration of end-points, routing of SIP messages, and overall coordination of media and signaling resources.
IMS core which is part of this layer primarily contains 2 important elements Call Session Control Function (CSCF) and Home Subscriber Server (HSS) database. These are explained below
HSS Home Subscriber Server
It is a database of user profiles and location information . It is responsible for name/address resolution and also authorization/authentication .
CSCF Call Session Control Function
Handles most routing, session and security related operation for SIP messages . It is further divided into 3 parts :
Proxy CSCF: P_CSCF is the first point of contact from any SIP UA. It proxies UE requests to subsystem.
Serving CSCF: S-CSCF is a powerful part of IMS Core as it decides how UE request will be forwarded to the application servers.
Interrogating CSCF: I-CSCF initiates the assignment of a user to an S-CSCF (by querying the HSS) during registration.
•Application Services Layer
The Application Services Layer contains multiple Application Servers (AS), such as:
Telephony Application Server (TAS) – for defining custom call flow logic
IP Multimedia Services Switching Function (IM-SSF)
Open Service Access Gateway (OSA-GW), and so on.
Additional Links :
To Read about IMS service Switching function and reverse service Switching function click here .
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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