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 .

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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
• 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

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:

• https://www.quora.com/What-is-the-difference-among-CPU-GPU-APU-FPGA-DSP-and-Intel-MIC
• https://wiki.onap.org/

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/