A 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 .
Traditionally , with IP protected protocols , licensed codecs maintaining a signalling protocol stack , network interfaces building communication platform was a costly affair. Cisco , Facetime , Skype were the only OTT ( over the top) players taking away from telco’s call revenue .
However with the advent of standardize , open source protocol and codecs plenty of CPaaS providers have crowded the market making more supply than there is demand. A customer wanting to quickly integrate real time communications on his platform has many options to choose from. This article provides an insight to how CPaaS solution are architectured and programmed
Call server + Media Server that can be interacted with via UA
Comm clients like sipphones , webrtc client , SDK ( software development kits ) or libraries for desktop , embedded and/or mobile platforms .
APIs that can trigger automated calls and perform preprogrammed routing.
Rich documentation and samples to build various apps such as call centre solutions , interactive auto-attendant using IVR , DTMF , conference solutions etc .
Some CPaaS providers also add features like transcribing ,transcoding , recording , playback etc to provide edge over other CPaaS providers
Advantages of using a CPaaS vs building your own RTC platform
Tech insights and experiences
companies who have been catering to telco and communication domain make robust solutions based on industry best practices which beats novice solution build in a fortnight anyday
keeping up with emerging trends
Market trends like new codecs , rich communication services , multi tenancy, contextual communication , NLP , other ML based enhancements are provided by CPaaS company
Auto Scaling , High Availability
A firm specializing in CPaaS solution has already thought of clustering and autoscaling to meet peak traffic requirements and backup/replication on standby servers to activate incase of failure
CAPEX and OPEX
using a Cpaas saves on human resources, infrastructure, and time to market. It saves tremendously on underlying IT infrastructure and many a times provides flexible pricing models
In a nutshell I have come across so many small size startups trying to build CPaaS solution from scratch but only realising it after weeks of trying to build a MVP that they are stuck with firewall, NAT, media quality or interoperability issues . Since there are so many solution already out in the market it is best to instead use them as underlying layer and build applications services using it such as callcentre or CRM services etc .
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.
If you have a hosted IPPBX or PBX in your data-centre or on premise and you need controlled but heavy outflowing traffic, it is a good idea to integrate a resilient and efficient SBC to provide seamless interconnectivity.
For an enterprises such as an Trading floor or warehouse with multiple phone types , softphones , hardphones , turrets etc distributed across various geographies and zones a device agnostic architectural setup is prime . Listing the essentials for setting up such a system. Note supplementary services are data-services , logging , licensing etc are important but kept out of scope to keep focus on functional aspects .
An enterprise application usually is structured in tiers or layers
Client tier – the networks clients communication to the central java programs . Runs on client machines
web tier – state full communication between client and business tier . Runs in server machine.
business tier- handles the logic of the application. The business tier uses the Enterprise Java Bean (EJB) container, which manages the execution of the beans
data tier – encompasses DB drivers . Runs on separate machines for database storage
Event services for Line status notifications
providers lines status notification across enterprise for inter zone and softphone to hardphone .
routing calls within enterprise and hardphone sites read more about resource zones later in the article
Call Control Manager (CCM)
consolidated set of all service and component that make up the VOIP platform besides media handlers . It includes SIP adapters , bridge managers , call processing frameworks , API frameworks , healthchecks etc .
Call processing framework ( CPF)
signalling and call routing logic , mostly in SIP and trunks . Manages identities such as Call Line information , Called Party Information , line status etc in shared memory.
Multiple shared Lines and their statuses
Incases where there is a need to process multiple calls from a single User agent device such as a softphone or hardphone ( common scenario for a turret phone) , the design involves assigning it multiple sip uris and each sip uri will establish a line.
When caller calls callee , the line is said to be BUSY , otherwise said to be IDLE. Transition of a shared sip line from IDLE to BUSY is transmitted to others via SIP PUBLISH as other UAs holding the same sip
Similarly any other event like transfer is propagated to other via SIP UPDATE
Clustering Call control managers (CCM)
A Call Communication manager (CCM) from various zones should be able to cowork on call and session management and advanced features such as routing from home guest zone to home zone , call transfer , refer , barge etc. Designing a clustered setup will also provide elasticity , fail-over and high availability. Can use clustered , HA compliant framework such as Oracle Communication Application Server , suited for enterprise level deployments.
Call Replication and distributed memory management
A node will store two types of data: active sessions and passive sessions. The active sessions are used by the node and stored in cache. The passive sessions are the replicas from the other nodes’ active sessions. The passives sessions are stored on a persistent storage.
Controlling Line Calls using AOR and Resource Zones
When dealing with many SIP endpoints , now referred to as resource, it is best to assign the resources to their respective zones. Thus a resource’s status updates will be only updated by its active resource zone while can be read by any resource zone.
Incoming request Zone vs Active Resource Zone
For an Incoming request such a INVITE , check whether the zone sending the request is its active resource zone or not .If the Active Resource Zone is the same zone on which the INVITE came in, then the call is handled by that zone. If the Active Resource Zone is a different zone, then the call needs to be forwarded to the Active Resource Zone.
Bridges for Local Media connections
Although call signalling is handled by a resources active resource zone only, we can still create media bridges in local zone of the resource .
Local MM bridges are used to auto answer an incoming sip line call and create trunk , especially from hardphones which do not support provisional responses.
Interzone proxy Handler
proxies call control messages between active and non active resource zones. Primarily mapping the sip messages with all custom headers inbetween the communication device interfaces.
Dial Trunk using multiple dedicated sip lines and connect via Media Bridge
To save up on call routing /connection time and to support te ability to add as many users on call at runtime , a dedicated media bridge is established for every call.
A sip line activated is auto-answered by MM , creates a trunk and waits for other endpoint to join the bridge. The flow is as follows :
As INVITE arrives for an IDLE sip line , it is connected to a trunk and auto answered by a local MM bridge .
Since the call is already answered , when caller dials number for callee , collect the DTMF digits over RTP using RFC 2833 DTMF events.
Run inter-digit timer for digit collection and detect end of dialing on timeout.
The dialed trunk connection is made and call is added to media bridge
When provisional responses are received on the trunk connection, generate in-band call progress tones (ringing, proceeding etc) via the MM
When the line answers, the progress tones have to be stopped and the called party gets bridged to the calling party via the media bridge.
Call Diversion involves forwarding calls from zone to another zone. joinjed parties get call UPDATE status and forward response .
Call barge is the processing of joining an ongoing call . The barge event is usually propagated to joined parities via SIP INFO. Private lines do not allow barge in and are exclusively reserved for only few users.
Hold-Resume and Music on Hold in multi-line evironment
While a regular p2p call involves simple reinvite based hold and resume with varrying SDP, the scenario is slightly more detailed for hold resume on bridged trunk connection , as explained below.
As the calls made are on bridge , a hold signal involves a RE-INIVITE with held-SDP to media manager (MM). If hold status on trunk is 200 OK the hold status will be sent to other call interfaces connected on the trunk. Else if hold is denied ,403 is sent back to hold-initiates.
Music on hold is an one way RTP mostly from media server.
For a bridged scenarios , separate Music on hold bridges are kept on Media Managers. When an UA has to hold , it is removed from original bridge and place on music on hold bridge . To be unhold/ resume it is placed back into the orignal bridge from music on hold bridge .
user initiates conference, the conference feature can execute on the zone where the user was logged on, irrespective of zones where the other conference attendees join from . The Call processing framework of originators zone completes the SDP exchange to establish two-way speech path among all the parties.
Incases there are multiple connections from a zone , a local MM conference bridge can be created for them which would connect back to originators MM conf bridge . this two part conf bridge will be transparent to the sip line sand users .
For provisioning inputs and settings setup a Diagnostics , Administration and Configuration platform which can process APIs for data services , licences , alarms or do remote device control such as using SNMP
Session Border Controllers (SBC)
At network level SBC operations include
bridging multiple interfaces in different networks even between the IPv4 and IPv6 networks
auto NAT discovery and STUN
protocol conversion such as TLS to UDP etc
Flood detection and IP filtering
For SIP specific functionalities , SBC does
SIP validation involving checks on syntax and message contents also consistency checks are performed.
stateful and call aware. tracing, monitoring and checking for validitya and health of all the SIP messages
Codec filtering , reordering , media pinning, transcoding, or call recording
Data replication brings High Availability (HA) with hot backups or even Active-Active solutions.
Traffic sharing and routing roles of SBC can include
IP-based and Digest-based authentication
limiting traffic by number of concurrent calls or calling rate.
Dialplan and/or Custom routing
Dispatching/Load-balancing to a backend cluster of servers
SBC’s can be physical hardware boxes or software based applications, as the name suggests their purpose is to control the session at border between the enterprise and external service provider.
SIP to PSTN – SIP is an IP protocol whereas PSTN is a TDM one , achieving interoperability is also the KRA of an SBC
SIP trunking – SBC provide a secure sip connectivity to connect calls to sip trunks which provide bulk calls functionality at a flat pricing.
support for various fixed or mobile endpoints – SBC ensure they are RFC compliant and can extend SIP to any kind of telecom endpoint like PSTN , GSM, fax , Skype , sipphone , IP phones etc.
NAT / Network address translator – To meet the packet routing challenges across a firewall or even during private -public mapping. A combo of DHCP servers and NAT provider comes very handy to reroute or perform hole punching such that signalling and media packets are not dropped and meet the required endpoint. More about NAT here – NAT traversal using STUN and TURN.
Load balancing – Reverse proxies and Load balancers is a much adopted industry practise to mask the inner IPs of the VoIP platform and also route traffic appropriately between control and media server .
Security , QoS and Regulatory compliance – since SBCs are required to typically support a large array of clients they adhere to regulatory and industry accepted standards ,which also involves security features like AAA, TLS/SSL and other means for quality of assurance like logging and fault detection, preventing DDoS etc . In many cases SBC can also encrypt / decrypt RTP streams for probing , tapping or lawful inspection .
Terminating at carriers , PSTN and IP gateways
Additional SBC features
Inaddition to above it is good to have if an SBC provides extra features like forking , emergency number dialing ( 911 ) or active directory integration . Real Time Analysis and monitoring of call and metrics are also expected from a SBC since they reside on edge of the network and are more vulnerable to threats . For example Dialogic Mediant SBC’s and gateways , Audio Codes SBCs
With the shift from on premise PBXs to cloud based VM or microservice architecture , SBC vendors adopt a lager umbrella of services also including automation scripts for checks , reporting tools / consoles , developer friendly APIs to manage sessions via SBC and even WebRTC gateways to connect browser endpoints .
Any VOIP dependant system which deals with bulksome voice / video traffic from external endpoints is a usages scenarios. Listing few
Contact Call centres
Remote work / offsite monitoring
CRM solution for sales/marketing
Connecting webrtc click to dial from webpage to enterprise representatives
connecting enterprise UCC clients to PSTN endpoints
Market trends are really not in favor of Telecom Service /providers with increasing use of OTT ( Over The Top ) application like watsapp , Facebook messenger , Google hangouts , skype , viber , etc .
OTT ( Over The Top ) Applications
What is an OTT ?
An Over The Top ( OTT ) application is one which provides communication services over Internet . Therefore these bypass the communication billing system setup by a Telecom Operator , resulting in no gain or loss of revenue to Telecom Operator who is providing the Internet service to user in first place .
Hence we see that OTT are major threat and concern for Telecom Operators whose traditional and obviously expensive ( when compared to OTTs free service ) billing models are facing disruption .
Telecom Regulatory bodies around the world
The telecom regulatory authorities in some of the countries are for example listed as :
Canadian Radio-television and Telecommunications Commission (CRTC) – Canada
Ministry of Information Industry (MII) – China
Autorité de Régulation des Communications Électroniques et des Postes (ARCEP) – France
Bundesnetzagentur (BNA) – Germany
Telecom Regulatory Authority of India (TRAI) – India
Ministry for Communications and Informatization of the Russian Federation (Minsvyaz) – Russia
Infocomm Development Authority of Singapore (IDA) – Singapore
Independent Communications Authority of South Africa (ICASA) – south Africa
Federal Communications Commission (FCC) , National Association of Regulatory Utility Commissioners (regulators of individual states) (NARUC) , CTIA – The Wireless Association (CTIA) – USA
Such telecom regulatory bodies get to decide whether to enforce differential price to end consumers for using OTT so that telecom service providers can benefit or keep the Internet fair and open by passing Net Neutrality Laws and Bills and amendments .
what is Net Neaurality ?
The fundamental principle of Net Neurality is that Telecom Operators should not block , slow down or charge consumers extra for using other services as their means of communication. This states that it is wrong to charge users above the regular data rates for using VOIP apps and other internet based communication services.
The following counteries have adopted principles of Net Neutrality by passing bills or making law .
Chile – Chile’s General Law of Telecommunications, “No [ISP] can block, interfere with, discriminate, hinder, nor restrict the right of any Internet user of using, send, receive, or offer any content, application, or legitimate service through the Internet, as well as any activity or legitimate use conducted through the Internet.”
Brazil – ” Internet Bill of Rights ” makes equal access to internet mandatory in Brazil .
Netherlands – Even European Union has adopted Netherlands’ Net Neutrality amendment which reads “traffic should be treated equally, without discrimination, restriction or interference, independent of the sender, receiver, type, content, device, service or application.”
USA – Citizens make ‘We the People’ platform to ‘Restore Net Neutrality By Directing the Federal Communications Commission (FCC) to Classify Internet Providers as ‘Common Carriers‘. Therefore not allowing them to either throttle speed by paid prioritization , discriminate in pricing or block any broadband access to legal content . Above facts are from this tech.firstpost.com article.
Inspite of the fact that I Support Net Neutrality with all my heart , as a telecom engineer I understand the cost investment made by Telecom operators in providing am efficient communication network to its subscribers ( Access , Network and Application layers ). Therefor I do have my sympathies with the Telcos and to level out the wide ranging conflict between Telcos and ISP ( Internet Service Providers ) , I pen down the following points which reflect the Telecom Operators Problems and also highlight the solutions that can be adopted to counteract the OTT threat .
Depleting revenue for Telco
Messaging – OTT messaging cost operators $13.9 billion, or 9% of message revenue in 2013
Voice – Voice services under threat from VOIP services like Skype, Viber
OTT apps – Voice & Message apps have been the operator’s biggest headache. Its time Operator should launch its own OTT Services
Data Traffic – The utilization is yet to reach its peak. Will face challenges from WiFi access
Critical Pain areas – Erosion of Operator’s revenue from voice and (especially) messaging
Telco’s OTT aPPLICATION
At this stage it is crucial for a telecom Service provider / Operator to enter the Apps market and bring forth a Messenger which is more powerful , interactive and awesome than a OTT application. Fortunately the Operator can always couple this application with his background telecom infrastructure to provide the edge in performance and functionalists .
Road block while developing a OTT application for a Telecom Service Provider :
Investment in Data Network is not being utilized due to lack of service
Reuse of Existing business Logic and extending the service reach across devices and networks is tough
Operator already has full fledged network Infrastructure in Place
Desire for minimum CAPEX while investing in new technologies
compete with OTT players and open new revenue streams is a challenge
Next we find the way of solving the problems and integrating them together to form a Solution .
OTT Application for Telecom Service provider
Introduce new services to benefit from investment on Data Plans and Bandwidth
Expose REST API to enable 3trd party Integration with existing network Infrastructure
Partner with individual OTT players to make new services that do not compete on core competencies like billing etc
Use protocols like SIP that reduce CAPEX and have goto market more quickly
Go for enriched service that lead to better user experience
This writeup outlines the process of creating a OTT application for a Telecom Service Provider . Components for the application include cloud Address Book , Video Chatting , Location share , Contact synchronization ,REST based thin client , OS and device agnostic etc shown in the figure below
telco’s OTT app
The Application is designed to close knit with Operator’s own infrastructure hence the crucial entities like Network Address Book , Location Service are synced and fetched from Backend Network .
OTT application Feature Overview
Smart Address Book
Automatic: Get contacts from Gmail, Facebook
Fast search by first, last name, frequently
Roadmap: View calendar events
Personal: Get image from Gmail and display in contacts list
Share own location during chatting
Get map for calculating the distance between two chat users
Roadmap : Trigger device (say Switch on/off AC before reaching home) from a threshold distance away from home location
Session Based Chat
Voice Input for texting
Presence information of contacts
RoadMap: Legacy message integration
Voice call to mobile
Voice call to PSTN
Video call to other @imAll user
Share images during voice call to other
Compatible with IOS, windows
Can run as native app on ipad
Can run as browser client on windows
RoadMap: native app for android, windows phone,blackberry10
To upgrade the application and provide enganced and enrich service support the I propose the following roadmap.
From plain vanilla voice and video calling ( supported by every other OTT application ) our application should progress towards legacy telecom support whihc included PSTN , GSM , ISDN etc . This requires backbone of telecom network and a good setup for media codec conversion to suit various legacy media codecs .
Road Map from Traditional to New age services
Voice and video calling
Legacy services support like MMS and SMS
Integration with 3rd party Vendors
Give new enriched services like Multilingual support , file transfer , screen-sharing etc
give facility to integrated web plugins for web calling
To keep the interest of customers it is essential that the application be supported on other popular OTT services like skype , Gtalk . for exmaple a caller should be able to make call from Skype / Gtalk to our application .Multilingual capabilities, support for larger protocol spectrum will just act like icing on the cake .
How does it benefit the Operator??
Saves on development cost and time
Device Agnostic OTT Applications
Simplified Service deployment
Saves licensing cost per client
Reuses existing Messaging and Address Book service logic.
Open New Revenue Streams for operator
No separate SIP stack required for the client
Faster Time to Market
Update : At the time of writing this post I did not anticipate the wave of change that bring focus on subjects like “net neutrality” , ” Save the internet” and “free internet” . However I see now that I had described this phenomenon way in advance for my time .
With the fast pace of telecom evolution both towards the access network front ( ie GSM , UMTS , 3G , 4G , LTE , VOLTE ) to core network side ( ie application servers , registrar , proxies , gateway , media server etc ) a CSP ( content service provider ) is trying hard to keep up with the user expectation . The user expects a plethora of services , reduced cost and high speed bandwidth . If this was not enough a CSP also has competition OTT ( Over The Top ) Players who provide communication and messaging for FREE .
You can read on how OTT’s players are disruption the revenue streams of traditional telecom operators and how can Telco’s develop their own OTT app , integrated with their backend system to answer to that challenge here – OTT ( Over the Top ) Communication applications
The following points outline the major business challenges faced by telecom operators today .
Technology Evolution Challenges
The increased data speeds and further more increasing hunger for the data overwhelms the existing network infrastructures.
Ensure uniform service experience across the network technologies to check the customer churn.
Access / Radio Technology independent delivery of services.
Enhance Reuse for exiting investments.
Multiple Service Platform Challenges
Typical network constitutes of Multiple Service Platforms increasing network complexity and integration challenges many fold.
Heterogeneous multiple SDP Solutions typically deployed to cater to Multiple Types of Networks/ Standards/Variants
Service Islands makes introduction of seamless services a challenging task for the CSP
Transport Upgrade and Convergence of Wireless Wireline
Retain investments in copper wire systems while migrating towards next generation Fiber Optic systems.
Severe competition among wire-line and wireless operators to provide latest services to retain subscriber base.
Fixed Mobile Convergence leading to a diminishing gap among the revenue shares of various operators in the space, and leading to losses for wire-line only players.
MPEG 2 MPEG-2 (a.k.a. H.222/H.262 as defined by the ITU) generic coding of moving pictures and associated audio information combination of lossy video compression and lossy audio data compression methods, which permit storage and transmission of movies using currently available … Continue reading →
Audio signals are electronic representations of sound waves—longitudinal waves which travel through air, consisting of compressions and rarefactions and Audio Signal Processing focuses on the computational methods for intentionally altering auditory signals or sounds, in order to achieve a particular goal. … Continue reading →
Interfaces of webrtc and tracks to stream addition Process to perform webrtc handshake 1.Setup Client side for the caller PeerConnectionFactory to generate PeerConnections PeerConnection for every connection to remote peer MediaStream audio and video from client device 2.caller creates SDP … Continue reading →
Web resources are usually build on request/response paradigm such as HTTP , SIP messages . This means that server responds only when a client requests it to. This made web intercations very slow and unsuited for VOIP signalling Long Poll … Continue reading →
Wi‑Fi is a trademark of the Wi-Fi Alliancefamily of radio technologies commonly used for wireless local area networking (WLAN) of devices. Current and older Wifi standards standards operate on varying frequencies, deliver different bandwidth, and support different numbers of channels. … Continue reading →
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 →
Multiplying the capacity of a radio link using multiple transmission and receiving antennas to exploit multipath propagation.Key technology for achieving a vast increase of wireless communication capacity over a finite electromagnetic spectrum. Antenna configuration – implies antenna spatial diversity by … Continue reading →
NLP has great potential in cognitive and artificial intelligence , but also with increasing human to machine interaction and enhancement in Machine learning ,NLP is set to revolutionize the Voice over IP space. Continue reading →
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 →
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