Category: WebRTC SaaS

WebRTC CPaaS ( Communication Platform as a Service )

A CPasS ( communication platform as a service ) is a cloud-based communication platform like B2B cloud communications 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, and network interfaces building a communication platform was a costly affair. Cisco, Facetime, and Skype were the only OTT ( over the top) players taking away from the telco’s call revenue. However, with the advent of standardised, 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 into how CPaaS solutions are architectured and programmed.

SIP based Communication Platform as a Service

Sample CPass Architecture build on open source technologies

Over all Archietcture of Real Time Comunication ecosystem with Media management, CDR , processing pielines , real time analytics.

Data Streams for realtime analytics and telemetrics

I have written an article before on Steps for building and deploying WebRTC solution , which includes standalone, cloud hosted and TURN based NAT handler systems .

A typical CPaaS solution provides

  • 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

Self hosted Datacentre vs Cloud server

Cloud-based infrastructure 

Cloud Services as Amazon Web service, Google Cloud, Microsoft Azure, IBM Cloud, Digital Ocean is great resources to host the multiple parts of a CPaaS system such as gateways, media servers, SIP Application servers, other servers for microservices including accounting, profile management, rest services etc. Often virtualized machines ( VMs) mounted on a larger physical remote datacentre are an ideal choice for VoIP and cloud communication providers.

  • (+) pay as you go
  • (+) no stress on resource management like cooling, rack space , wiring etc
  • (+) easy to setup
  • (-) not in premise, security is not in control
  • (-) outages in cloud infrastructures datacentre could lead to service disruption

Self hosted / inpremises Servers / private cloud

Marinating datacentre provides flexibility to extend and or develop tightly controlled use cases. It is often a requirement for secure communication platforms pertaining to government or banking communications such as turret phones.

Some approaches are to set up the server with Openstack to manage SDN ( software-defined network). Other approaches also involve VMWare to virtualize servers and then using docker container-managed via Kubernetes to dynamically spawn instances of server as load scaled up or down.

  • (+) more secure and controlled
  • (+) no monthly recurring fees to cloud vendors
  • (-) maintenance of racks and servers
  • (-) requires planning for high availability and geographical deployment for redundancy

Using existing SDK vs building your own RTC platform from scratch

I have come across so many small size startups trying to build CPaaS solutions from scratch but only realising it after weeks of trying to build an MVP that they are stuck with firewall, NAT, media quality or interoperability issues. Since there are so many solutions already out in the market it is best to instead use them as an underlying layer and build applications services using it such as call centre or CRM services making custom wrappers.

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 and would potentially try to abstrct away the implementation details from their SDK users or clients.

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.

Call Rate charging and Accounting Services

Call Rates are very critical for billing and charging the users. Any updates from the customer or carriers or individuals need to propagate automatically and quickly to avoid discrepancies and negative margins.

CDR ( Call Detail Record ) processing pipeline

CDRs need to be processed sequentially and incrementally on a record-by-record basis or over sliding time windows.
CDR can also be used for a wide variety of analytics including correlations, aggregations, filtering, and sampling.

Updating rate sheet ( charges per call or per second )

The following setup is ideal to use the new input rate sheet values via web UI console or POST API and propagate it quickly to the main DB via a queuing system such as SQS. Serverless operations such as using AWS lambda can be used via a trigger-based system for any updates. This ensures that any new input rates are updated in realtime and maintain fallback values in separate storage as s3 bucket too

CDR (Call Detail Record ) managemenet, billing, dispute management

In current Voip scenarios a call may be passing thorugh various telco providers , ISP and cloud telephony serviIn current VoIP scenarios, a call may be passing through various telco providers, ISP and cloud telephony service providers where each system maintains its own call records and billing. This in my opinion is duplication and missing a single source of truth. A decentralized, reliable and consistent data store via blockchain coudl potentially maintain the call records making then immutable and non diputable. Some more details on the concept are in the article below.

Blockchain integarted with Voice over IP platforms

Session Border controller for WebRTC

Unified communication services build around WebRTC should be vendor agnostic and multi-tenant and be supported by other Communication Service Providers (CSPs), SIP trunks, PBXs, Telecom Equipment Manufacturers (TEMs), and Communication Platform as a Service (CPaaS). This can happen if all endpoints adhere to SIP standards in most updated RFC. However since not all are on the boat , Session border controllers are a great way to mitigate the differences and provide seamless connectivity to signalling and media , which could be between WebRTC, SIP or PSTN, from TDM to IP .

Session Border Controllers ( SBC )  assist in controlling the signalling and usually also the media streams involved in calls and sessions.

They are often part of a VOIP network on the border where there are 2 peer networks of service providers such as backbone network and access network of corporate communication system which is behind firewall.

A more complex example is that of a large corporation where different departments have security needs for each location and perhaps for each kind of data. In this case, filtering routers or other network elements are used to control the flow of data streams. It is the job of a session border controller to assist policy administrators in managing the flow of session data across these borders. – wikipedia

SBC act like a SIP-aware firewall with proxy/B2BUA.

What is B2BUA?

A Back to back user agent ( B2BUA ) is a proxy-like server that splits a SIP transaction in two pieces:

  • on the side facing User Agent Client (UAC), it acts as server;
  • on the side facing User Agent Server (UAS) it acts as a client.

B2BUAs keep state information about active dialog. Read more here .

Remote Access

SBC mostly have public url address  for teleworkers and a internal IP for enterprise/ inner LAN . This enables users connected to enterprise LAN ( who do not have public address ) to make a call to user outside of their network. During this process SBC takes care of following while relaying packets .

  1. Security
  2. Connectivity
  3. Qos
  4. Regulatory
  5. Media Services
  6. Statistics and billing information

Topology hiding

SBC hides and anonymize secure information like IP ports before forwarding message to outside world . This helps protect the internal node of Operators such as PSTN gateways or SIP proxies from revealing outside.

Explaining the functions of SBC in detail

1. Security

SBCs are often used by corporations along with firewalls and intrusion prevention systems (IPS) to enable VoIP calls to and from a protected enterprise network. VoIP service providers use SBCs to allow the use of VoIP protocols from private networks with Internet connections using NAT, and also to implement strong security measures that are necessary to maintain a high quality of service. The security features includes :

  • Prevent malicious attacks on network such as DOS, DDos.
  • Intrusion detection
  • cryptographic authentication
  • Identity/URL based access control
  • Blacklisting bad endpoints
  • Malformed packet protection
  • Encryption of signaling (via TLS and IPSec) and media (SRTP)
  • Stateful signalling and Validation
  • Toll Fraud – detect who is intending to use the telecom services without paying up

2. Connectivity

As SBC offers IP-to-IP network boundary, it recives SIP request from users like REGISTER , INVITE  and routes them towards destination, making their IP. During this process it performs various operations like

  • NAT traversal
  • IPv4 to IPv6 inter-working
  • VPN connectivity
  • SIP normalization via SIP message and header manipulation
  • Multi vendor protocol normalization

Further Routing features includes  :
Least Cost Routing based on MoS ( Mean Opinion Score ) : Choosing a path based on MoS is better than chooisng any random path . 

Protocol translations between SIP, SIP-I, H.323.

In essence SBC achieve interoperability, overcoming some of the problems that firewalls and network address translators (NATs) present for VoIP calls.

Automatic Rerouting

connectivity loss from UA for whole branch is detected by timeouts . But they can also be detected by audio trough SIP OPTIONS by SBC .  In such connectivity loss , SBC decides rerouting or sending back 504 to caller .

SBC 2 (1)

4. QoS
To introduce performance optimization and business rules in call management QoS is very important . This includes the following :

  • Traffic policing
  • Resource allocation
  • Rate limiting
  • Call Admission Control (CAC)
  • ToS/DSCP bit setting
  • Recording and Audit of messages , voice calls , files
  • System and event logging

5. Regulatory

Govt policies ( such as ambulance , police ) and/ or enterprise policies may require some calls to be holding priority over others . This can also be configured under SBC as emergency calls and prioritization.
Some instances may require communication provider to comply with lawful bodies and provide session information or content , this is also called as Lawful interception (LI) . This enables security officials to collect specific information rather than examining all the traffic that passes through a particular router. This is also part of SBC.
6. Media services

Many of the new generation of SBCs also provide built-in digital signal processors (DSPs) to enable them to offer border-based media control and services such as- DTMF relay , Media transcoding , Tones and announcements etc.

WebRTC enabled SBC’s also provide conversion between DTLS-SRTP, to and from RTCP/RTP. Also transcoding for Opus into G7xx codecs
and ability to relay VP8/VP9 and H.264 codecs.

7. Statistics and billing information

SBC have an interface with and OSS/BSS systems for billing process , as almost all traffic that pass through the edge of the network passes via SBC. For this reason it is also used to gather Statistics and usage-based information like bandwidth, memory and CPU.  PCAP traces of both signaling and media information of specific sessions .

New feature rich SBCs also have built-in digital signal processors (DSPs). Thus able to provide more control over session’s media/voice . They also add services like Relay and Interworking, Media Transcoding, Tones and Announcements, DTMF etc.

Session Border Controller (SBC)
Session Border Controller for WebRTC , SIP , PSTN , IP PBX and Skype for business .

Diagram Component Description

Gateways provide compression or decompression, control signaling, call routing, and packetizing.

PSTN Gateway : Converts analog to VOIP and vice versa . Only audio no support for rich multimedia .

VOIP Gateway : A VoIP Gateway acts like a translator converting digital telecom lines to VoIP . VOIP gateway often also include voice and fax. They also have interfaces to Soft switches and network management systems.

WebRTC Gateway : They help in providing NAT with ICE-lite and STUN connectivity for peers behind policies and Firewall .

SIP trunking : Enterprises save on significant operation cost by switching to IP /SIP trunking in place of TDM (Time Division Multiplexing). Read more on SIP trunk and VPN  here. 

SIP Server : A Telecom application server ( SIP Server ) is useful for building VAS ( Value Added Services ) and other fine grained policies on real time services . Read more on SIP Servers here . 

VOIP/SIP service Provider :   There are many Worldwide SIP Service providers such as Verizon in USA , BT in europe, Swisscom in Switzerland etc .

Building a SBC

The latest trends in Telecommunications industry demand an open standardized SBC to cater to growing and large array of SIP Trunking, Unified Multimedia Communications UC&C, VoLTE, VoWi-Fi, RCS and OTT services worldwide . Building an SBC requires that it meet the following prime requirements :

  • software centric
  • Cloud Deploybale
  • Rich multimedia (audio , video , files etc) processing
  • open interfaces
  • The end product should be flexible to be deployed as COTS ( Commercial Off the shelf) product or as a virtual network function in the NFV cloud.
  • Multi Configuration , should be supported such as Hosted or Cloud deployed .
  • Overcome inconsistencies in SIP from different Vendors
  • Security and Lawful Interception
  • Carrier Grade Scaling

Flow Diagram 

SBC WebRTC to SIP

Thus we see how SBC became important part of comm systems developed over SIP and MGCP. SBC offer B2BUA ( Back to Back user agent) behavior to control both signalling and media traffic.

Setting up ubuntu ec2 t2 micro for webrtc and socketio

Setting up a ec2 instance on AWS for web real time communication platform over nodejs and socket.io using WebRTC.

Primarily a Web Call  , Chat and conference platform uses WebRTC for the media stream and socketio for the signalling . Additionally used technologies are nosql for session information storage , REST Apis for getting sessions details to third parties.

Below is a comprehensive setup if ec2 t2.micro free tier instance, installation with a webrtc project module and samples of customisation and usage .

Technologies used are listed below :

Server

  1. ec2 instance t2.micro covered under free tier
  2. domain name
  3. SSL certificate

Core module for Web Calling feature

  1. WebRTC
  2. Node.js
  3. socket.io

UI components

  1. javascript
  2. css
  3. html5
  4. bootstrap
  5. jquerry

Supporting setup for session management

  1. Code version-ing  and maintenance
  2. git
  3. npm

Sample Project https://github.com/altanai/webrtcdevelopment

Amazon’s free tier ec2

Amazon EC2 : These are elastic compute general purpose storage servers that mean that they can resize the compute capacity in the cloud based on load . 750 hours per month of Linux, RHEL, or SLES t2.micro instance usage. Expires 12 months after sign-up.

Some other products are also covered under free tier which may come in handy for setting up the complete complatorm. Here is a quick summary

Amazon S3 : it is a storage server. Can be used to store media file like image s, music , videos , recorded video etc .

Amazon RDS : It a relational database server . If one is using mysql or postgress for storing session information or user profile data . It is good option .

Amazon SES : email service. Can be used to send invites and notifications to users over mail for scheduled sessions or missed calls .

Amazon CloudFront : It is a CDN ( content delivery network ) . If one wants their libraries to be widly available without any overheads . CDN is a good choice .

Alternatively any server from Google cloud , azure free tier or digital ocean or even heroku can be used for WebRTC code deployment . Note that webrtc capture now requires htps in domain name.

Server Setup

Set up environment by installing nvm  , npm  and git ( source version control)

1. NVM ( node version manager )

cURL: 

curl -o- https://raw.githubusercontent.com/creationix/nvm/v0.31.1/install.sh | bash

or Wget:

wget -qO- https://raw.githubusercontent.com/nvm-sh/nvm/v0.37.2/install.sh | bash

To check installation

command -v nvm
nvm

2. NPM( node package manager)

sudo apt-get install npm
Screenshot from 2016-05-16 12-41-42

2. Git

sudo apt-get install git
Screenshot from 2016-05-17 11-25-01

 SSL certificates

Since 2015 it has become mandatory to have only https origin request WebRTC’s getUserMedia API ie Voice, video, geolocation , screen sharing require https origins.
Note that this does not apply to case where its required to only serve peer’s media Stream or using Datachannels . Voice, video, geolocation , screen sharing now require https origins

For A POC purpose here is th way of generating a self signed certificate
Transport Layer Security and/or Secure Socket Layer( TLS/SSL) is a public/private key infrastructure.Following are the steps

1.create a private key

 openssl genrsa -out webrtc-key.pem 2048

2.Create a “Certificate Signing Request” (CSR) file

 openssl req -new -sha256 -key webrtc-key.pem -out webrtc-csr.pem

3.Now create a self-signed certificate with the CSR,

 openssl x509 -req -in webrtc-csr.pem -signkey webrtc-key.pem -out webrtc-cert.pem

However in production or actual implementation it is highly recommended to use a signed certificate by CA as For examples include

Web Server

create https certificate using self generate or purchased SSL certificates using fs , node-static and https modules . To know how to create self generated SSL certificates follow section above on SSL certificates.

var fs = require(‘fs’);
var _static = require(‘node-static’);
var https = require(‘https’);

var file = new _static.Server("./", {
cache: 3600,
gzip: true,
indexFile: "index.html"
});

var options = {
key: fs.readFileSync(‘ssl_certs/webrtc-key.pem’),
cert: fs.readFileSync(‘ssl_certs/webrtc-cert.pem’),
ca: fs.readFileSync(‘ssl_certs/webrtc-csr.pem’),
requestCert: true,
rejectUnauthorized: false
};

var app = https.createServer(options, function(request, response){
request.addListener(‘end’, function () {
file.serve(request, response);
}).resume();
});

app.listen("8080");

Web servers work with the HTTP (and HTTPS) protocol which is TCP based. As a genral rule TCP establishes connection whereas UDP send data packets

Scoketio signalling server as npm

Socket.io determines which of the following real-time communication method is suited to the particular client and its network bandwidth .

  • WebSocket
  • Adobe Flash Socket
  • AJAX long polling
  • AJAX multipart streaming
  • Forever Iframe
  • JSONP Polling

The socket.io server needs a HTTP Server for initial handshake. The general steps for socketio signalling server

1.require socket.io and keep the reference

var io = require(‘socket.io’)

2.Create your http / https server outline in section on webserver

3.bind your http and https servers (.listen)

io.listen(app, {
    log: false,
    origins: ‘*:*’
})

4. Optionally set transport 

io.set(‘transports’, [
‘websocket’
])

5.setup io events 

io.sockets.on(‘connection’, function (socket) {
//Do domething
})

Note that Socket.io or websockets require an http server for the initial handshake.

Install ssocketio npm module

npm install socket.io

Complete code for signalling server

const io = require("socket.io")
    .listen(app, {
        log: false,
        origins: "*:*"
    });
io.set("transports", ["websocket"])

var channels = {};

io.sockets.on("connection", function (socket) {

    console.log("connection");
    var initiatorChannel = "";

    if (!io.isConnected) {
        io.isConnected = true;
    }

    socket.on("namespace", function (data) {
        onNewNamespace(data.channel, data.sender);
    })

    socket.on("new-channel", function (data) {
        if (!channels[data.channel]) {
            initiatorChannel = data.channel;
        }
        console.log("new channel", data.channel, "by", data.sender)
        channels[data.channel] = {
            channel: data.channel,
            users: [data.sender]
        };

    })

    socket.on("join-channel", function (data) {
        console.log("Join channel", data.channel, "by", data.sender)
        channels[data.channel].users.push(data.sender);
    })

    socket.on("presence", function (channel) {
        var isChannelPresent = !!channels[channel.channel];
        console.log("presence for channel ", isChannelPresent)
        socket.emit("presence", isChannelPresent)
    })

    socket.on("disconnect", function (channel) {
        // handle disconnected event
    })

    socket.on("admin_enquire", function (data) {
        switch (data.ask) {
            case "channels":
                socket.emit("response_to_admin_enquire", channels)
                break;
            case "channel_clients":
                socket.emit("response_to_admin_enquire", io.of("/" + data.channel).clients());
                break;
            default :
                socket.emit("response_to_admin_enquire", channels)
        }
    })
})



function onNewNamespace(channel, sender) {
    console.log("onNewNamespace", channel);
    io.of("/" + channel).on("connection", function(socket) {
        var username;
        if (io.isConnected) {
            io.isConnected = false;
            socket.emit("connect", true)
        }
    
        socket.on("message", function (data) {
            if (data.sender == sender) {
                if(!username) username = data.data.sender;
                socket.broadcast.emit("message", data.data)
            }
        })
    
        socket.on("disconnect", function() {
            if(username) {
                socket.broadcast.emit("user-left", username)
                username = null;
            }
        })
    })
}

WebRTC main HTML5  project

This is the front  end section of the whole exercise . It contains JavaScript , css and html5 to make a webrtc call

<body id="pagebody">
<div id="elementToShare" className="container-fluid">
    <!-- ................................ top panel ....................... -->
    <div className="row topPanelClass">
        <div id="topIconHolder">
            <ul id="topIconHolder_ul">
                <li hidden><span id="username" className="userName" hidden>a</span></li>
                <li hidden><span id="numbersofusers" className="numbers-of-users" hidden></span></li>
                <li><span id="HelpButton" className="btn btn-info glyphicon glyphicon-question-sign topPanelButton"
                          data-toggle="modal" data-target="#helpModal"> Help </span></li>
            </ul>
        </div>
    </div>
    <!-- .............alerts................. -->
    <div className="row" id="alertBox" hidden="true"></div>
    <!-- .......................... Row ................................ -->
    <div className="row thirdPanelClass">
        <div className="col-xs-12 videoBox merge" id="videoHold">
            <div className="row users-container merge" id="usersContainer">
                <div className="CardClass" id="card">

                    <!-- when no remote -->
                    <div id="local" className="row" hidden="">
                        <video name="localVideo" autoPlay="autoplay" muted="true"/>
                    </div>
                    <!-- when remote is connected -->
                    <div id="remote" className="row" style="display:inline" hidden>
                        <div className="col-sm-6 merge" className="leftVideoClass" id="leftVideo">
                            <video name="video1" hidden autoPlay="autoplay" muted="true"></video>
                        </div>
                        <div className="col-sm-6 merge" className="rightVideoClass" id="rightVideo">
                            <video name="video2" hidden autoPlay="autoplay"></video>
                        </div>
                    </div>
                </div>
            </div>
        </div>
    </div>
    <!--modal help -->
    <div className="modal fade" id="helpModal" role="dialog">
        <div className="modal-dialog modal-lg">
            <div className="modal-content">
                <div className="modal-header">
                    <button type="button" className="close" data-dismiss="modal">&times;</button>
                    <h4 className="modal-title">Help</h4>
                </div>
                <div className="modal-body">
                    WebRTC Runs in only https due to getusermedia security contraints
                </div>
                <div className="modal-footer">
                    <button type="button" className="btn btn-default" data-dismiss="modal">Close</button>
                </div>
            </div>
        </div>
    </div>
</div>
</body>

the document start script that invokes the JS script 

$('document').ready(function () {

    sessionid = init(true);

    var local = {
        localVideo: "localVideo",
        videoClass: "",
        userDisplay: false,
        userMetaDisplay: false
    };

    var remote = {
        remotearr: ["video1", "video2"],
        videoClass: "",
        userDisplay: false,
        userMetaDisplay: false
    };

    webrtcdomobj = new WebRTCdom(
        local, remote
    );

    var session = {
        sessionid: sessionid,
        socketAddr: "https://localhost:8084/"
    };

    var webrtcdevobj = new WebRTCdev(session, null, null, null);

    startcall();
});


Screenshot from 2016-05-17 12-12-37.png

Common known issues:

1.Opening page https://<web server ip>:< web server port>/index.html says insecure

This is beacuse the self signed certificates produced by open source openSSL is not recognized by a trusted third party Certificate Agency.
A CA ( Certificate Authority ) issues digital certificate to certify the ownership of a public key for a domain.

To solve the access issue goto https://<web server ip>:< web server port> and given access permission such as outlined in snapshot below

image

2.Already have given permission to Web Server , page loads but yet no activity .

if you open developer console ( ctrl+shift+I on google chrome ) you will notice that there migh be access related errros in red .
If you are using different server for web server and signalling server or even if same server but different ports you need to explicity go to the signalling server url and port and give access permission for the same reason as mentione above.

3.no webcam capture on opening the page

This could happen due to many reasons

  •  page is not loaded on https
  • browser is not webrtc compatible
  • Media permission to webcam are blocked
  • the machine does have any media capture devices attached
  •  Driver issues in the client machine while accessing webcams and mics .

4.socketio + code: 0, message: “Transport unknown”

Due to the version  v1.0.x of socket.io while performing handshake . To auto correct this , downgrade to v0.9.x

Steps for building and deploying WebRTC solution

Step 1  : USE Local machine to test the client server WebRTC funcationality

Pick any WebRTC API and run its demos . It works kool . download and run in local-machine with nodejs server . Awesome . Everything is Awesome !!

You can learn more about some WS based WebRTC API here:  https://altanaitelecom.wordpress.com/2014/12/02/current-state-of-webrtc/.

If you are a diehard telecom engineer and only want SIP based WebRTC solutions go here : https://altanaitelecom.wordpress.com/2014/07/16/interoperability-between-webrtc-sip-phones-and-others/

Steps for building and deploying WebRTC solution Step 1 : Pick a WebRTC API and run locally ( ie open 2 browsers and run on local machine )
Steps for building and deploying WebRTC solution
Step 1 : Pick a WebRTC API and run locally ( ie open 2 browsers and run on local machine )

Step 2 : Use cloud Server and different client Browsers  

Now what good is it doing to anyone if its running locally on my machine with addresses like localhost and 127.0.0.1  . Let us put it on the cloud and at-least let my colleague / friends enjoy it .  Cloud Web Server and Nodejs signalling server . That is okay use amazon’s Ec2. works for most of the people most of the time .

Steps for building and deploying WebRTC solution Step 2 : Put Server on cloud and WebRTC clients on different machine
Steps for building and deploying WebRTC solution
Step 2 : Put Server on cloud and WebRTC clients on different machine

Here is when we discover the issues of ICE ( Interactive Connectivity Establishment ) I have mentioned this in detail on the post NAT Traversal using STUN and TURN .  Briefly ICE helps us in coping up with NAT ( Network Address Traversal and Firewalls ) .

Note that this step only works if everyone you want to connect to is either on same intranet or on public internet without and UDP blocks / firewalls / restriction .

As we try to connect 2 WebRTC clients from different machine and different networks we find that network address from client’s OS and network card fails to connect to Signalling Server due to either Firewalls issues or other Network policies . We therefore use a STUN server to map the private IP to a publicly accessible IP that will help in completing the signalling

The Signalling is establishes using a STUN server for address mapping and NAT . One can use google’s default STUN server stun.l.google.com:19302. Easy and free .

Steps for building and deploying WebRTC solution Step 2.1 : Put Server on cloud and WebRTC clients on different machine + STUN for address discovery ( NAT traversal )
Steps for building and deploying WebRTC solution
Step 2.1 : Put Server on cloud and WebRTC clients on different machine + STUN for address discovery ( NAT traversal )

There you go everything is looking good from here now , both peers join the session successfully  , but the video may appear black . This is so because the media under most inter network conditions fails to flow between private and public network .

This is where step 3 comes into picture ie using a TURN ( media relay ) server .

Step 3: TURN server to Call people in a inter-network fashion 

Sure the architecture I have setup is bound to work everywhere where the network is open and public . However error in connectivity , errors in console , blank video are the problems that might appear when one tries to connect from private to public connections.

To bypass network firewalls , corporate net policies , UDP blocks and filters we require a TURN server which help in media traversal across different networks in a relay mechanism.

Now we have 3 options to choose from

1.  Use a wildly popular http://numb.viagenie.ca/

2. Build your own TURN server with RFC 5766 ( COTURN )  , or rather easier would be to use any open source TURN server code available in Github.

3. Pay and use a commercial TURN service provider or you can even use their trail version to see if things work out for you ( example Xirsys)  .

Remember you can use any TURN service it does not affect your WebRTC API functionality . All we need to do is add it to Peerconnection configuration like

&lt;/address&gt;&lt;address&gt;peerConnectionConfig: {<br>
iceServers:[&lt;/address&gt;&lt;address&gt;{"url": &lt; stunserver address &gt;},&lt;/address&gt;&lt;address&gt;{"username":"xx","url":&lt; turn server address&nbsp;transport=udp&gt;,"credential":"yy"},&lt;/address&gt;&lt;address&gt;{"username":"xx","url":&lt; turn server address transport=tcp&gt; , "credential":"yy"}]&lt;/address&gt;&lt;address&gt;},&lt;/address&gt;&lt;address&gt;

There we go , now anyone from anywhere should be able to use our WebRTC setup for making audio , video calls or just exchanging data via DataChannel ( like screen-sharing , file transfer , messages , playing games , collaborative office work etc )  .

Steps for building and deploying WebRTC solution TURN based media Relay for WebRTC traffic
Steps for building and deploying WebRTC solution
TURN based media Relay for WebRTC traffic

The setups covers scenarios wherein user is on office corporate network , home network , mobile network , no problem as long as he / she has a webrtc enables browser ( read Chrome , Mozilla , Opera ) .

It is noteworthy that ideally voice should be traversing on TCP while video and data can go around in UDP however unless restrained the WebRTC API’s self determine the best protocol to route the packets / stream .

Debug helper

Common issues around media playback

  • DOMException: The play() request was interrupted by a new load request
  • webrtcdevelopment_min.js:1 [Violation] Only request notification permission in response to a user gesture.

Read more about best WebRTC frameworks and code in this book

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/