Calls from the website for Genesys and Cisco

From this moment, customers who visit your site can call you directly using only a computer, just place a button on the site “Call me from the site.”

It’s quite natural, when a customer, during choice a product on the site or trying to make an active operation on a card, wants to be able to communicate with the company directly from the site – by telephone call.

Traditionally, in WEB-browsers on the PC there was no built-in support for audio / video calls from the site, so all had to use third-party plug-ins, and on mobile platforms (Android, iOS) all had to develop their own mobile application.

Limitations of traditional solutions

Large App sizes (for downloading via the Internet), with big requirements to resources and slowness

Large App sizes (for downloading via the Internet), with big requirements to resources and slowness

Permanent security problems with plug-ins

Permanent security problems with plug-ins

Difficulties in installation (neccessary rights on the PC, compatibility, etc.)

Difficulties in installation (neccessary rights on the PC, compatibility, etc.)

No support for all platforms

No support for all platforms

Regular updates needed

Regular updates needed

Restrictions when working with NAT/Proxy

Restrictions when working with NAT/Proxy

WebRTC standard as an alternative solution

This is a non-commercial project that allows you to use a modern web browser as a full-fledged unified communications terminal using simple JavaScript

This is a non-commercial project that allows you to use a modern web browser as a full-fledged unified communications terminal using simple JavaScript

WebRTC allows you to make a call from the site (voice / video calls) inside the browser without using any plug-ins or third-party clients

WebRTC allows you to make a call from the site (voice / video calls) inside the browser without using any plug-ins or third-party clients

WebRTC is an API developed by the World Wide Web Consortium (W3C) for 7 years

WebRTC is an API developed by the World Wide Web Consortium (W3C) for 7 years

Built-in WebRTC security

Built-in WebRTC security

Another problem faced by many companies with a long history is that: they already have a Contact Center or telephony solution that operators use to handle calls and it is undesirable to deploy one more solution for the web-calls service from the site.

Therefore, SMARTNOVA LIMITED has developed a solution that integrates with your current contact center using standard SIP technology, which automatically allows the contact center operator to service the sip call from the site in a single window with traditional calls, and the manager to receive statistics or to listen to calls in the same reporting and recording systems as traditional calls do.

The “call from the site” service has successfully proven in the integration with

Genesys logo

Genesys PureConnect

Genesys Engage

Cisco logo

Cisco Unified Contact Center Enterprise

Cisco Unified Contact Center Express

Advantages for your Customers

Service call from the site is free

Service call from the site is free

Direct connection to an operator groups (without the IVR) and, as a result, a quick and effective solution of the issue

Direct connection to an operator groups (without the IVR) and, as a result, a quick and effective solution of the issue

An easy and safe way to get contact center support anywhere in the world

An easy and safe way to get contact center support anywhere in the world

Works with almost any device

Works with almost any device

No need to install plug-ins or third-party software

No need to install plug-ins or third-party software

Advantages for the Company

Webcall service embeds communication opportunities on the Company’s web page

Webcall service embeds communication opportunities on the Company’s web page

For the Contact Center, the WebRTC call will pass transparently (like a normal phone call). As a result, a unified queue for receiving multimedia messages from customers and unified reporting

For the Contact Center, the WebRTC call will pass transparently (like a normal phone call). As a result, a unified queue for receiving multimedia messages from customers and unified reporting

Reducing the “0800” and short dialing numbers costs to the contact center

Reducing the “0800” and short dialing numbers costs to the contact center

Low consumption of hardware resources in comparison with analogues

Low consumption of hardware resources in comparison with analogues

Authentication and extended information about the user and call context

Authentication and extended information about the user and call context

DTMF

DTMF transmission and the ability to interact with IVR services

Solution architecture (Cloud Network Sheme) for Genesys

Solution architecture (Cloud Network Sheme) for Genesys

Solution architecture (InHouse deployment option) for Cisco

Solution architecture (InHouse deployment option) for Cisco

Solution architecture (cloud-base option) for Cisco

Solution architecture (cloud-base option) for Cisco

Security of “call from the site” service

Encrypted Voice traffic (SRTP)

Encrypted Voice traffic (SRTP)

Signal traffic is transmitted via HTTPS. Protected by the TLS encryption protocol

Signal traffic is transmitted via HTTPS. Protected by the TLS encryption protocol

SRTP (Secure RTP) – Voice traffic is protected using encryption and authentication algorithms AES_CM_128_HMAC_SHA1_80

SRTP (Secure RTP) – Voice traffic is protected using encryption and authentication algorithms AES_CM_128_HMAC_SHA1_80

The possibility of built-in authentication

The possibility of built-in authentication

SRTCP provides feedback on lost packets and statistics

SRTCP provides feedback on lost packets and statistics

Modern browsers – reliable and secure applications

Modern browsers – reliable and secure applications

Ready to get started? Contact us now!

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Supported codecs and protocols for “call from site” service to sip

Client support for NAT / Firewall

Call service Nova.Webcalls provides the ability to exchange media traffic between the server and the client in case they are behind NAT and Firewall

Call service Nova.Webcalls provides the ability to exchange media traffic between the server and the client in case they are behind NAT and Firewall

The service also provides support for client in roaming, when during the established connection the client device has switched to another network (switching WiFi to 3g, for example

The service also provides support for client in roaming, when during the established connection the client device has switched to another network (switching WiFi to 3g, for example

Supported audio codecs

Narrowband audio: G.711

Narrowband audio: G.711

Wideband audio: Opus

Wideband audio: Opus

Supported video codecs

H.264

H.264