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WebRTC to SIP

ExpertFlow WebRTC

Providing Video solutions  for the web users. Using this solution, a web user can initiate a video call request to the Cisco Contact Center Express  (UCCX) and Enterprise (CCE) during an web chat session of Expertflow Hybrid Chat solution.

Communication Flow


  1. A web user wants to connect  to Cisco Contact Center agent  via WebRTC  Video without having to download or install any software 
  2. This user request  will initiate a call from Customer WebRTC to Cisco Call Center agent
  3. This WebRTC is converted by Expertflow’s WebRTC -SIP gateway to a SIP call, Which is then  routed to Cisco Cubes/Gateway from where the routing will be handled by the Cisco end.
  4.  The initiating web user will be asked to allow system resources (mic, camera) access from his WebRTC capable  web browser
  5. Upon access grant, the video call will be started.

The call is encrypted via WebRTC from the customer’s browser to the EF WebRTC-SIP gateway, and from the gateway to Cisco CUBE using SIP with TLS.




Solution Prerequisite 

Following are the mandatory prerequisites for a smooth installation of the solution up to 50 agents.

​Hardware Requirements

Item

Minimum requirement

CPU

8 cores

RAM

8 GB

Disk

250 GB


​Software Requirements

Item

Minimum requirement

OS

Debian 10

Node

Latest


TLS Requirements 

Certificates from a valid signing authority or Domain signed certificate required for https protocol support.


Port Utilization For Product Installation

We will require Internet access for doing the installation of Expertflow WebRtc Server.

For Product connectivity with internal and external components

Ports mentioned in this section should be open for the mentioned product connectivity with different internal and external components.




Source Host

Destination Host

Source Port

Destination Port

Communication Protocol

Scope

Description

Web Audio Call/video Call

WebRTC Server

any

7443 

WSS 

Public


Web Audio Call/video Call

WebRTC Server

any

8021

TCP

Public 


Web Audio Call/video Call

WebRTC Server

any

3000

HTTPS 

Public 



Ports enablement between EF WebRTC Server to Cisco Cube

WebRTC Server

Network Protocol

Application Protocol

Destination Server

1719

UDP

H.323 Gatekeeper RAS port

Cisco Cube 

1720

TCP

H.323 Call Signaling


2855-2856

TCP

MSRP


3478

UDP

STUN service


3479

UDP

STUN service


5002

TCP

MLP protocol server


5003

UDP

Neighborhood service


5060

UDP & TCP

SIP UAS

Cisco Cube

5070

UDP & TCP

SIP UAS


5080

UDP & TCP

SIP UAS

Cisco Cube

8021

TCP

ESL


16384-32768

UDP

RTP/ RTCP multimedia streaming

Cisco Cube

5066

TCP

Websocket


7443

TCP

Websocket


8081-8082

TCP

Websocket



Integration on Website

Expertflow will provide the JS/HTML link to the website owner, which they need to add in the website header.  Expertflow will also provide the sample HTML  page that can be used as a reference for the website owner/developer. 


Setting UP a Video Call

For setting up a video conferencing system , We have a  Debian server running the latest stable FreeSWITCH build, IP phones( Zoiper/ Jabber) and Cisco SX  set . 

The default port5,060,506,150,805,081

  • codec,H.261 Video (passthru),mod_h26x
  • codec,H.263 Video (passthru),mod_h26x
  • codec,H.263+ Video (passthru),mod_h26x
  • codec,H.263++ Video (passthru),mod_h26x
  • codec,H.264 Video (passthru),mod_h26x


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