Skip to main content

Managed VoIP

When you need VoIP expertise for the current servers management or to design a new deployment, you can find support here. Our expertise covers the following software

  • Kamailio
  • RTPengine
  • FreeSwitch

We deploy servers in the AWS cloud, Kubernetes, and on bare metal.

WebRTC gateway

 

If you develop a browser application to make and receive calls, you need to relay SIP messages from the browser web-socket connections to SIP servers. Kamailio is designed exactly for this use case and allows you to achieve this goal.

Session Border Controller with High Availability

 

Kamailio and RTPengine can be configured to proxy communications between public and private network segments. As an example. Deployed in the DMZ to separate the Public internet and private PBX.

Another use case is IPv6/IPv4 protocol conversion. When you need to develop a dual-stack application for the App Store and Google Play. SBC accepts IPv6 and IPv4 client connections from the Internet and delivers calls to your IPv4-only servers.

A cluster of two pacemaker nodes can be used for deployment. In this case, you will achieve High Availability with "active-backup" failover for call signaling and RTP media.

VoIP server deployment to Kubernetes

 

At present, Kubernetes is a mainstream technology for workload management, but this introduces complexity for SIP call delivery. This happens because Kubernetes actively uses NAT technology.
For this use case, Kamailio and RTPengine software can be deployed to a Kubernetes cluster to proxy SIP call signaling and RTP media.

Contacts

 

Company address:
8200, Bulgaria, region Burgas, city Pomorie, st. Veleka 7, ap. 30

Company EIK number: 205090167
VAT number: BG205090167
Commercial register link.

Phone: +359 87 721 3539
Email: Rudenko.Zhannetta@gmail.com