Multimedia Communication and Collaboration Solutions
Multimedia Communication and Collaboration Solutions
At our International R&D Center, we design and develop New Generation Communication Systems including Audio and Video Communication, Call Control Applications and WebRTC technology.
We benefit from our Process, Software and Test capabilities to develop fast and high quality products. Depending on the requirements of the product and market, we use "Waterfall", "Agile" or KANBAN methods in Software Development Life Cycle and implement Continuous Integration Practices.
Our Product Design and Development capabilities are as follows:
- New Generation VoIP Network Concepts, Components (various types of Media Gateways, Media Gatekeepers, Media Servers, etc.), Services, Architecture and Protocols (WebRTC, SIP, H.248, RTP, TCP/IP, 3GPP IMS Protocols, UDP, SCTP etc.).
- Respective services and messaging protocols in order to provide interoperability with conventional systems (ISUP, TUP, PRI, QSIG etc.)
- Structure, basic functions and units of conventional TDM-based telephone switchboards
- SIP protocol and SIP servers
- Various international telecommunication standards: IETF RFCs including mainly RFC 3261, SIPconnect 1.1, ITU, 3GPP and ETSI standards
In today’s highly competitive environment, it is crucial that the products and solutions we develop meet the customer needs and expectations and feedback is provided to development groups immediately to detect and solve any potential problems. This applies to both product quality and rapid release of new products and features. We use end-to-end Software Development Process Automation with the Continuous Integration Infrastructure and effective test automation at our Testing and Verification Center and offer a significant competitive advantage in Product Development process.
Unified Communications Products
The key for today's communication solutions is user experience. Our International R&D offers easy integration between the products we work on and standard end-user devices. We enhance user experiences by improving terminal device options with HTML5, REST APIs device container and Web RTC technologies; thus serving millions of home and business users on these platforms.
SIP Application Server
IP based multimedia communication applications with multiple features for business and home users provides communication through devices such as Audio, Video, Instant Message, Presence, Mobility, Conference, Mobile, Tablet and PC Web Browser. The SIP Application server we developed is an essential product for SaaS (Software as a Service), Hosted Integrated Service, SIP Trunk and IMS TAS services.
WebRTC Gateway is a bridge between traditional VoIP and the free Internet ecosystem. It allows conventional audio service providers to offer service to competitive applications on the Internet.
We provide Web Centric APIs so that application developers and web service providers can develop applications on various communication services including, audio, video, presence, contacts sharing, instant messaging and communication. The customers can also design their own terminal devices and customize them according to user experience.
VIO-Video Communication Media
International R&D starts releasing exciting new products and ideas to the market that benefit from its technological competence and vision. This enables differentiating products supported with the state-of-the-art technology to meet with customers.
We offer corporations innovative audio and video instant conference applications media; namely ViO Academy for education and ViO Medica for healthcare both using the ViO product infrastructure over cloud.Click here for more information.
Carrier-Class Voice Products
Traditional circuit-switched telecommunication systems (PSTN, Public Switch Telephone Network) are transitioning to IP based VoIP systems of high sustainability, scalability and flexibility as a result of increasing use, service quality, easy management and new service needs.
Our R&D team performed research, product and service design, development and integration on telecommunication networks for many years and worked on the sustainability of such structures while providing support to many customers on the international market. Now, the team uses this know-how and experience in new generation VoIP systems to develop scalable and secure solutions with high quality, capacity and rich service content, therefore responding to requirements of different segments.
In addition to the fully IP based VoIP solutions that we offer, hybrid solutions compatible with traditional TDM systems allows for gradual, smooth and easy transition from conventional systems to VoIP structures.
Our R&D team designs and develops software and continues product software support in accordance with various service demands of our customers in today's telecommunication world where different standards (ANSI, ETSI and ITU) are used on different continents.
We provide software development and product support services for large customers including BT, Avaya, Genband, Verizon, AT&T, BellCanada, Vodafone and Sasktel.
VoIP Call Server:
Telecommunication systems have been transformed with the development of the Internet, broadband and mobile technologies; thus, operators and companies are inclined towards telecommunication solutions that bring together user experience and the telecommunication network providing such experience. This convergence requires smart and advanced call and session control servers that offer a seamless experience for users by combining various networks, media, protocols, architectures and devices.
Call servers are critical network elements for creating a smooth user experience by switching voice calls and multimedia sessions. They are among the main elements of advanced architectures such as IMS.
The call server being developed by our R&D team is a communication platform converging IP and IMS supporting voice and multimedia. The server is able to run on commercial servers and offers different features for users and applications for different segments. The features include call and session control, network connectivity, switching, signaling and protocol interaction, core and edge interaction.
Our call server includes both IP based users and users on conventional circuit-switched systems within the same structure, thus offering a lower cost and easy to manage solution.
It is among the leader solutions in the international market and supports more than 700 services, together with more than 60 trunk signal types and more than 30 line types. It uses protocols such as H248, MGCP, SDP, H323, SIP in VoIP systems, along with protocols such as Number 7 (ISUP and TUP variants), CAS and DPNSS.
Media Gateways are used to deliver services to the user in telecommunication networks. The services purchased by the end-users can be offered both over telephone or over IP based systems.
We develop software and offer product software support for Gateways with different size and structures. They can be used on IP and TDM networks, undertake ATM and IP backbone task for both end-users and for many products through custom interfaces.
Some of our software developed using C, C++ and Java languages include the following;
STM-1 Optic transmission interface development
GigE (Gigabit Ethernet) interface development
TL1 real-time test system development
Geographical backup system
Alignment with wireless network (2G, 3G ve 4G/LTE compatibility)
Call statistics collection system.