OpenSER transformed into Open Session Initiation Protocol Server or OpenSIPS, as it is known, in 2008 and became foundational for VoIP development. It is open source, free and evolves to make it easier to develop scripts for a variety of VoIP apps and manage servers. It is the preferred choice for VoIP projects due to auto-scaling, command line interface and selectable memory allocator support among other features. Should you plan to develop VoIP projects then choosing OpenSIPs is a wise decision. The best way to go about it is to hire OpenSIPS developer with proven expertise.
Why openSIPS is to be preferred:
OpenSIPS has competition from Kamailio and Asterix, to name a couple but it does have its advantages such as a B2B module that makes it easy to implement topology hiding. If you want a proxy server then SIP is ideal whereas Asterisk is more of a media server. To its credit this platform is more robust and it can handle more traffic coupled with high speed, making it ideal for implementation in VoIP services and telecom. Using it in enterprises would bring carrier grade performance along with features like connecting a variety of SIP devices and trunks to work seamlessly within network architectures. If factors like NAT traversal, load balancing and least cost routing and failover are important, as they should be, then this platform wins hands down. You can hire OpenSIPS developers for OpenSIPS solution development to match your needs.
Need a proxy server? Look at SIP based on OpenSIPS:
One component of the VoIP network and infrastructure is the SIP proxy server. This is an essential part of the architecture in carriers, telecom and ITSPs. In such environments you need a SIP Proxy Server to handle user agent requests and forward them to the right destination as well as modify headers and handle least cost routing. OpenSIPS has a decided advantage in that it can handle high throughputs of millions of simultaneous calls besides taking care of routing and integration.
At the same time, OpenSIPS is flexible and feature rich to be used for a variety of other VoIP applications in this environment such as session border controllers and IP PBXs that serve as back to back user agents. You get better integration and ease of deployment when OpenSIPS is the foundation to a superb performing layer 7 protocol for your voice and media traffic. OpenSIPS becomes your defacto tool for routers, gateways, switches, application servers and call registrars, permitting seamless operations with high reliability factor. In today’s VoIP environment you do not have only call traffic; media and data also form part of the traffic and, in this regard, OpenSIPS integrates well with your IP multimedia subsystem network.
Hire openSIPS developers to customize proxies and fine-tune performance to suit your working model and performance goals. One thing leads to another and you can get a variety of VoIP apps built by the same developer with flawless performance, so necessary for your reputation and customer satisfaction.
A lot is possible with OpenSIPS:
While OpenSIPS makes for an excellent proxy server, it is just as capable and versatile as a building platform for telecom VoIP solutions such as class 4 softswitch, WebRTC solutions and least cost routing with failover server. Hire openSIPS developer to start on a basic project like a SIP server and you can take it from there to more application specific OpenSIPS solution developments such as the ones outlined above. You can just as well employ developers to troubleshoot existing installation, offer consultancy for upgrades, fix bugs or for technical support to third party software solutions.
Telecom/carriers will find openSIPS advantageous since its use for class 4 routing covers SIP aliases, CPL, dialplan, algorithm based dispatching, prefix based routing and Enum or Geolocation routing. For class 5 implementations you have the benefit of B2B, call queuing, authentication and white/black list management. In IP PBX solutions you can integrate it smoothly with other platforms like Freeswitch or Asterisk to provide variety of features like busy lamp field, SMS gateway and resource list server.
For telecom operators and ITSP services it is essential to look at the broader picture when they set up infrastructure or start operations or think of upgrading existing ones. OpenSIPS is a nice choice providing convergence of variety of VoIP applications. It has a positive bearing on cost and performance too
Hiring the right people:
Anyone who has a background in IT with specialization in open source communication platforms like Asterisk and Freeswitch or OpenSIPS can claim to be an expert but real life is different. It needs experience to build up expertise that results in flawless in demanding telecom environments and this is where you separate men from boys. If you wish to hire openSIPS developers look at the work they have done in OpenSIPS solution development such as carrier grade softswitches and session border controller.