At Sheerbit, we specialize in delivering robust, flexible, and secure SIP (Session Initiation Protocol) development services tailored to your business communication needs. Whether you’re building a VoIP application, a cloud-based PBX, or a complete softswitch system, our expert SIP developers ensure high-quality signaling, seamless interoperability, and real-time communication at scale.
As the backbone of modern VoIP solutions, SIP plays a crucial role in initiating, maintaining, and terminating voice, video, and messaging sessions across IP networks. With years of experience in VoIP development, telecom software engineering, and open-source SIP platforms, Sheerbit is your trusted partner for end-to-end SIP development.

We build tailor-made SIP solutions including softswitches, call servers, SIP clients, and enterprise communication systems. From backend logic to front-end UI, everything is designed to fit your business model and user experience.
Our team develops web, desktop, and mobile SIP softphones with features like HD calling, call transfer, voicemail, contact sync, and chat integration. We use frameworks like SIP.js, JsSIP, and native SDKs for seamless cross-device functionality.
We integrate your VoIP infrastructure with reliable SIP trunk providers, enabling cost-effective, scalable voice connectivity. This includes secure trunk configuration, number provisioning, and traffic optimization.
From OpenSIPS and Kamailio to FreeSWITCH and Asterisk, we set up and manage your SIP server architecture, enabling advanced routing, load balancing, and high availability.
Need to connect legacy PSTN systems with VoIP? We build SIP gateways that handle protocol translation, media bridging, and seamless SIP-to-PSTN or SIP-to-analog integration.
Security and reliability are critical. We implement TLS, SRTP, and authentication protocols, and offer debugging, SIP trace analysis, and fraud prevention to ensure secure and uninterrupted service.
Our solutions strictly follow SIP RFC standards, ensuring seamless interoperability with SIP-enabled devices, carriers, and third-party platforms. Whether you're integrating with existing systems or deploying a new network, compatibility is guaranteed.
Enable HD voice and video communication over IP networks with low latency, jitter control, and adaptive jitter buffers. Our SIP systems ensure crisp, uninterrupted conversations even under fluctuating network conditions.
Support for dynamic call routing, least-cost routing (LCR), failover, hunt groups, and number translation is built into every solution, giving you total control over how calls are processed and directed.
Designed for growth, our SIP platforms support multi-tenant deployments with user-level permissions, dedicated trunks, isolated billing, and centralized management—ideal for service providers or enterprise use.
Protect your SIP infrastructure with TLS encryption, SRTP media security, authentication protocols, SIP firewall rules, and protection against toll fraud, spoofing, and SIP attacks.
Gain insights into call performance, system health, and SIP signaling with integrated real-time dashboards, CDR reporting, alerting systems, and diagnostic tools like SIP trace and packet capture.
Hire a dedicated ReactJS developer or expand your ReactJS development team as per your unique needs.
SIP (Session Initiation Protocol) is a signaling protocol used to initiate, manage, and terminate voice, video, and messaging sessions over IP networks. It enables devices and servers to establish real-time communications like VoIP calls.
We work with all major open-source SIP platforms including Asterisk, FreeSWITCH, OpenSIPS, and Kamailio, and can integrate with commercial SIP infrastructure if required.
Yes. We build highly scalable SIP infrastructures with support for load balancing, distributed SIP proxies, media servers, and performance tuning—ideal for large VoIP service providers and enterprises.
Yes. We provide 24/7 support, performance monitoring, SIP trace/debugging, log management, and system updates to keep your infrastructure running securely and efficiently.
Simply contact us through our website or schedule a discovery call. Our SIP experts will assess your requirements, recommend a solution, and provide a custom proposal tailored to your needs.
Hear how businesses like yours scaled faster with us.
Looking to develop next-gen communication solutions? Sheerbit specializes in custom VoIP development, WebRTC applications, Asterisk and FreeSWITCH integration, OpenSIPS/Kamailio solutions, SIP trunking, MVNO platforms, and telecom API development.
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"Sheerbit Technologies did a fine job in developing VoIP Softphone App, I will continue to do more work with the Sheerbit Technologies team."