#opensips

[gventuretech]

OpenSIP development plays a transformative role in shaping the future of VoIP technology. Its scalability, flexibility, and interoperability make it an indispensable tool for building a wide range of communication solutions, from enterprise PBX systems to WebRTC applications and IoT integrations. As the demand for seamless and reliable communication continues to grow, OpenSIP development will remain at the forefront of innovation, driving the evolution of VoIP technology for years to come.

[inextrixtechnologies]

OpenSIPS 3.4.x is the latest stable release of this open-source communication engine, offering a powerful and flexible platform for building next-generation telephony services. This LTS (long-term support) version prioritizes consolidation of past improvements, including performance enhancements and comprehensive testing, while still delivering new functionalities like script improvements and module updates. OpenSIPS 3.4.x empowers developers and businesses to build reliable, efficient, and scalable communication solutions.

[vindaloo-softtech]

Hire VoIP Developers for your Business - Vindaloo Softtech

The recent pandemic forces businesses to adopt the remote working model and allows them to tap into a more comprehensive resource pool for project execution. Observing the demand for dedicated VoIP developers for building custom enterprise-grade VoIP solutions for businesses, Vindaloo Softtech announced the launch of “Hire VoIP developer services”. Vindaloo Softtech is a VoIP development company known for its innovative solutions and services and houses several capable and expert VoIP developers. The company has been offering expert VoIP development services to its customers for many years, and now it has decided to meet the need for dedicated VoIP developers.

“Our team of expert VoIP developers has provided VoIP development services to our customers for 6+ years. Our customers are delighted with our VoIP development services. However, sometimes we receive requests from customers that they would like to hire VoIP programmers for their VoIP development projects. So we launched the “Hire VoIP developer” service as the demand rose. Now anyone can hire an experienced VoIP developer for their VoIP Development projects on flexible pricing plans”, shared a spokesperson of Vindaloo Softtech.

According to the details about the model, customers can hire VoIP developers with in-depth knowledge and experience in all popular VoIP technologies. This hiring model is perfect for VoIP development projects that use different VoIP technologies to develop the final product. On the other hand, if a business needs an expert in a specific VoIP technology, it can hire an expert in the respective technology. For example, if the whole VoIP development project is on Asterisk or FreeSWITCH, the business can hire a dedicated Asterisk developer or a FreeSWITCH developer from Vindaloo Softtech. Furthermore, the model is flexible enough to allow customers to hire more than one VoIP developer for as long as they want. The benefits of hiring remote VoIP developers surely outweigh the costs.

Currently, the company has the resource pool for its customers to hire a dedicated developer in any of the following disciplines:

Asterisk Developers

FreeSWITCH Developers

Kamailio or Opensips Developers

Talking about the hiring model, the company has three different augmented team hiring models to let its customers hire developers.

1. Full time

The company lets its customers hire VoIP experts for 160 hours per month. The hired developer works exclusively on client projects full-time.

2. Part-time

In this model, the customers of Vindaloo Softtech can hire VoIP software developers for 80 hours per month. It means the hired VoIP developer will work for 4 hours per day on the VoIP development project of that customer.

3. Hourly

In this model, Vindaloo Softtech lets its customers hire VoIP developers for a predefined number of hours. The customers of this VoIP development company can hire one or more developers for a specific number of hours which can be used within a week or a month based on that customer’s requirement. This model is focused on VoIP service providers who may need expert-grade VoIP experts to achieve short-term goals or unforeseen needs.

The company’s spokesperson has shared that the hired VoIP developer will perform all required operations for the client, such as development, customization, bug fixing, support, etc. The company has launched a web page under the ‘Hire Dedicated Resources’ section to answer all the queries that may raise the customer’s mind. The webpage contains all the required information related to the company’s services. Get in touch with us!

[bytebranllp]

Selecting the Ideal Tech Stack for Your VoIP Project

Software development involves various technologies and processes that are constantly evolving with new tools and frameworks. While these advancements simplify development, they also add complexity when selecting the right technologies for a project. Beyond the technical aspects, successful software development requires strong teamwork, project management and communication among diverse teams. These skills are crucial for keeping projects on track and within budget. Our blog article aims to give you a clear understanding of the crucial factors to consider when choosing the right technology stack for your software development project.

Understanding the tech stack

A tech stack is a set of tools and technologies used to build and run an application to cover everything from servers and databases to frontend and backend frameworks. Here’s a quick look at its main components:

Frontend: Handles the user interface and experience with technologies like HTML, CSS, JavaScript, and frameworks like React, Angular and Vue.js.

Backend: Manages business logic, database interactions, and server setup using languages like Python or Ruby and backend frameworks.

Databases: Store and manage application data, including relational databases like MySQL and NoSQL databases like MongoDB.

DevOps and Cloud Services: Ensure your application is integrated, scalable, and well-maintained.

VoIP Platforms

FreeSWITCH, Kamailio, OpenSIPS, and Asterisk are top platforms for creating VoIP services.

FreeSWITCH is an open-source platform that connects and routes communication protocols.

Kamailio and OpenSIPS are high-performance SIP servers for managing voice, video, and real-time communication.

Asterisk is a free framework that turns a standard computer into a powerful communication server.

Tech Stacks in Practice

Have you ever wondered what technology stacks power your favorite brands? Let's explore the tech stacks some industry leaders use and see what they rely on to deliver their services.

Uber Tech Stack:

Web Servers: NGINX, Apache

Databases: MySQL, PostgreSQL, MongoDB

Server-side Framework: Node.js

Programming Languages: Python, Java, JavaScript, Objective-C

Uber's tech stack combines robust web servers with versatile programming languages and frameworks, enabling smooth communication and data management across its platform.

Instacart Tech Stack:

Server: NGINX

Databases: PostgreSQL, Redis

Server-side Framework: Rails

Programming Languages: Ruby, Python, Objective-C

Instacart uses a streamlined stack focused on efficiency and reliability. It strongly emphasizes fast database operations and a robust server-side framework.

Reddit Tech Stack:

Server: NGINX

Databases: PostgreSQL, Redis

Server-side Framework: Node.js

Programming Languages: JavaScript, Python

Reddit's stack is optimized for handling large amounts of user-generated content. It uses powerful databases and a modern server-side framework to maintain its performance and scalability.

These examples highlight brands' different choices in building their tech stacks tailored to their specific needs and operational demands.

Choosing the Right Tech Stack for VOIP App Development

Selecting the right tech stack for app development can be overwhelming, especially for non-technical entrepreneurs. While developers often have their preferred stacks, it's wise to seek guidance from a subject-matter expert. An expert can provide valuable insights and help you avoid future issues. If you're short on resources, start with these simple steps:

Understand Your Users' Needs: Focus on what your users need. For example, if most of your users are on mobile devices, consider a "mobile-first" tech stack. If your app is "mobile-only," your tech stack will likely differ from other types of applications.

Define Your Development Needs: The complexity and scale of your app will influence your tech stack choice:

Small Projects: Well-defined stacks like Python-Django or Node.js-React are ideal for simple apps like MVPs or one-page apps.

Medium Projects: E-commerce stores or mid-market apps require more complex stacks with multiple layers of programming languages and frameworks.

Large Projects: Complex marketplaces or social apps need a robust tech stack designed to handle high-volume use and maintain performance.

Evaluate Your Resources: When choosing a tech stack, consider your available resources. Even the best tech stack won't be effective without the right developers to implement it. Many frameworks and tools are open-source and free to use, but remember to account for costs related to servers and hosting when making your decision.

Essential Components for a VoIP Software Technology Stack

Let’s understand these components.

Real-Time Communication Protocols: SIP (Session Initiation Protocol) and RTP (Real-Time Transport Protocol) are essential for managing voice, video, and messaging sessions. WebRTC enables real-time communication directly in browsers and mobile apps, making development easier.

Codecs: Codecs like G.711, G.722, G.729, and Opus compress and decompress voice data, balancing call quality and bandwidth usage.

Programming Languages: Python and Java are commonly used for their simplicity, readability, and strong support for multithreading and concurrency.

Web and Mobile Frameworks: Angular and Node.js are popular for web development, while Flutter and React Native are favored for cross-platform mobile apps.

Databases: SQL databases like MySQL and PostgreSQL handle structured data, while NoSQL databases like MongoDB offer scalability for unstructured data.

Server Technologies: Cloud services like AWS, Google Cloud, and Microsoft Azure provide scalable, secure environments with various development tools.

Testing Tools: Selenium for UI testing and Apache JMeter for load testing help ensure the software performs well and offers a good user experience.

Take Away

A tech stack is essential for a startup or business's success, so selecting the right technologies, languages, frameworks, and tools is essential. 

Choosing the right technology stack for VoIP software development depends on your project's specific needs and goals. To make the best choice, follow these three rules: prototype and test, seek expert advice both within and outside your team and stay flexible. Doing so will help you build a strong and scalable product. Consulting experienced developers or software development companies can help you get tailored recommendations. With the right choice, your VoIP application will meet your current needs and be adaptable to future changes.

[2600hz]

Under the hood: OpenSIPs and FreeSWITCH

At 2600hz we like to say that we've "re-written the telephony stack." Which either elicits looks of belief and wonder or hardcore skepticism. Which is fair. So we're going to be temporarily replacing our usual Wednesday Bug series with a short series on the architecture stack we've designed to improve telecom like it's never been done before. The seven components are: 

Session Border Controllers (for SIP)

Load Balancers (for SIP)

Media Servers

SMS / MMS / SMPP

Dialplan /  Call Processing

Messaging Buses

Databases

Today, we'll talk about two of those components and our experiences using them. These components are OpenSIPs--our load balancer, and FreeSWITCH--our media server. 

Why We Selected These Components OpenSIPs has a long history of being an extremely stable, flexible SIP proxy, allowing you to set up a load-balanced, redundant setup for many millions of calls per day. One of its greatest strengths is its rock-solid reliable performance and its ability to do reasonably complex tasks in a short amount of code. But OpenSIPs can't natively handle audio and frankly, wasn't really designed to.

FreeSWITCH has a long history of being an extremely stable and flexible media switch,  allowing you to handle many thousands of simultaneous phone calls on a minimal amount of hardware without degrading call quality. In addition, FreeSWITCH has built-in functionality to support integration with OpenSIPs which is crucial as it would otherwise be hard to work with.

How OpenSIPs and FreeSWITCH Are Different At the fundamental level, OpenSIPs and FreeSWITCH serve two different purposes. OpenSIPs was originally designed to be a scripting engine and a proxy for SIP packets. In English, think of it as the "traffic cop" with the neon vest that stands in a busy intersection. You can program it to tell it how to direct traffic in different ways, based on volume, quality or location or other factors.

So if we say OpenSIPs is the load balancer that directs traffic (call flow) along the road which is the Internet...where does all that traffic get directed towards? You guessed it. FreeSWITCH. 

FreeSWITCH is a media server. Think of a media server as being similar to a language translator between two people who speak two completely different languages. In its simplest form, a media server takes audio from one person, processes it, and passes it on to another person. It also provides translation services, meaning if one person "talks" another language it can convert that language on the fly to something the other party can understand.

Technically speaking, there are major differences in how you would architect a traffic cop versus a translator. The analogies to real life work well, so lets continue with those.

In the OpenSIPs as "traffic cop" analogy, think for a minute about the length of time a traffic cop is concerned about the traffic flowing around it. A traffic cop directs cars down a street, but once the cars pass by, the cop no longer cares who they were, where those cars go,  or what direction they continue to travel in. In this regard, a traffic cop may see millions of cars pass them per day but is only ever concerned with the few cars it is currently directing at any given time.

In the FreeSWITCH as "translator" analogy, lets contrast the length of time a translator must focus on a conversation between two people. (For the purposes of this analogy, we will represent a FreeSWITCH box as a building filled with thousands of translators.) A translator must focus on only one conversation at a time while being able to understand and speak the language of both sides. The translator can't leave in the middle of a conversation easily to be replaced with another translator – they must stay for the duration of the entire conversation.

Comparing OpenSIPs and FreeSWITCH For Capacity (and why you shouldn't) I've seen lots of people try to compare the capacity of OpenSIPs to that of FreeSWITCH. This doesn't make much sense. Looking back at the above analogies we can see the fundamental differences between OpenSIPs and FreeSWITCH.  In technical terms, this translates into different ways to analyze capacity based on work done by threads.

Both OpenSIPs and FreeSWITCH utilize threads in order to process calls. The difference lies in what the threads are doing.

For OpenSIPs you have a Linux thread running which is the equivalent to a traffic cop. A single thread might pass 10,000 calls per hour from point A to point B, processing a light-weight instruction script ("is this person authorized? OK, let them through!") for each call. The real concern around capacity for OpenSIPs is how many simultaneous requests OpenSIPs will get at any one point in time. Think of this as the difference between a four-way intersection and an eight-way intersection. You might need only one traffic cop at a four-way intersection but need an extra cop at an eight-way intersection simply because of the extra load.

For FreeSWITCH you have a Linux thread running which is the equivalent of one translator. A single thread will only ever process one call, for the entire duration of that call.

So, when planning capacity, you need to ask yourself: OpenSIPs – How many simultaneous people are going to register a phone, make a call or hangup on a call at the same exact time? Divide that number by how many calls each thread can process and you know how many threads you need. FreeSWITCH – How many simultaneous people are going to be on the phone at any given time? That is how many threads you need.

Now that we understand the difference between OpenSIP and FreeSWITCH and their purposes and capacities, stay tuned for next Wednesday's post on FreeSWITCH support for OpenSIPs (or any proxy.) 

[ob4by]

阿里云 opensips nat内网穿透

书接上文，上次在阿里云安装好opensips之后，发现无法在公网ip监听。为了能够快速测试后续改用内网服务器搭建sips服务器。现在功能差不多了，于是就需要将opensips服务器重新部署到公网上。不得不再次面对这个蛋疼的问题。通过搜索之后发现可以通过rtpproxy实现内网穿透？（这个说不知道准不准去，没有深入研究实现原理） 方法也比较简单，安装rtpproxy： sudo apt-get install rtpproxy 安装之后修改opensips.cfg配置文件，添加如下内容： #rtpproxy loadmodule "rtpproxy.so" modparam("rtpproxy", "rtpproxy_sock", "unix:/var/run/rtpproxy/rtpproxy.sock")…

[sheerbittech]

Sheerbit: The Top VoIP Development Company for Custom, Scalable Solutions

Introduction

In today’s fast-paced digital landscape, clear and reliable communication is no longer a luxury—it’s a business imperative. Voice over Internet Protocol (VoIP) technology has revolutionized how organizations connect, collaborate, and serve their customers. However, not all VoIP development companies are created equal. Selecting the right partner can mean the difference between a smooth deployment and ongoing technical headaches. This is where Sheerbit shines. As a leading VoIP development company, Sheerbit combines deep technical expertise, bespoke solutions, and unwavering customer support to deliver communication platforms that scale with your business.

Understanding VoIP and Its Business Impact

VoIP enables voice calls, video conferences, and multimedia data to traverse IP networks rather than traditional telephone lines. This shift reduces costs, boosts flexibility, and integrates seamlessly with cloud-based and on-premise systems. Organizations that adopt VoIP enjoy features such as advanced call routing, click-to-dial, call analytics, and integration with CRM or helpdesk platforms—empowering teams to work smarter and respond faster to customer needs.

Common Challenges in VoIP Deployments

Even with compelling benefits, VoIP projects can falter if not handled by seasoned professionals. Organizations often face:

Quality of Service (QoS) issues that lead to dropped calls or latency

Security vulnerabilities exposing voice traffic to eavesdropping or fraud

Complex integrations with legacy PBX systems or third-party applications

Scalability hurdles when call volume spikes or new offices come online

Ongoing maintenance and lackluster support after go-live

Addressing these challenges demands a partner who understands both the networking fundamentals and the unique needs of your business.

Why Sheerbit Stands Out

Sheerbit has built its reputation as the best VoIP development company by focusing on three core pillars: technical excellence, client-centric customization, and comprehensive support.

1. Technical Excellence

Every Sheerbit engineer brings extensive experience with leading VoIP platforms—Asterisk, FreeSWITCH, OpenSIPS, Kamailio, and WebRTC frameworks. Whether you need a robust SIP trunking solution or a cutting-edge WebRTC application, Sheerbit’s team writes clean, scalable code and adheres to industry best practices for network performance and reliability.

2. Custom VoIP Solutions

Off-the-shelf VoIP packages rarely fit every business scenario. Sheerbit specializes in tailor-made development services, from crafting custom dial plans and interactive voice response (IVR) systems to integrating advanced call-center features like predictive routing and real-time analytics. With Sheerbit, you can hire VoIP developers dedicated to understanding your workflows and delivering solutions that align perfectly with your objectives.

3. End-to-End Support

The deployment of a VoIP system is just the beginning. Sheerbit offers full-lifecycle services: consulting and needs assessment, architecture design, development, testing, deployment, and post-launch maintenance. Their DevOps-driven processes ensure seamless updates, continuous monitoring, and rapid resolution of any issues—minimizing downtime and safeguarding call quality.

Key Service Offerings

VoIP Development Services: Sheerbit engineers build feature-rich VoIP applications, including softphones, mobile VoIP apps, and web-based conferencing tools. They ensure interoperability across devices and browsers, delivering user experiences that mirror or exceed traditional phone systems.

Custom Integrations: Leverage your existing investments by integrating VoIP with CRMs like Salesforce or HubSpot, helpdesk platforms such as Zendesk, or bespoke databases. Sheerbit’s APIs and middleware ensure call data syncs accurately with your business systems.

SIP Trunking & PBX Migration: Whether you’re migrating from a legacy PBX to a modern SIP-based infrastructure or establishing new SIP trunks for international call routing, Sheerbit’s proven migration framework guarantees minimal service interruption.

Security & Compliance: Voice services must be secure. Sheerbit implements TLS/SRTP encryption, robust firewall configurations, and fraud-detection modules. They also assist with regulatory compliance (e.g., GDPR, HIPAA) to protect sensitive communications.

Success Stories

Global Retail Chain Enhances Customer Support A multinational retailer struggling with call center overload engaged Sheerbit to deploy a scalable Asterisk-based IVR with predictive call routing. Post-launch, average wait times dropped by 40%, and customer satisfaction scores rose significantly.

Healthcare Provider Integrates VoIP with EHR Sheerbit developed a HIPAA-compliant FreeSWITCH solution for a healthcare network, integrating audible call prompts directly into the electronic health record system. Clinicians saved an average of 10 minutes per patient, boosting operational efficiency.

How to Hire Sheerbit’s VoIP Developers

Engaging with Sheerbit is straightforward. After an initial consultation to assess your needs, you’ll receive a detailed proposal outlining scope, timelines, and pricing. You can choose to hire VoIP developers on a project basis or onboard them as part of your extended team. Flexible engagement models include fixed-price projects, time-and-materials contracts, or dedicated-team arrangements.

Pricing & Engagement Models

Sheerbit offers transparent, competitive pricing tailored to project complexity and resource requirements. Typical engagement tiers include:

Standard Package: Core VoIP deployment with essential features

Advanced Package: Custom development, integrations, and analytics

Enterprise Package: Full-scale solutions with ongoing support and SLAs

The Implementation Process

Discovery & Planning: Define objectives, technical requirements, and success metrics.

Design & Architecture: Create network diagrams, call-flow maps, and infrastructure plans.

Development & Testing: Build features in agile sprints, perform comprehensive QA, and conduct pilot testing.

Deployment & Training: Roll out the solution, configure networks, and train your IT staff and end users.

Support & Optimization: Provide 24/7 monitoring, periodic performance reviews, and iterative enhancements.

Conclusion & Call to Action

Selecting the best VoIP development company can transform your organization’s communications, delivering cost savings, operational agility, and superior customer experiences. With Sheerbit’s proven expertise in custom VoIP solutions, end-to-end support, and dedication to quality, your business is poised for seamless, future-ready communications.

Ready to elevate your voice infrastructure? Contact Sheerbit today to schedule a free consultation and discover how you can harness the power of a tailored VoIP solution built by industry experts.

[gventuretech]

OpenSIPS Development: Elevate Your Communication Infrastructure involves leveraging OpenSIPS, an open-source Session Initiation Protocol (SIP) server, to enhance communication infrastructure. OpenSIPS is utilized to optimize Voice over Internet Protocol (VoIP) networks, install and configure load balancing for VoIP systems, and tailor solutions to boost communication infrastructure.

[vindaloo-softtech]

Transform Your Business with Vindaloo Softtech's Innovative IT Solutions

Experience top-tier IT services with Vindaloo Softtech. Specializing in VoIP Software Development, Web App Development, Custom CRM Development, and Staff Augmentation, we provide innovative, cost-effective solutions tailored to your business needs. Our products—PepperPBX, PimentoPhone, CloveKonnect, CallCentr8, Paprikall —ensure seamless communication and operational efficiency. We leverage advanced technologies like FreeSWITCH, Asterisk, Kamailio, OpenSIPs, WebRTC, ReactJS, and PHP. As members of GESIA, ESC, and ContactCenterWorld, we uphold industry standards and stay ahead of market trends. For more information please visit: https://www.vindaloosofttech.com/

Contact Us: Email: info@vindaloosofttech.com Call us: +1 702 200 8967

[isabellaswan0]

Get custom software, application and module development in OpenSIPS http://bit.ly/1KVhmnr

[gventuretech]

With OpenSIP doesn't have to be daunting, even for beginners. By understanding the fundamentals of SIP and VoIP, setting up OpenSIP, and leveraging its features and functionalities, you can create scalable and feature-rich communication applications tailored to your specific needs. Whether you're building a messaging app, a voice calling platform, or a video conferencing solution, OpenSIP provides the tools and resources you need to bring your ideas to life. So why wait? Start building your next communication app with OpenSIP today!

[ob4by]

ubuntu 18.04 安装opensips 3.1

ubuntu 18.04 安装opensips 3.1

OpenSIPS is an Open Source SIP proxy/server for voice, video, IM, presence and any other SIP extensions.OpenSIPS is a multi-functional, multi-purpose signaling SIP server used by carriers, telecoms or ITSPs for solutions like Class4/5 Residential Platforms, Trunking / Wholesale, Enterprise / Virtual PBX Solutions, Session Border Controllers, Application Servers, Front-End Load Balancers, IMS…