Alternatives to OpenSIPS

3CX is an open-standards, software-based IP PBX that provides complete Unified Communications right out of the box. 3CX is suitable for all industries and sizes. It can provide everything you need, from mobility and status to advanced call center features and more, at a fraction the cost. 3CX is ideal for remote companies and includes mobile apps, video conference, and website live chat. The installation, management, and maintenance of the PBX are so simple that you can easily manage it yourself, on your own appliance, server, or on a cloud-based platform. You can also choose to have it hosted on 3CX for an easy solution.

Why struggle with large providers like Twilio, where response times to support requests can feel endless? At Signalmash, we’re more than just another CPaaS provider. Our commitment is to deliver top-tier CPaaS support. With us, your developers gain access to shared Slack channels with our team for quick assistance, and you’re always welcome to reach out directly to our CEO. Our real-time, high-quality support means faster development and empowers you to offer outstanding service to your customers. Our SMS services include: - SMS API - SMS CPaaS - SMS UCaaS - SMS no-code sending platform - 10DLC campaign support - Short code SMS - Toll-free SMS Voice services available: - Contact center telecoms - Voice termination - Voice origination - Local numbers - Short code numbers - Toll-free numbers - SIP Trunking Our No-code telecoms solutions: - UCaaS for SMS - CCaaS for call management - AI-driven solutions Signalmash – get unmatched support every step of the way. Schedule a call with us today for expert guidance!

Kamailio® is an open-source SIP server, which evolved from the earlier OpenSER and SER, and is distributed under GPLv2+, capable of managing thousands of call setups each second. This robust platform can be utilized for constructing extensive systems for VoIP, real-time communication, presence, WebRTC, instant messaging, and various other applications. Additionally, Kamailio is well-suited for scaling SIP-to-PSTN gateways, PBX systems, or media servers such as Asterisk™, FreeSWITCH™, or SEMS with ease. Its extensive feature set includes support for asynchronous TCP, UDP, and SCTP, secure communication through TLS for voice, video, and text, as well as WebSocket compatibility for WebRTC. The server supports both IPv4 and IPv6, offers simple instant messaging and presence capabilities with an integrated XCAP server and MSRP relay, and enables asynchronous operations. Furthermore, it includes IMS extensions for VoLTE, ENUM functionality, DID and least cost routing, load balancing, and routing failover. Kamailio also provides comprehensive accounting, authentication, and authorization features, while supporting a variety of backend systems, including MySQL, Postgres, Oracle, Radius, LDAP, Redis, Cassandra, MongoDB, and Memcached. All these capabilities make Kamailio a versatile choice for modern communication infrastructures.

PBXware is the first and most established IP PBX Professional Open Standards Turnkey Telephony Platform. PBXware has been providing flexible, reliable and scalable Next Generation Communication Systems (NGCS) and VoIP solutions to small and medium-sized businesses (SMBs), enterprises and Internet Telephony Service Providers, (ITSPs), call centers and governments around the world since 2004. This is done by combining the best of the latest technologies. Bicom Systems softswitch is available in the Business, Call Center, and Multi-Tenant Editions. Each edition supports specific features that maximize performance, reliability, expandability, and expandability.

FusionPBX serves as a versatile solution for both single and domain-based multi-tenant PBX setups, functioning as a carrier-grade switch, call center system, fax server, VoIP server, voicemail server, conference server, and voice application server, among other roles. Meanwhile, FreeSWITCH™ stands out as a highly adaptable and scalable communication platform that operates across multiple threads and platforms. It boasts an impressive array of features, including unlimited extensions, voicemail-to-email, music on hold, call parking, call center capabilities, call queues, and phone provisioning, catering to businesses of all sizes. By delivering corporate-level phone system functionalities, it meets the diverse needs of small, medium, and large enterprises alike. Significant efforts have been invested in enhancing security, resolving issues, and refining the codebase. Furthermore, ongoing advancements are being made in preparation for an upcoming release, with notable progress achieved; the master branch has outpaced the current version in several respects. As the team works meticulously to finalize the essential updates, anticipation builds for the enhancements that will accompany this release. The commitment to continuous improvement ensures that the platform will remain competitive and relevant in the rapidly evolving telecommunication landscape.

Thirdlane is built for service providers and MSPs who need a multi-tenant, white-label UCaaS platform with deep API control. It delivers voice, video, chat, and contact center features while giving providers the ability to integrate billing, CRM, and IT workflows with ease. Why providers choose Thirdlane: - True multi-tenant PBX architecture built for scaling to thousands of tenants. - White-label flexibility - deliver a branded experience to your customers. - Thirdlane Connect apps unify messaging, calling, and meetings across all devices. - Built-in Contact Center for sales and support teams. - CRM screen pops and Click-to-Call to boost productivity. - Extensive APIs for automation and third-party integrations. Business outcomes: - Launch branded UCaaS without the heavy R&D cost. - Increase recurring revenue with scalable multi-tenant architecture. - Offer enterprise-grade features at a competitive price point. Thirdlane empowers providers to own their UCaaS offerings end-to-end, increasing margins and ensuring long-term customer loyalty.

Everything essential for engaging with the FreeSWITCH platform is provided here. FreeSWITCH serves as a Software Defined Telecom Stack, facilitating the shift from traditional telecom switches to a flexible software solution that can operate on any standard hardware, whether it’s a Raspberry PI or a powerful multi-core server, thus unleashing the telecommunications capabilities of various devices. When paired with our cloud platform, SignalWire, FreeSWITCH can connect seamlessly to external networks and adapt to any scale of operation. To simplify your initiation, a pre-configured virtual machine come complete with comprehensive written and video tutorials, allowing you to get started quickly—just install VirtualBox and you will be set up in no time. For those with advanced programming skills, there are detailed instructions available to compile FreeSWITCH from source or explore other manual installation methods. Additionally, if you have a Debian instance available, you can find thorough guidance on how to install our packages from the provided link, ensuring versatility in your approach to utilizing FreeSWITCH. This platform truly empowers both beginners and seasoned developers alike, making telecommunications more accessible than ever.

Asterisk is a versatile open-source framework designed for creating various communication applications. It transforms a standard computer into a robust communications server. Asterisk serves as the backbone for IP PBX systems, VoIP gateways, conference servers, and numerous other tailored solutions. Its users range from small enterprises to large corporations, call centers, telecom carriers, and government bodies across the globe. Available at no cost, Asterisk thrives under the sponsorship of Sangoma. Currently, over a million Asterisk-based communication systems operate in more than 170 nations. Remarkably, nearly all companies listed in the Fortune 1000 utilize Asterisk in some capacity. Most frequently implemented by system integrators and developers, Asterisk can serve as the foundation for a comprehensive business phone system or can be employed to augment an existing setup or connect disparate systems. Whether you're interested in crafting a personalized system with Asterisk or purchasing an efficient, affordable turnkey solution, explore the options available to determine the best fit for your needs. With its flexibility and wide-ranging applications, Asterisk continues to evolve, ensuring it meets the demands of modern communication.

The VaxVoIP SIP Server SDK is a powerful development toolkit that enables building secure, scalable SIP-based VoIP solutions including IP-PBX, IVR systems, auto-dialers, smart agents, call centers, and calling card systems. Compatible with the SIP protocol, it supports various SIP devices such as softphones, hardphones, and ATA adapters, providing flexibility in hardware integration. The SDK offers a rich set of PBX functionalities like call queues, call forwarding, call recording, voicemail, and sophisticated call routing to create feature-rich telephony applications. Security is a priority, with protections against SIP floods, scanning attacks, and brute-force intrusions to safeguard communications. Developers can access audio PCM data in real time, facilitating the incorporation of AI, machine learning, and smart agents to deliver intelligent, automated interaction. This integration enables next-generation voice communication solutions with enhanced responsiveness and adaptability. VaxVoIP simplifies complex VoIP development challenges, offering a reliable, secure foundation. It’s an ideal choice for developers aiming to build modern, secure, and intelligent SIP-based communication systems.

This PBX software platform is designed for easy installation and configuration, and it is fully compatible with our telephony hardware. It caters to small and medium businesses (SMBs) that desire a high-quality phone system typically associated with larger corporations, but without the hefty cost! OpenPBX leverages the user-friendly nature and efficiency of desktop PCs, seamlessly integrating them with traditional telephone handsets to provide advanced features and complete control over your communication system. Whether you need to connect to primary rate (E1/T1) trunk lines or standard POTS lines, Voicetronix offers the appropriate line interface hardware. Customers can take advantage of the OpenPBX Do-it-Yourself (DIY) option, allowing them to construct their own phone system using affordable PC servers running Linux and standard analog telephones. By configuring and maintaining their own phone system, users can eliminate the dependence on a technician for setup and support. Additionally, with a web browser, users can effortlessly navigate the system to select and listen to their voicemail messages, making communication management more convenient and efficient than ever before.

HoduPBX is the best solution for business communication. HoduPBX's unique user interface and optimal user experience allow businesses to communicate efficiently with their end users. This software provides unparalleled connectivity in an office environment and remote locations, making communication seamless. HoduPBX empowers agents' work management with multitasking capabilities and efficiency. Our IP PBX software is a great choice for business communications. It supports 1000+ concurrent calls and has 10,000 extensions. HoduPBX IP PBX Software offers various sophisticated features such as Add-on Modules, Reseller Module, SIP Trunking, Queue Pro, Voice Transcription, and Webphone. It also offers white label features across IOS, Android, Chrome Extension, Windows, Mac, Linux, and ZohoPhonebridge Plugin.

*astTECS IPPBX system is the ideal solution for any business, large or small. *astTECS PBX has all the top-end features at an affordable price and is TEC certified by DoT (Govt of India). *astTECS IPPBX solution allows your business to be flexible, innovative, competitive, and future-ready while reducing telecom expenses. *astTECS Call Center Dialer is an omnichannel contact centre solution that offers advanced features and a real-time dashboard for agent monitoring. Advanced analytics and AI-based call center dialer will improve agent productivity and customer engagement. *astTECS provides a complete call recording solution for your company. *astTECS voice recorder is easy to use and offers advanced features. Integration with any PBX is seamless. Supports ISDN/PRI, IP, analog lines and mobile to record all incoming and outgoing calls.

Vodia is a software-based PBX that runs on Windows and Linux. It offers all the advanced features you would expect from a full-featured PBX, including call queues, auto attendants and call forwarding. Vodia is also the pioneer in Multi-Tenancy. You can host Vodia on a dedicated VM( Virtual Machine) in the cloud, premises, and maintain multiple FQDNs for each customer. Sign up for a hosted demo license code for 60 days.

Issabel stands out as an open-source unified communications platform that merges various IP communication services into one cohesive system, effectively overcoming the limitations of traditional PBX systems by offering features like call center functionalities, email integration, management reporting, fax capabilities, video conferencing, audio recording, and CRM integrations. Its call center module is not only robust and flexible but also user-friendly, enabling efficient automation and management of contact centers while adeptly handling both inbound and outbound call campaigns through a straightforward agent interface and call management tools. Additionally, Issabel simplifies the administration of PBX solutions for service providers, allowing them to manage multiple clients from a single instance, thus alleviating the need for individual servers for each client. To further expand its capabilities, Issabel accommodates various add-ons that are designed to integrate seamlessly with the platform, enhancing features that interact with telephony and improve customer service while automating support-related actions. This adaptability makes Issabel a highly effective choice for organizations looking to streamline their communication processes.

Customers can now seamlessly operate from any location thanks to a comprehensive mobility solution that includes essential features for critical operations. Our services allow you to reduce costs through integrated redundancy and the rapid setup of SIP trunks, DIDs, and toll-free numbers. Pulsar360 Corporation’s Voice over IP (VoIP) offerings ensure superior voice communication that can be utilized over cost-effective broadband connections, whether from other providers or through our own managed carrier services. By consolidating their communication needs into a single network connection for both phone and high-speed internet services, clients not only save money but also enhance their productivity. Having dedicated over a decade to refining our VoIP platform, we have developed one of the most advanced and highly automated systems available in North America. This significant progress enables us to provide immediate setup of SIP trunks, hosted and hybrid VoIP PBXs, local and international DIDs, toll-free numbers, and sophisticated management features. With such innovations, businesses can experience a transformation in how they communicate and operate on a daily basis.

Yeastar K2 IPPBX is available as both hardware and software. It supports 2000 users, 500 concurrent conversations, and 400 VoIP trunks across multiple locations. This gives medium and large enterprises full flexibility and capability. Yeastar K2IP PBX is designed to meet the communication needs of today and tomorrow. It offers enterprise-level features and exceptional reliability. It also allows remote working with next-level Linkus UC Softphone.

R.E. Communications provides an IP-PBX system powered by Asterisk, compatible with all standard SIP phones and equipped with high-level features suitable for environments ranging from just 1 phone to as many as 4000. Additionally, these systems facilitate SIP trunking, allowing users to mimic the functionalities of a hosted solution while significantly reducing costs per seat. For instance, a system designed for 10 phones can efficiently operate with only 4 to 6 SIP trunks, resulting in notable savings compared to the expense of a hosted solution for the same number of seats. This versatility makes R.E. Communications’ offerings particularly appealing for businesses looking to optimize their communication infrastructure while managing expenses effectively.

The Taridium IPBX is an advanced software-based VoIP PBX solution that effectively takes the place of conventional hardware PBX systems. It operates seamlessly on standard server infrastructure or via cloud services without necessitating additional licensing costs. Users benefit from an End User Panel that empowers them to manage various aspects of their extensions, including call operations through an operator console, accessing voicemail, setting up call forwarding, and initiating impromptu telephone conferences among other features. You can customize your extension to forward calls only during specific hours by integrating your iCalendar feed or by selecting from a list of preferred caller IDs. Each user is provided with a meet-me style conference room, eliminating scheduling conflicts and enabling live management of conference calls, including options to mute or remove participants as needed. Inviting users is straightforward, requiring just a click to have the system initiate a call to them, thus avoiding the hassle of complicated login procedures. This system redefines user experience by offering intuitive controls and flexibility that enhance communication efficiency.

Our award-winning Avaya IP Office, accessible worldwide, provides a comprehensive unified communications platform that is both implemented and managed on-site. To enhance the interactions and experiences of both your customers and employees, delve into the extensive capabilities available with IP Office. You can analyze customer calls effectively by utilizing features such as call recording, tracking, and reporting. With the Avaya Call Reporting functionality integrated into IP Office, you can capture and document incoming calls seamlessly. Furthermore, you have the option to incorporate real-time call center agent seats and dashboards to optimize call management processes. When you pair Avaya IP Office with a subscription to Avaya Spaces, you unlock a versatile application that encompasses voice, video, messaging, conferencing, and calendar functionalities, all of which are accessible from any device and location. This integration not only streamlines communication but also enhances overall productivity across your organization.

The term "Free" in FreePBX signifies Freedom, highlighting that FreePBX, the leading open-source IP PBX solution globally, empowers users to create a customized phone system that meets their specific requirements. Explore the potential of open-source communication and examine the extensive range of features that FreePBX provides to help you construct your ideal phone system. By integrating commercial modules from Sangoma and various other add-ons, you can elevate your FreePBX installation into a powerful Unified Communications platform. Don't forget to check the Knowledge Base on the FreePBX Wiki for additional resources! Alternatively, you can reach out to FreePBX professionals for assistance with installation, updates, troubleshooting, and more. Available for free download and use, FreePBX's strength is supported by a worldwide community of developers dedicated to maintaining its high compatibility and customizable nature, ensuring it delivers all the essential functions needed to create a scalable business phone system that accommodates any budget. This collaborative effort fosters innovation and continuous improvement, making FreePBX a reliable choice for communication needs.

Founded in January 2015, sipXcom emerged from a fork of the sipXecs project, initiated by the eZuce, Inc. development team. The sipXecs and SIPfoundry communities were struggling to expand their developer base due to a limiting contributor agreement, prompting the creation of sipXcom, which features an open contribution model. The source code for sipXcom is licensed under the copyleft-friendly AGLP v3 (Affero General Public License), ensuring greater flexibility for developers. Currently under development, SWARM is the codename for the upcoming iteration of the sipXcom and sipXecs projects, with a production launch expected in early 2017. This new platform will adopt a microservices-based architecture, enhancing scalability, reliability, and configurability beyond what sipX currently offers. SWARM is engineered to be compatible with any computing environment, including dedicated, virtual, or cloud servers, and also allows for hybrid setups that merge on-premise and cloud solutions. With these advancements, sipXcom aims to significantly improve user experience and functionality in the ever-evolving communication landscape.

MikoPBX offers a comprehensive phone solution characterized by high performance, exceptional stability, and extensive features. It can be deployed on-premises through hardware installation or hosted on various cloud platforms like Google Cloud, Amazon Web Services, and Microsoft Azure. Notably, MikoPBX is a free telephone system that provides all essential business functionalities at no cost. Users will appreciate the intuitive web interface, which includes features such as call recording, conference calling, voicemail, call transfers, call pickup, call queues, an interactive voice response (IVR) menu, as well as customizable inbound and outbound call rules. Additionally, the system includes options for managing holiday schedules and night-time call settings. MikoPBX supports the integration of plugins and modules developed globally, allowing for enhanced functionality tailored to specific needs. Importantly, MikoPBX does not impose user licensing fees, making it an economical choice for businesses looking to expand without incurring extra costs for new staff or call recording capabilities. This flexibility in scaling ensures that businesses can grow seamlessly while maintaining a robust communication infrastructure.

Enterprise interaction can be understood as the integration of collaboration and communication efforts. We facilitate secure enterprise interaction through your own private or hybrid cloud environment. The Universal Cloud Communications Stack (UCCS) introduces an innovative approach to utilize big data architectures, allowing for real-time communications and collaboration, which enhances the scalability of transaction-oriented services on a worldwide basis. Key immediate advantages include rapid deployment and automated management that utilize standard infrastructure frameworks. Furthermore, by utilizing virtualized, secure, and scalable containers, we provide hosted and managed communication and collaboration services with great ease, paving the way for new business models for Managed Service Providers (MSPs) and enabling enterprises to achieve significant cost reductions. In fact, communication systems offer the most substantial opportunity for cost savings across all areas of IT. The synergy of contemporary software architecture and data-driven distributed infrastructure presents a powerful opportunity to revolutionize the industry while meeting evolving business needs.

The REVE Session Border Controller (SBC) is a high-performance, software-driven solution aimed at telecom operators and service providers, designed to secure, manage, and enhance real-time IP communications. Capable of handling up to 70,000 simultaneous calls from a single server, it features a distributed architecture that allows for both media and signaling scalability. The SBC incorporates strong security measures, such as protection against Denial of Service (DoS) attacks, fraud detection, and topology hiding to safeguard communications. Additionally, it enables seamless SIP trunking, supports cloud communications, and enhances mobile connectivity. Among its essential features are intelligent call routing, Least Cost Routing (LCR), normalization of rate plans, and real-time billing accessible via a mobile-friendly interface. Importantly, the system prevents users' balances from going negative by automatically ending calls once the balance hits zero, ensuring users maintain control over their communication expenses. This comprehensive approach to managing IP communications illustrates the SBC's vital role in the modern telecom landscape.

Dial-Office is an adaptable and comprehensive IP PBX software crafted specifically for small and medium-sized businesses as well as start-ups eager to leverage the advantages of VoIP technology. Utilizing the SIP standard framework, this IP PBX software offers a variety of sophisticated phone functionalities that were once limited to expensive, traditional circuit-switched enterprise telephone systems. The Dial-Office IP PBX software ensures dependable and affordable unified communications for small to mid-sized enterprises over a well-established Local Area Network. This award-winning VoIP solution guarantees users exceptional voice clarity and a range of business management tools aimed at boosting productivity, reducing communication expenses, and improving customer service. In addition to these benefits, the platform boasts features that cater to the specific needs of modern businesses, making it a versatile choice for enhancing communication efficiency.

Yeastar P-Series PBX Systems is more than a PBX. It is a converged system designed for SMEs with higher expectations in business communications. The P-Series is a "PBX-Plus-More” solution that unifies voice, video, applications and collaboration. This allows SMEs to achieve more productivity, fill the UC gap and complete more complex tasks.

VitalPBX is a unified communications PBX platform based on Asterisk (Centos 7). VitalPBX is a robust, scalable platform that will allow you to manage your PBX in a simple and intuitive manner. VitalPBX has a simple user interface and tooltips throughout the system. Telephony systems should be aware of security risks and take steps to avoid them. VitalPBX offers a variety of levels of protection against cyber threats in VoIP.

Virtual IP PBX (Private Branch Exchange), or Phone System Appliance – CompletePBX is the heart and soul of Xorcom's hardware and software telephone systems. It's built on top of Asterisk and Linux platforms and is designed for stability and high performance. CompletePBX was built from the ground up to offer ease-of-use, advanced features, and security at all levels. Get a fully functional, free CompletePBX 5 virtual machine evaluation now! CompletePBX certification training is available online for free. It will guide you through the basics of the system. You can save a lot of time and effort if you do it yourself. We can help you get there quicker.

Experience cutting-edge telecommunications software combined with a robust communication network designed to enhance your business capabilities. Our hosted solutions are engineered to elevate both your outbound and inbound calling operations while simultaneously cutting down costs. We offer enterprise-grade SIP Trunking at competitive wholesale prices, ensuring unlimited capacity for high-quality, high-volume outbound VoIP services. By implementing our system, you can lower your office telephony expenses and enhance functionality through features like unique call routing, call recording, and comprehensive reporting. Our specially designed network is equipped to manage incredibly high call volumes while maintaining outstanding call quality and unwavering reliability. If you need to reach more clients, Blue Telecoms provides advanced features such as industry-leading answer machine detection, outbound number rotation, and lead recycling strategies. We utilize only tier 1 VoIP carriers and infrastructure, guaranteeing superior service and reliability. Additionally, our multiple carrier partnerships ensure that your critical projects remain uninterrupted, no matter the circumstances. With Blue Telecoms, you can take your communication capabilities to new heights.

Sangoma Wholesale Carrier Services, previously known as VI Communication Services, has been tailored to enable Internet telephony service providers (ITSPs) and managed service providers (MSPs) to deliver the high-quality services their clients desire seamlessly. Companies can utilize our communication APIs to integrate voice and messaging capabilities into their applications, facilitating effortless interaction with their customers. As an ITSP, MSP, or value-added reseller (VAR), you can enhance the offerings for your current retail SIP trunking clientele by providing additional phone numbers and VoIP services that may not be accessible within your current network, thereby generating new revenue opportunities. Alternatively, if you're just beginning your journey, you have the option to launch a new retail SIP trunking service and incorporate the features you wish to provide to your clients, including messaging, conference bridging, and a variety of other functionalities to enhance their experience. This flexibility allows businesses to stay competitive in a rapidly evolving market.

thinQ provides a "better" way to power voice and SMS/MMS messaging applications in the cloud. Get up and running quickly with our integrated thinQ io portal and APIs. We make it easy to bring your 40 high-quality carriers to Twilio and Nexmo, Plivo and SignalWire, as well as enterprise contact centers and telecom VoIP providers. Our APIs allow you to quickly add and control inbound and outbound voice, texting, and other features to your existing tech stack. You can also connect to existing SIP providers to improve call delivery and quality. Our fully integrated thinQ.io Web platform makes it easy to purchase and port phone numbers, create route profiles, trunks, enable text messaging, as well as create route profiles, trunks, and other functions. You'll find a "better" way to communicate if you add call-specific analytics, SIP ladders, and our five-star support team to mitigate, route around outages, create customized routes and blocks, etc.

IdentityServer serves as a versatile and standards-adhering framework for OpenID Connect and OAuth 2.0 tailored for ASP.NET Core, ensuring complete control over user interface, user experience, business logic, and data management. It has received official certification from the OpenID Foundation and encompasses a wide array of protocols and standards developed by the OpenID Foundation and IETF working groups. Additionally, it presents limitless hosting possibilities, facilitating deployment on-premises, in cloud environments, behind VPNs, and on various platforms such as Windows, Linux, Docker, or Kubernetes. Duende Software enhances this offering with supplementary products, including IdentityServer for Redistribution and a Backend for Frontend (BFF) security framework. To assist developers in effectively utilizing these solutions, extensive documentation and training resources are readily accessible. Our commitment to being fully compliant with standards is paramount, and we strive to provide you with comprehensive access to all facets of the OAuth and OpenID Connect protocol family. Furthermore, this approach empowers organizations to implement secure authentication and authorization processes tailored to their specific needs.

The Cisco Unified Border Element (CUBE) serves as a robust Session Border Controller (SBC) designed for enterprises, enabling secure, scalable, and adaptable voice and video communications between corporate networks and provider SIP trunks. It encapsulates four essential roles: session management, security enforcement, protocol interworking, and boundary definition. CUBE effectively facilitates signaling compatibility between SIP and H.323 protocols and manages media interworking, which includes DTMF, fax, modem support, and codec transcoding, while also executing address and port translations to ensure privacy and conceal network topology. Additionally, it offers features such as billing and call detail record (CDR) normalization, quality-of-service (QoS) optimization, bandwidth management, as well as media services that include forking and proxying capabilities. Moreover, CUBE can be implemented either on Cisco routers or as a virtual solution on Cisco UCS servers, accommodating centralized, distributed, or hybrid deployment strategies, which makes it versatile for various organizational needs. This adaptability allows businesses to tailor their communication infrastructure to meet evolving demands effectively.

Digital transformation in the realm of communication and collaboration solutions is vital for modern enterprises. By harnessing the concepts of Software-Defined Networking (SDN), AudioCodes has developed the Software-Defined Voice Network (SDvN), which serves as an open solution aimed at assisting large organizations in their digital transformation journey. This innovative platform simplifies voice network operations and enhances the efficiency of call routing. It accomplishes this by integrating various communications silos, enabling connectivity with SIP trunk services, and optimizing call routing through the use of the corporate IP network. Additionally, SDvN combines AudioCodes' session border controllers (SBCs) with comprehensive end-to-end call routing, policy management, user management, and tools for managing voice networks. As a result, businesses can establish a unified network that seamlessly integrates both on-premises and cloud-based communication solutions, eliminating the need to overhaul existing systems. This flexibility ensures that enterprises can modernize their communication infrastructure while maintaining their current investments.

The D-TAC "HD" series billing platform offers a comprehensive post-paid CDR billing solution that seamlessly integrates with traditional switches, IP switches, gateways, proxy servers, soft switches, and border session controllers. This robust platform is tailored to deliver high-quality billing for both VoIP and TDM-based CDR networks. D-HD equips telecom businesses with specialized billing and automation features that simplify operational management. Each division within your company receives customized screens, functionalities, and tools aimed at enhancing overall productivity. From customer support systems to rate management resources, the D-HD system is designed to facilitate efficient task execution. For those in need of CDR management capabilities, advanced invoicing options, cash receipt tracking, carrier cost analysis, ASR reporting, and statistical insights into route or trunk group performance, the D-TAC HD Billing System serves as the optimal solution. Central to our offering are billing and authentication services, which include comprehensive AAA RADIUS capabilities and additional AARA features, ensuring a well-rounded approach to telecom billing. Furthermore, this system is adaptable to evolving industry demands, making it a reliable choice for future growth.

Wavetel was founded in 2010 and has since grown to be a major provider of telecom services in the UK, and around the globe. Wavetel began as a wholesale operator that offered Voice, SMS, and Data services to other operators. Wavetel's B2B services were launched in 2013 by the company. These include VoIP telephony services and Connectivity, Virtual Numbers. SIP Trunks. Microsoft Teams Direct Routing. Mobile, SMS, Call Center Solutions are available to its valued clients worldwide. Our voice messaging solutions have evolved over the years to be an enterprise solution. It can integrate with third-party CRMs and business apps through our programmable APIs.

Alianza's Perimeta Session Border Controller (SBC) is a cloud-native and entirely software-based solution tailored to deliver secure, scalable, and adaptable voice and multimedia communication services. This SBC accommodates a range of deployment options, including VoLTE, VoNR, SIP trunking, and direct routing for Microsoft Teams, making it ideal for fixed-line, wireless, and enterprise interconnect scenarios. Operating on a zero-trust security framework, Perimeta provides exceptional protection for cloud infrastructures. Its innovative separation of media and signaling planes facilitates precise scaling in response to fluctuating demands. Additionally, it incorporates a secure distribution engine (SDE) for efficient load balancing and distributed admission management (DAM), which simplifies large-scale deployments. Compatible with various cloud environments—public, private, and hybrid—the SBC can be implemented on platforms like AWS, Azure, OpenStack, VMware, and ATCA hardware. This versatility ensures organizations can choose the optimal environment for their specific needs, enhancing operational efficiency and security.

A2Billing serves as a class 4 and class 5 softswitch featuring inline billing capabilities, aimed at delivering VoIP services for residential, business, and wholesale markets, along with support for calling cards, call-back services, and telephone number resale, all underpinned by our expert support. This free and open source software, tailored for Asterisk, facilitates comprehensive telecom customer management through various interfaces, including those for administrators, agents, customers, and online signups, while offering adaptable inline rating and billing for calls and services in real-time, ensuring a seamless experience. Additionally, its robust features and support structure make it a popular choice for telecom providers looking to enhance their offerings.

Arch is a sophisticated gateway designed to safeguard, monitor, and tailor AI agents through effortless API integration. Leveraging the power of Envoy Proxy, Arch ensures secure data management, intelligent request routing, comprehensive observability, and seamless connections to backend systems, all while remaining independent of business logic. Its out-of-process architecture supports a broad range of programming languages, facilitating rapid deployment and smooth upgrades. Crafted with specialized sub-billion parameter Large Language Models, Arch shines in crucial prompt-related functions, including function invocation for API customization, prompt safeguards to thwart harmful or manipulative prompts, and intent-drift detection to improve retrieval precision and response speed. By enhancing Envoy's cluster subsystem, Arch effectively manages upstream connections to Large Language Models, thus enabling robust AI application development. Additionally, it acts as an edge gateway for AI solutions, providing features like TLS termination, rate limiting, and prompt-driven routing. Overall, Arch represents an innovative approach to AI gateway technology, ensuring both security and adaptability in a rapidly evolving digital landscape.

Sansay's VSXi Session Border Controller (SBC) is a robust, software-driven solution aimed at communication service providers, delivering essential features such as security, NAT, protocol normalization, and effective session management. Capable of handling up to 500,000 concurrent calls and managing 1 million SIP registrations, it ensures scalability for extensive deployments. The VSXi excels in facilitating SIP trunking applications by efficiently managing Least Cost Routing and NAT traversal needs. Additionally, it provides programmable SIP trunking functionalities, which support both authenticated and unauthenticated trunks, incorporating features like extension-to-extension dialing, call forwarding, and failover mechanisms. Furthermore, it allows seamless integration with Microsoft Teams via Direct Routing, enabling users to make, receive, and transfer calls across Teams and the public switched telephone network (PSTN). The flexibility of the VSXi extends to its deployment in cloud environments, and it boasts APIs that support extensive integration and customization options, catering to the diverse needs of service providers. This versatility makes the VSXi an invaluable asset in enhancing communication infrastructures.

Unified Communications, OTT, and telecom providers can configure, launch, and monetize any type communication service in IP networks. This includes Unified Communications, Hosted IP PBX / IP Centrex and SIP Trunking. DID Management, DID Management, OTT Voice and calling cards. VoLTE/VoWiFI are some of the key functional modules. Key functional modules: Class 4 & 5 SIP softswitch, for residential and enterprise telephony service; convergent online billing, service provisioning and charging (optional).

Wiretap Telecom provides an extensive range of VoIP and SIP trunking services designed for businesses in need of scalable, dependable, and secure communication options. Our intuitive user portal enables effortless management of services, featuring tools such as real-time analytics for calls and faxes, comprehensive trunk operation logs, and efficient call routing management. Additionally, it supports diverse messaging solutions like Email2SMS, SMS App, and smooth PBX integration, which caters to various messaging requirements. With the capability for international calls to any country and the convenience of managing multiple accounts through a single platform, Wiretap Telecom guarantees global connectivity and efficient operations. Our SIP trunking services work seamlessly with a variety of PBX systems, including FreePBX, Yeastar, 3CX, and Asterisk, making integration simple while enhancing voice functionality. By prioritizing user experience and operational efficiency, Wiretap Telecom stands out as a leader in innovative communication solutions for businesses.

Cool Telecom offers a cloud-based phone system tailored for businesses of all sizes, featuring a versatile interface that integrates CRM and Helpdesk functionalities to boost sales and enhance communication for support teams. Clients can quickly acquire phone numbers from over 50 countries, adjust their workforce according to seasonal demands, and benefit from comprehensive insights via real-time analytics. The platform is accessible on any device, allowing users to make and receive calls from virtually anywhere. With the multifunction app from Cool Telecom, users can efficiently manage call routing based on IVR choices, agent expertise, time zones, and other criteria, all facilitated by a user-friendly dashboard. Additionally, the platform provides advanced, real-time analytics regarding agent and team performance, enabling monitoring through a live feed and allowing data cross-referencing with existing CRM systems for deeper insights and opportunities for enhancement. This functionality ensures that businesses not only remain connected but also continuously improve their customer service and operational efficiency.

UploadThing is a versatile, open-source solution for file uploads tailored specifically for full-stack applications using TypeScript. It streamlines the integration of file uploads by enabling developers to set up File Routes that define permitted file types, size limits, and the number of uploads allowed. The platform offers SDKs compatible with a range of frameworks, such as Next.js, Astro, SolidStart, SvelteKit, and Nuxt, along with backend support for Express, Fastify, H3, and more. Client-side file uploads are made easy through the use of components and hooks, allowing for smooth integration with popular frontend libraries like React and Vue. With a focus on both security and ease of use, UploadThing manages authentication on the developer's server while overseeing the file upload process on its own infrastructure. This method ensures that server-side authorization can be achieved without the burden of file processing on the developer's end, ultimately enhancing the overall user experience. Furthermore, this solution is ideal for developers seeking a reliable and efficient way to handle file uploads in their applications.

The Session Border Controller (SBC) from Ekinops, which is certified by Microsoft, serves as an enterprise solution aimed at ensuring security and interoperability with top IP-PBX systems and major SIP environments. It can be deployed as a virtual network function (referred to as ONEvSBC) or as part of Ekinops' integrated voice and data Multi-Service Access Routers (known as ONeSBC), facilitating SIP trunking applications such as Microsoft Teams Direct Routing and Unified Communications. This SBC is equipped with robust security mechanisms that include network demarcation, hidden topology, and NAT traversal, enhancing network safety. Additionally, it supports a wide range of audio codecs, both wideband and narrowband, through media transcoding, which guarantees superior voice quality by prioritizing voice and video traffic, along with implementing Call Admission Control (CAC). Furthermore, Ekinops’ SBC is fully compliant with the SIP Connect 1.1 and 2.0 standards, which helps minimize service validation delays, thus expediting the integration process for new PBX vendors. This comprehensive solution not only enhances communication efficiency but also ensures a seamless user experience across various platforms.