Alternatives to Ribbon PSX SWe

CallTrackingMetrics is the only SaaS platform that uses call tracking and conversion intelligence to inform contact center automation--resulting in a more personalized customer experience. Find out which marketing campaigns are generating leads or conversions and use that data for automated call flows and to power your contact centre. Our phone, text, online, and live chat tools allow you to unify communications across your organization. CallTrackingMetrics is trusted by more than 100,000 users worldwide to manage communications for their sales, marketing, and service teams. Call tracking features include reliable dynamic numbers insertion (DNI), for session-level attribution, local and toll-free tracking numbers, and omnichannelattribution across calls, texts and form fills. Contact center features include a browser-based softphone and smart routing options.

Ribbon’s EdgeMarc IP Business Gateway Appliances offer an economical solution for service providers, managed service providers, and resellers to effectively manage bundled voice and data services. These devices merge internet access of up to 1 Gbps with tools for ensuring voice security and monitoring quality of service. The product lineup features various options, including gateways for analog lines and stations (FXO/FXS) as well as power over Ethernet (PoE) capabilities. Centralized configuration and management are facilitated through the EdgeView Service Control Center, enabling seamless tracking of the customer experience across hundreds or even thousands of locations. Positioned at the customer’s site, EdgeMarc serves as a critical asset for service providers, providing insights into issues and streamlining troubleshooting processes, while also delivering detailed real-time access to each location and comprehensive reporting tools for network-wide management. This comprehensive solution not only enhances operational efficiency but also significantly improves service reliability and customer satisfaction.

Our proprietary technology has been providing telecom services since more than 20 years. We are experts in Hosted IP-PBX and SIP Trunks, Call Center Software, Wholesale, Data Services, etc. Our exceptional service, rates, and products meet the FCC standards for STIR/SHAKEN network compliance. The XCast Unified Communications is Scalable and International with unlimited, free, planet-wide, extension-to-extension, all with a web-based Administrator Control Panel (ACP) to help you manage all your features. Promotional plans: Hosted PBX Basic - $9.95/month Sip Trunking Basic - $4.95/month

ProSBC stands out as a robust, carrier-grade Session Border Controller tailored for both service providers and enterprises, offering advanced functionalities. Capable of handling up to 60,000 simultaneous sessions, it boasts features such as media transcoding, STIR/SHAKEN compliance, fraud detection, and configurations designed for high availability. Its versatility allows it to operate seamlessly across a range of platforms, including VMware, KVM/Proxmox, AWS, Microsoft Azure, and bare-metal servers. This makes ProSBC an ideal choice for diverse applications like SIP trunking, enterprise SIP trunking, CPaaS access, and TDM-to-SIP migration. Furthermore, it is designed to be interoperable with leading VoIP platforms and carrier networks, ensuring wide compatibility. The system is user-friendly, with a web-based interface that simplifies deployment, operation, and management. Additionally, the optional TB Analytics feature pack provides valuable insights for diagnosing network issues. Importantly, the software's scalability—from a modest 100 sessions to a robust 60,000—ensures that your SBC can grow alongside your needs, providing flexibility and reliability as your demands evolve. This adaptability makes ProSBC an excellent investment for future-proofing communications infrastructure.

FRAFOS has developed the ABC Session Border Controller (SBC), a versatile solution tailored for VoIP service providers and enterprises, which can operate seamlessly on high-performance hardware, dedicated appliances, and virtual environments. This SBC provides robust session border control, signaling mediation, call routing, and sophisticated media server functionalities, effectively catering to the evolving demands of VoIP and Next Generation Networks (NGN). Among its notable features are the Cluster Configuration Manager (CCM) for centralized management of multiple SBCs, integrated support for announcements and transcoding directly on the device, and the ability to record calls using SIPREC. Additionally, the ABC SBC bolsters network security through mechanisms such as rate limiting, content control, topology hiding, and media management, which collectively mitigate risks of fraud and denial-of-service attacks. Furthermore, it is capable of handling up to 5,000 concurrent calls on standard hardware and provides load balancing for VoIP traffic, making it a highly effective tool in modern communication infrastructures. Its comprehensive feature set ensures that it meets the demands of a wide range of users, from small enterprises to large service providers.

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.

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.

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.

Teleinx stands out as a cutting-edge telecommunications and data solutions provider, delivering top-tier voice, data, and SMS services tailored for carriers, CLECs, PBX providers, call centers, and MVNOs. Among their offerings are wholesale outbound voice termination services that encompass both domestic and international calls, as well as short-duration termination and toll-free termination within the USA. They facilitate the acquisition of DID and toll-free numbers, allowing businesses to create a local or national footprint with rapid provisioning and SMS/MMS functionality. In addition, Teleinx provides advanced messaging solutions featuring APIs for SMS and MMS, which support shortcodes, toll-free numbers, and local numbers to significantly boost customer interaction. Their comprehensive robocall mitigation services incorporate SIP interconnect, ANI and DNIS filtering, and STIR/SHAKEN compliance measures, ensuring that networks are safeguarded against fraudulent activities. By equipping businesses with these robust services, Teleinx helps them navigate the complexities of modern communication while enhancing their operational efficiency.

The Mediatrix Sentinel CS is an advanced Session Border Controller (SBC) tailored for small to medium-sized enterprises, capable of being deployed in virtual settings whether on-site or within the cloud. It employs the same Microsoft Teams-certified DGW software as its hardware counterparts, guaranteeing smooth integration and support for Direct Routing capabilities. Among its notable features are SIP normalization, network demarcation, survivability, and a range of security protocols, including TLS/SRTP encryption and SIP-based firewalls designed to thwart DoS attacks. The Sentinel CS can manage over 6,000 concurrent sessions and accommodate more than 15,000 registered users without the need for extra licensing. To maintain service availability, it intelligently reroutes calls to secondary servers or the PSTN in case of primary server failures. Additionally, it offers QoS monitoring, remote mass management, and sophisticated troubleshooting tools, which not only help in reducing network maintenance costs but also enhance overall customer service quality. This makes the Sentinel CS an ideal choice for businesses looking to ensure reliable communication solutions amidst varying operational demands.

The Ribbon SBC Software Edition (SBC SWe) is a versatile, cloud-native Session Border Controller that enhances and secures real-time communications in a variety of deployment scenarios. It provides the same comprehensive functionality found in Ribbon's hardware-based SBC 5400 and SBC 7000 models, featuring capabilities such as extensive audio codec transcoding, signaling interworking, call admission control, and robust security protocols, including encryption and defenses against Denial of Service attacks. The SBC SWe can be implemented as a Virtual Machine (VM) on standard industry servers in data centers, as a Virtual Network Function (VNF) within private clouds utilizing OpenStack, or on public cloud platforms like Amazon Web Services (AWS), Microsoft Azure, and Google Cloud Platform (GCP). This adaptable deployment model enables organizations to efficiently scale their communication infrastructure without compromising on performance or security. Additionally, the SBC SWe's cloud-native architecture ensures that enterprises can easily integrate it into their existing systems and workflows, fostering seamless communication across diverse environments.

ULAP SBCaaS (Session Border Controller as a Service) is an innovative, policy-driven solution for voice and video that empowers businesses to safely implement IP-based real-time communication services, such as VoIP, Unified Communications (UC), and Customer Experience (CX) in a cloud environment. Serving as the initial protective barrier against various cyber threats, including service theft, spoofing, and distributed denial-of-service (DDoS) attacks, ULAP SBCaaS efficiently manages SIP traffic at the periphery of your network, linking your systems with external networks while performing essential functions like protocol translation, connection security, and traffic management to ensure smooth and secure communications. Additionally, it facilitates effortless integration with popular digital office tools such as Microsoft Teams and Zoom, thereby providing a unified communication experience across all devices. With robust security protocols in place, ULAP SBCaaS safeguards virtual communications from threats like eavesdropping and DoS attacks, ensuring that sensitive information remains protected. This comprehensive solution not only enhances communication security but also streamlines the operational efficiency of various business processes.

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.

The NetVanta 4000 Series is perfectly suited for carrier bundled service solutions reaching speeds of up to 1 Gbit/s, as well as providing enterprise-level internet access that ensures secure and rapid connectivity for businesses. Various models in this series come equipped with features such as power over Ethernet (PoE) integrated into the built-in switch, VPN capabilities, and specialized voice monitoring services. Additionally, every NetVanta 4000 unit is capable of functioning as an eSBC, offering a comprehensive application platform directly at the customer's location. Moreover, these devices can scale through software enhancements as your VoIP requirements increase, accommodating up to 1,000 concurrent calls. The NetVanta 4148 model specifically includes four routed, autosensing Gigabit interfaces, with the option of connecting two via fiber, alongside an 8-port Gigabit Ethernet switch. These Gigabit interfaces are versatile, allowing for both LAN and WAN configurations, and provide Ethernet redundancy with instant failover in case one link experiences an outage. The model number for this unit is 17004148F1, and when upgraded with the enhanced feature pack, the NetVanta 4148 can efficiently handle up to 500 VPN tunnels, making it a robust choice for organizations needing reliable and scalable networking solutions.

Cataleya's Orchid Link serves as a highly adaptable, software-driven Session Border Controller (SBC), aimed at improving real-time communication for both service providers and enterprises. It provides a range of deployment methods, such as on-premises setups and as-a-Service options, catering to various operational requirements. Orchid Link incorporates WebRTC gateway capabilities, allowing for the effortless integration of web-based voice and video communications into pre-existing SIP systems. This feature enables the expansion of communication services to web browsers without the need for major upgrades to the core infrastructure. Additionally, the SBC prioritizes security by implementing DTLS for secure session initiation and SRTP for media encryption, safeguarding communications between WebRTC endpoints and SIP networks. Its intelligent service-aware firewall acts as a proactive defense mechanism, offering protection against potential cyber threats while ensuring smooth communication flow. Overall, Orchid Link provides a comprehensive solution that enhances communication efficiency and security for modern organizations.

Mitel's OpenScape Session Border Controller (SBC) is a highly adaptable solution that offers extensive SIP interoperability, sophisticated media management, and strong security features. Tailored for VoIP infrastructures, this controller efficiently terminates SIP sessions on the WAN side, which helps ensure that traffic comes from verified sources while also scrutinizing SIP and media packets for any protocol discrepancies or irregularities. The OpenScape SBC actively regulates firewall "pin holes" for RTP and SRTP media streams. Its compatibility extends to Mitel’s MiVoice Business, MiVoice MX-ONE, and MiVoice 5000, in addition to OpenScape Voice, OpenScape 4000, and OpenScape Business, making it a versatile choice for organizations of all sizes. This solution accommodates various deployment methods, including virtualized software for VMware ESXi, and is set to support commercial off-the-shelf hardware like Lenovo servers in the near future, alongside a gateway option featuring Kontron servers. As such, it provides organizations with the flexibility to choose the most suitable infrastructure for their specific requirements.

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.

Ericsson's Session Border Controller (SBC) is a high-performance, virtualized solution aimed at ensuring both security and interoperability between IMS and various networks, managing both signaling and media effectively. It serves as a comprehensive solution for a range of services including Voice over LTE (VoLTE), Voice over Wi-Fi (VoWiFi), Video over LTE (ViLTE), Rich Communication Services (RCS), fixed VoIP, interconnection, and web communication applications. This SBC can be implemented on virtually any x86 hardware across several cloud platforms, such as OpenStack, VMware, and Ericsson NFVI. It is also validated on Ericsson's software-defined infrastructure (SDI) and can be part of standardized industrial solutions. The architecture of the SBC leverages centralized control while managing the media plane in areas of high traffic, which helps to reduce media latency and optimize transmission costs. With its extensive features and convergent capabilities, the Ericsson SBC allows service providers to consolidate their needs into one product, leading to considerable operational expenditure (OPEX) reductions. This unification of services not only enhances efficiency but also simplifies the management of diverse communication solutions for operators.

The Avaya Session Border Controller for Enterprise (SBCE) is a powerful tool specifically crafted to enhance and safeguard real-time communications within corporate networks. It plays a vital role in environments that utilize SIP-based Unified Communications (UC), offering key features like security, interoperability, and the ability to scale. The SBCE is offered in two distinct versions: Standard Services and Advanced Services, tailored to meet diverse organizational requirements. In a high availability (HA) setup, SBC servers are arranged in pairs, with one functioning as the primary unit and the other as the backup. Both units are managed by either a single Element Management System (EMS) or a replicated pair of EMS. This solution enables customers to leverage Avaya’s extensive expertise in deploying SIP trunks and efficiently managing a large user base within enterprises. Furthermore, organizations can ensure uninterrupted communication even during maintenance or unexpected failures, making it an essential asset for any enterprise focused on reliability.

Nokia's Session Border Controller (SBC) plays a crucial role in safeguarding signaling and control media as they traverse the boundaries of the IMS network. Positioned at the access network edges, it secures various forms of IP access while facilitating the delivery of all IMS services effectively. Additionally, it is located on the periphery of peering networks, enabling seamless roaming, IMS interconnection, and corporate SIP trunking operations. The SBC encompasses a range of functions, including the encryption and decryption of bearer traffic, firewall capabilities, and topology hiding for enhanced security, as well as media handling, transcoding, SIP interworking, and NAT traversal. It may be implemented as an independent SBC or integrated into Nokia’s VoLTE/VoWiFi framework. For VoIP services aimed at residential and enterprise customers, the SBC is equipped with robust protections against DoS/DDoS attacks, ensures network topology concealment from access points, facilitates NAT traversal, and supports various codecs, notably G.729, G.722, and T.38 Fax. Furthermore, within a VoLTE core network, it regulates access and supports mobility features, offering an EVS codec and compatibility with eSRVCC, VoWiFi, NPLI, S8HR roaming, and additional functionalities, thus enhancing the overall communication experience.

The Ingate SIParator® serves as an effective and adaptable Enterprise Session Border Controller (E-SBC) designed for secure and interoperable SIP connectivity, enabling seamless integration of PBXs and Unified Communications (UC) systems with SIP trunking providers. By streamlining SIP trunking processes, the SIParator facilitates the connection of remote UC endpoints, the aggregation of SIP trunks, and the distribution of sessions across various sites and service delivery points. It is widely used for securing Real-Time communications, ensuring SIP interoperability, and providing extensive connectivity options. This robust solution is compatible with all existing network infrastructures and includes a standard SIP proxy along with a SIP registrar. Furthermore, it supports Network Address Translation (NAT) and Port Address Translation (PAT), as well as Transport Layer Security (TLS) and Secure Real-Time Transport Protocol (SRTP) for encrypting both SIP signaling and media. By addressing common security concerns related to enterprise VoIP, the SIParator ensures a safe and dependable communication environment for organizations. Its versatility and reliability make it a preferred choice for businesses looking to enhance their communication frameworks.

PXO+ softphone features: (1) PXO+ is the best softphone for RTC (2) PXO+ has hundreds of configurable features and is very flexible (3) PXO+ is perfect for RTP and SIP feature testing (4) PXO+ supports more than 25 speech and audio codecs (5) PXO+ supports more than 8 video codecs (6) PXO+ provides scripting via VoiceXML (7) PXO+ works well for RTC QA and VoIP scenarios (8) STIR/SHAKEN

Alcatel-Lucent Enterprise's OpenTouch Session Border Controller (SBC) is a robust software solution that ensures the security of SIP trunks and enterprise communications. Acting as the critical boundary between the business and SIP trunking services, it defends against threats such as VoIP attacks, SIP denial of service, fraud, and eavesdropping. The OpenTouch SBC delivers secure and scalable connectivity for both SIP and media, featuring capabilities like audio transcoding and NAT traversal for enhanced audio and video interactions. This solution guarantees affordable and secure communication over the internet and with various SIP service providers. Additionally, the SBC incorporates a web-based management system with pre-built configuration templates, simplifying the setup process to just a few clicks for certified SIP trunking providers. With its ability to provide cost-effective interoperability, the SBC supports numerous protocol adaptations for a wide range of SIP trunking providers, ensuring flexible and efficient communication solutions for enterprises. Overall, the OpenTouch SBC stands out as a comprehensive solution for enterprises looking to enhance their communication security and efficiency.

The Microsoft MCP Gateway serves as an open-source reverse proxy and management interface for Model Context Protocol (MCP) servers, facilitating scalable and session-aware routing along with lifecycle management and centralized oversight of MCP services, particularly within Kubernetes setups. Acting as a control plane, it adeptly directs requests from AI agents (MCP clients) to the corresponding backend MCP servers while maintaining session affinity, effectively managing multiple tools and endpoints through a singular gateway that prioritizes authorization and observability. Additionally, it empowers teams to deploy, update, and remove MCP servers and tools through RESTful APIs, enabling the registration of tool definitions and the management of these resources with security measures such as bearer tokens and role-based access control (RBAC). The architecture distinctly separates the management of the control plane, which includes CRUD operations on adapters, tools, and metadata, from the data plane's routing capabilities, which support streamable HTTP connections and dynamic tool routing, thus providing advanced features like session-aware stateful routing. This design not only enhances operational efficiency but also fosters a more secure environment for managing AI services.

The Sangoma Enterprise Session Border Controller (SBC) is a hardware solution crafted to enhance and protect real-time communications specifically for medium to large enterprises. Capable of handling between 25 to 250 simultaneous calls, it incorporates strong security measures such as TLS and SRTP encryption to safeguard against eavesdropping, thus ensuring safe interactions between IP phones and IP PBX systems. This Enterprise SBC enables smooth transitions across various phone systems or SIP trunks, guaranteeing continuous service even in the face of severe disruptions. Furthermore, it provides a secure link for remote users to connect to the corporate phone system without requiring a VPN, significantly boosting both flexibility and accessibility. This combination of features makes the Sangoma SBC an essential tool for businesses aiming to maintain secure and reliable communication channels.

Dialogic BorderNet Session Border Controllers (SBCs) are powerful and adaptable solutions engineered to secure and oversee real-time communications within a variety of network settings. These SBCs utilize a cohesive software platform that is compatible with several deployment options, such as commercial off-the-shelf (COTS) hardware, virtualized environments like VMware and KVM, as well as public cloud systems, including Amazon EC2, thereby ensuring both investment security and flexible deployment. BorderNet SBCs can scale efficiently, accommodating anywhere from 25 to 100,000 sessions per instance and managing up to 1,000 sessions per second, which makes them ideal for both smaller and larger operational needs. Additionally, they offer extensive security functionalities, including topology hiding, protection against denial-of-service (DoS) attacks, and support for Transport Layer Security (TLS) and Secure Real-Time Transport Protocol (SRTP), enabling safe encrypted signaling and media. Furthermore, their robust architecture allows organizations to adapt to changing communication demands without compromising security or performance.

Countless service providers and businesses globally rely on our solutions for transforming networks with a focus on driving business growth. When it's essential to connect Teams with the wider world without any hiccups, Ribbon's Session Border Controllers (SBCs) enable secure external business calls. Transitioning to a cloud-native model allows for the consistent delivery of rapid services and increased revenue, whether utilizing a public or private cloud. Enhance your customers' experiences with secure, automated communication services that meet carrier-grade standards. Swiftly design and implement innovative services within a multi-layered network environment. Ribbon's IP Optical networks are characterized by their openness, automation, and agility, integrating optimized hardware with sophisticated automation software. The IP Wave portfolio offers a well-rounded solution in IP Optical technology, specifically designed to tackle the complexities of constructing, managing, and operating multi-layer communication networks. Discover the unique advantages that IP Wave presents in comparison to other available solutions on the market. By choosing IP Wave, you position your organization at the forefront of telecommunications innovation.

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.

Introducing a cutting-edge TelePhony Platform tailored for individual enterprises, wholesale distributors, and residential VoIP service providers! This open-source option allows anyone to kickstart their telephony venture without any initial investment through ASTPP. As a highly effective VoIP Billing Software, ASTPP aims to enhance user experience by offering a well-rounded telephony solution. Having evolved from a basic billing platform to “A Smart TelePhony Platform,” the latest iteration of ASTPP includes sophisticated functional modules designed to consolidate multiple services necessary for operating a VoIP business. It seamlessly incorporates hosted IP PBX systems, Class 4/5 Softswitch capabilities, and an all-encompassing invoicing and billing framework powered by Smart Technology. Users can efficiently manage origination and termination rates separately, unlocking revenue potential through both least cost routing and cost-based routing strategies. This advancement not only simplifies operations but also maximizes profitability in the competitive VoIP market.

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 NetVanta 6250 Series is engineered to provide integrated voice and data solutions, encompassing both legacy T1 connections and Ethernet services. It employs SIP or MGCP protocols for VoIP functions, ensuring compatibility with top-tier soft switches, servers, and gateways. These units serve as gateways that transform IP signaling from the carrier into conventional TDM analog and digital voice services efficiently. This capability allows the NetVanta 6250 Series to deliver voice services concurrently to both IP telephones and traditional telephony systems. Each device in the NetVanta 6250 lineup features a Gigabit combo port suitable for either copper or fiber connections, four Fast Ethernet ports for WAN or LAN connectivity, a USB port for wireless backup or storage solutions, and four T1 ports tailored for legacy PBX systems. Moreover, users can enhance functionality through additional software upgrades that introduce SBC features, VPN capabilities, and voice quality monitoring (VQM), along with a user-friendly graphical interface that visualizes data related to the mean opinion score (MOS). The versatility and upgrade potential of the NetVanta 6250 Series make it a robust choice for modern communication needs.

AudioCodes presents the Mediant Cloud Edition (CE), a cloud-native Session Border Controller (SBC) that is engineered to deliver scalable, secure, and cost-efficient voice and video communication services within virtualized settings. Its automatic elasticity feature enables dynamic resource scaling according to real-time traffic needs, guaranteeing peak performance without the necessity for manual adjustments. Sharing the same foundational code as AudioCodes' physical SBCs, the Mediant CE offers extensive SIP interoperability, sophisticated media processing capabilities like transcoding, and strong security measures. The architecture, based on microservices, along with a scalable media cluster, allows for rapid and economical deployment of new communication services. Furthermore, installation is simplified via native tools such as AWS CloudFormation, which allows for seamless integration with existing automation and DevOps processes, enhancing operational efficiency and responsiveness to market changes. This makes the Mediant CE an ideal solution for organizations looking to modernize their communication infrastructure.

We provide a comprehensive, cloud-based virtual telephone system that encompasses features such as voice menus, voicemail-to-email, conferencing, customizable time and caller routing, internal extensions, and call forwarding tailored by time and day, among other capabilities. This innovative solution liberates you from the constraints of conventional phone systems, enabling effortless global connectivity. There’s no need for any specialized hardware or software, making it accessible for anyone. Our platform is user-friendly and compatible with any service provider, ensuring smooth integration. When you purchase at least one local or toll-free virtual phone number from SendMyCall, you will receive a complimentary advanced business phone system loaded with features. This system allows you to redirect incoming calls to your employees no matter where they are in the world, accommodating any phone type they might use, whether it be a landline, mobile, VOIP, or softphone. Additionally, our Local Virtual Numbers grant your business a global presence across over 60 countries, enhancing your reach and professionalism. With our service, you can experience the freedom to manage your communications effectively from any location.

**CallAtlas.io: Revolutionizing Call Tracking Software** CallAtlas.io is a cutting-edge call tracking software that provides businesses with the tools they need to manage their inbound calls effectively. With features such as international local and toll-free phone numbers, a paper call network, call recording, and call routing capabilities, CallAtlas.io is the go-to solution for businesses looking to streamline their call management processes. **Key Features:** - **International Local and Toll-Free Phone Numbers:** CallAtlas.io offers a wide range of international local and toll-free phone numbers, allowing businesses to establish a local presence in multiple countries. This feature enables businesses to expand their reach and attract customers from around the world. - **Paper Call Network:** CallAtlas.io's paper call network allows businesses to generate leads through print advertising. By assigning unique phone numbers to different ads, businesses can track the effectiveness of their print campaigns and make informed decisions about their advertising strategies. - **Call Recording:** With CallAtlas.io, businesses can record inbound calls for quality assurance and training purposes.

Nimbata is the #1 call tracking and conversion attribution platform enabling marketers and businesses to track phone calls generated by their marketing campaigns. It provides features such as call recording, call tracking, and call analytics to give businesses insights into the performance of their marketing campaigns across multiple channels, including website, social media, PPC advertising or even SEO. Nimbata offers a range of native integrations with the most popular tools in your martech stack, such as Google ads, Meta Ads, GA4, Hubspot, Salesforce, and Slack, as well as an API that enables you to help businesses streamline their workflows and improve their data analysis capabilities. It provides valuable insights into the performance of marketing campaigns, allowing marketers to optimize their strategies and improve their ROI.

SIPStation provides a SIP trunking solution that allows companies to transition to VoIP, leading to lower telecommunications expenses while maintaining high service quality. This service guarantees cost reductions when moving away from conventional telephony providers and is designed to integrate effortlessly with various PBX systems, including Switchvox, PBXact, FreePBX, among others. Notable functionalities include the ability to port existing phone numbers, facilitating a smooth migration from traditional phone lines without the need to replace current VoIP-compatible PBX systems, as well as SMS capabilities for competitive messaging services. Furthermore, its scalability allows businesses to easily adjust the number of SIP trunks they use based on their specific needs, while a bursting feature permits exceeding the total trunk count temporarily. Additionally, Direct Inward Dialing (DID) offers low-cost phone and toll-free numbers, and an integrated failover mechanism ensures uninterrupted call routing to alternative numbers during service disruptions. This comprehensive approach not only enhances communication efficiency but also empowers businesses to adapt swiftly to changing demands.

The Oracle Enterprise Session Border Controller (E-SBC) delivers top-tier security and scalability for real-time communications across enterprise networks. It integrates easily with both on-premises and cloud-based SIP services, offering robust protections against threats such as DDoS attacks and fraud. With its flexible deployment options and ability to handle complex protocol issues, Oracle's E-SBC ensures high availability, business continuity, and optimized performance across your communication systems. Its comprehensive monitoring tools enable proactive troubleshooting, ensuring that your communications remain reliable and secure.

Microtalk is a leading provider of cloud communication services that help businesses of all sizes simplify their voice and connectivity needs. Its Cloud Phone System supports office, home, and remote work setups, offering IVR, secure call recording, team call queues, and WebRTC integration for seamless online calling. For larger operations, Microtalk’s Contact Centre solutions deliver high-performance VoIP trunks, skills-based routing, toll-free numbers, and a cloud-hosted predictive dialer to boost customer service efficiency. Businesses can also leverage Microtalk’s business-grade fibre internet, which supports speeds up to 10Gb with SD-WAN and MPLS integration for resilience. SMS solutions add further flexibility, allowing companies to launch mass texting campaigns, manage opt-outs automatically, and monitor performance in real time. With over 10,000 customers in 15 countries since 2008, Microtalk has built a reputation for reliable, innovative, and affordable communication technology. Its solutions require no capital investment, offering predictable per-user costs for easy scalability. Backed by 24/7 support and an in-house R&D team, Microtalk ensures continuous improvement and tailored solutions for every client.

Spoof Caller ID is a telecom service that enables users to make voice calls while altering the caller ID displayed on the recipient's device. This service is mainly aimed at businesses, call centers, quality assurance teams, VoIP specialists, and telecom engineers who require reliable verification of routing, display behavior, and network stability across various carriers and geographical areas. The application utilizes SIP-based voice routing and accommodates numerous call routes, allowing users to select the most appropriate call path based on quality, regional needs, or carrier specifications. In addition, it offers features such as balance management, usage monitoring, and an optional test call function, providing valuable tools for both small users and larger enterprises. Ultimately, this service enhances communication options while ensuring that users can tailor their calling experience to their specific requirements.

08UK offers advanced inbound number solutions that boost customer interaction, simplify communication, and broaden business outreach. Our range of services encompasses 0800 freephone numbers to incentivize customer calls, budget-friendly 03 numbers, virtual geographic numbers to establish a local presence, and international toll-free numbers for worldwide accessibility. Companies can implement Customer Satisfaction (CSAT) surveys to gather immediate feedback, which aids in enhancing service quality and agent efficiency. Additionally, our solutions provide increased credibility, tracking for marketing campaigns, sophisticated call management through IVR and call routing systems, and automatic disaster recovery to ensure continuous service. By utilizing these offerings, businesses can maintain strong connections, remain competitive in the market, and prioritize a customer-centric approach. The comprehensive nature of our services makes it easier for organizations to adapt to changing customer needs and preferences effectively.

Toll-free and local numbers are available from more than 160 countries. Forward incoming calls from any phone, PBX, or SIP. Our global network offers multiple layers of redundancy. You can make international outbound calls by using any of your business numbers as a Caller ID. Showing a local caller ID will increase your success rate. To make outbound calls, connect our SIP trunks with your phone system. You can easily set up and modify smart calling routing features such as auto-attendant /IVR, location and time-based routing, and simultaneous ringback tones. Custom greetings, call recording, voicemail, and call recording are all possible. You can easily add or remove virtual numbers, modify calling plans, download call detail records and invoices, modify advanced features, and reach technical support 24/7. Contact your account manager.

VoIP Route's hosted phone service utilizes cutting-edge cloud technology to enhance your business connectivity while eliminating the hefty expenses associated with conventional premise-based systems, all without sacrificing quality. Effortlessly connect and collaborate using phones, smart devices, computers, or laptops. As a premier SIP trunk service provider, VoIP Route caters to both IP-PBX systems and analog/digital telephone adapters or gateways. Our commitment to delivering outstanding service at competitive prices has earned us the trust of both partners and customers alike, all backed by unparalleled support. Residential VoIP leverages advanced technology to ensure optimal performance while significantly reducing costs. By replacing outdated phone lines with digital data, you can utilize the Internet for your calls, making residential VoIP substantially more affordable than traditional phone services. This transition not only cuts down on monthly expenses but also enhances the overall communication experience for users.

OpenSIPS is an open-source SIP proxy/server designed for handling voice, video, instant messaging, presence, and various other SIP extensions. This versatile signaling SIP server caters to carriers, telecom operators, and ITSPs, providing solutions for diverse applications such as Class 4 and Class 5 residential platforms, trunking and wholesale services, enterprise and virtual PBX systems, session border controllers, application servers, load balancers, IMS platforms, call centers, and much more. Its high throughput capability supports tens of thousands of calls per second and millions of simultaneous connections, making it a preferred choice for various SIP scenarios and services. OpenSIPS excels due to its flexible routing and integration options, which allow for custom routing logic through routing scripts and numerous interfacing APIs. With over 120 modules available, it facilitates effective application development for SIP handling, backend operations, and routing logic. Additionally, the 3.2 release of OpenSIPS underwent a thorough security audit, enhancing its safety and reliability for users across the board. This robust framework not only meets current demands but also continuously evolves to address future challenges in the telecommunications landscape.

Zaplee boasts an impressive uptime of over 99%, ensuring that your new cloud phone service remains reliable at all times. Integrate our advanced phone solution into your existing setup swiftly and effortlessly. Our dedicated support team is available around the clock, addressing any concerns almost instantaneously. Trust Zaplee to be compatible with all your devices, both those you currently have and those you may acquire in the future. Effortlessly connect with VoIP, SIP, landline, and mobile phones. You can design personalized greetings for both business hours and after-hours needs. If you prefer not to use your own voice, utilize our text-to-speech feature to generate greetings in various male and female voices, complete with a range of accents. Choose a local number from over 40 countries and thousands of cities, and if you need a toll-free or 800 number, Zaplee has you covered as well. Organize inbound calls according to the specific departments and employee extensions you've established, or based on availability during designated hours. Additionally, enhance your customer experience by offering a dial-by-name directory for easy navigation. With Zaplee, communication has never been more streamlined or user-friendly.

Our carrier-grade Session Border Controller (SBC) and cloud telephony solutions are designed for high performance and scalability, catering to the needs of communication service providers and SMS aggregators with our mobile OTT solution and A2P SMS platform. We also provide secure communication platforms that enhance real-time business collaboration for enterprises. Our comprehensive white-label offerings enable you to deliver cloud telephony services under your own brand identity. With a targeted focus on the IP-based communication sector, we offer a diverse range of products that encompass everything from backbone infrastructure to supporting middleware components. We simplify the process for termination clients by converting their rate plan formats into a native format efficiently. Our system allows for effortless uploading of VoIP rate plans, which are then transformed into the appropriate templates automatically. Additionally, all templates undergo automatic conversion to ensure they align with the native rate plan format seamlessly, and our technology is designed to prevent call drops by ensuring that user account balances do not fall into the negative. This suite of features positions us as a leader in the communication technology space, providing robust solutions tailored to modern business needs.