Alternatives to Sangoma Enterprise SBC

Ultatel is a recognized industry leader in business communications. With the power of cloud VoIP technology, we help businesses to be more productive and connect with their customers wherever they are. Our plans come fully customizable and scalable with unlimited Calling, SMS, Fax, Chat, Video & 40+ Advanced Features. The best part is, there are no hidden fees or charges! Thanks to our Transparent Pricing model, what you see is what you pay (unlike some other vendors). A Gartner Category Leader and G2 High Performer, Ultatel aims to provide a unified communications platform that adapts to your companies' needs. Powered by our FlexScale technology, you can scale up or down seamlessly and instantly with no downtime or penalty. Furthermore, our Award-Winning Customer Support department is there for you 24/7/365. With a 94% first-contact resolution, you'll have the peace of mind of being connected beyond limits. Contact us today and schedule your discovery call or demo.

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.

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.

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 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.

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.

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.

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.

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.

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 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.

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.

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.

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 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.

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.

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.

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 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.

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.

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.

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.

The PSX Software Edition (PSX SWe) from Ribbon is a centralized and virtualized policy and routing solution tailored for service providers to effectively manage and enhance VoIP services in a variety of network settings. It offers advanced session control for session border controllers (SBCs), media gateways, and call controllers, accommodating a wide range of protocols, including SIP, ENUM, H.323, TDM, SS7/C7, IN/AIN, and IMS. Thanks to its distributed architecture, service providers can set up local instances that improve performance and resilience while still upholding centralized control over policy, routing, and administration, thus removing the complexities of synchronization across multiple locations. Prominent applications feature Least Cost Routing (LCR), Quality of Service (QoS) routing, toll-free number routing, and number translation services, all of which enhance operational efficiency. Moreover, the PSX SWe serves as an IMS Breakout Gateway Control Function (BGCF) and incorporates call validation treatment for STIR/SHAKEN along with Reputation Scoring services, effectively reducing the risk of call spoofing. This multi-faceted approach ensures that VoIP services remain reliable and secure, ultimately benefiting both providers and end-users.

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.

The Secunet SBC, crafted in collaboration with Secunet, is engineered to meet the stringent security requirements of institutions that demand top-tier protection. This innovative solution merges FRAFOS technology with the robust secunetWall firewall, creating a secure environment that ensures comprehensive protection and filtering of data streams across various levels, including network, transport, voice, and application layers. With its certification from the German Federal Office for Information Security (BSI) under ID BSI-DSZ-CC-1089, it achieves the coveted CC EAL 4+ rating, signifying its resilience against significant cyber threats. Furthermore, the Secunet SBC boasts exceptional integration capabilities, as acknowledged by telephone system manufacturer Mitel, which has granted it the Mitel compatible seal while recognizing Secunet as a trusted Mitel Developer Partner. Among the standout features are secure VoIP communication between internal and external networks, adaptable audio and video gateway functionalities, user-friendly GUI-driven administration, remote configuration and updates, and enhanced system reliability through failover and georedundancy across multiple locations. Additionally, this product not only safeguards communications but also positions organizations for future technological advancements.

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.

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.

Protect your communications by ensuring your VoIP calls are encrypted and safe from eavesdropping. If you value your privacy during conversations, you’ll appreciate our complimentary encryption services for all types of communications, including text, voice, and video, utilizing TLS/SRTP and ZRTP protocols. Our VoIP softphone seamlessly integrates with your contacts, providing a consolidated list for quick access from various sources such as Outlook, Windows, Mac, LDAP, XMPP, XCAP, as well as mobile platforms like Android and iOS. You can also easily identify incoming calls before answering, enhancing your communication experience. Zoiper is compatible with a wide array of operating systems, whether you are on macOS, Linux, or Windows, and also supports iOS, Android, and web browsers. We strive to continuously improve our offerings and expand our compatibility. Forget the complex jargon; our solution encompasses Audio, Video, Fax, Presence, and Instant Messaging, all accessible through a user-friendly interface designed for your convenience. With Zoiper, you can communicate confidently and efficiently across multiple platforms.

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.

OfficeConnect™, provides unified voice, conferencing and messaging as well as faxing. Our VoIP phones are more flexible and cheaper than legacy on-premises systems. You can keep your projects moving and synchronize teams with best-in class audio and video conference made easy. UniVoIP encrypts all voice traffic, both from remote and on-site telephones. It uses SRTP with 128 bit AES encryption. SRTP conforms to both the federal standards FIPS 140-2 Certified encryption and the Department of Defense's PKI Digital Certificate.

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.

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.

A hosted phone system, also known as PBX (Private Branch Exchange), enables businesses of all sizes to utilize an advanced enterprise-level telephone system without the significant expense associated with physical telephone infrastructure. This system is fully managed and operated by GlobalPhone, your provider of Voice-over-IP services. Operating in the cloud grants you the flexibility to work from various locations, including the office, home, or on a mobile device, while still having complete access to the office phone system. You gain access to features typically reserved for Fortune 500 companies, such as auto-attendants, call centers, hunt groups, conferencing options, call recording, and voicemail services. Unlike traditional phone networks that rely on copper wires to connect to an in-house PBX, SIP trunking eliminates the need for these outdated copper connections by employing Voice over Internet Protocol (VoIP) technology to establish a virtual phone line or trunk, facilitating the transmission of voice traffic to your phone system. This modern approach not only reduces costs but also enhances the overall functionality and reliability of your business communications.

The Total Access 900e Series is engineered to support Ethernet or multi-T1 converged voice and data solutions, providing powerful IP routing, firewall, and VPN capabilities. It includes four T1 interfaces, one Gigabit interface, and two Fast Ethernet interfaces for both LAN and WAN applications, along with various models equipped with user interfaces boasting up to 24 FXS ports for delivering analog voice services. Additionally, these devices can incorporate SBC functionality, allowing for up to 100 simultaneous calls. For example, the Total Access 908e model provides 8 FXS ports through an Amphenol connection, along with four T1/FT1 ports and three Ethernet interfaces (one Gigabit and two Fast) for network connectivity. The part number for this model is 4243908F1. Similarly, the Total Access 916e model features 16 FXS ports via an Amphenol connection, four T1/FT1 ports, and three Ethernet interfaces (one Gigabit and two Fast) for both WAN and LAN access, identified by part number 4243916F1. In summary, the Total Access 900e Series presents versatile options for organizations seeking reliable voice and data convergence solutions.

The Total Access 900 Series is engineered to support both Ethernet and multi-T1 applications, featuring four T1 interfaces, a Gigabit interface, and two Fast Ethernet interfaces for connections to either LAN or WAN environments, along with multiple platforms that provide varying user interfaces, accommodating up to 24 FXS ports for analog voice communication. Additionally, these devices can incorporate SBC functionality, facilitating up to 100 simultaneous calls. Specifically, the Total Access 908 model is equipped with a single T1 network interface and 8 FXS user interfaces, providing analog voice delivery, along with an integrated DSX-1 port for PBX connections (T1 CAS or PRI), and a Gigabit Ethernet interface for WAN or LAN connectivity, identified by part number 4213908F1. Meanwhile, the Total Access 916 model boasts a similar single T1 network interface, but offers 16 FXS user interfaces for analog voice delivery, also including an integrated DSX-1 port for PBX connectivity (T1 CAS or PRI) and a Gigabit Ethernet interface for WAN or LAN, with part number 4213916F1. Furthermore, the series concludes with the Total Access 908, which provides a single T1 network interface along with 24 FXS user interfaces designed specifically for analog voice delivery, ensuring robust communication solutions.

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.

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.

DID Logic is a global provider of VoIP services that supplies local, geographical, toll-free, and mobile virtual numbers across 139 nations. Our infrastructure consists of 12 Points of Presence (PoPs) around the globe, enabling us to offer your users a seamless local calling experience through either an enterprise-level SIP trunk connection or PSTN forwarding. With our local SIP trunk service, you gain direct and low-latency access to regional voice carriers. You can quickly deploy our services and connect to your IP PBX without delay. By leveraging local VoIP gateways in 12 countries, you can effectively reduce latency in your communications. Each PoP is intricately linked to various voice carriers, ensuring reliability and efficiency. We maintain our own network and utilize hardware-based switches, which enhances our operational capabilities. Furthermore, we possess our own Autonomous System and procure IP transit directly from Tier 1 carriers to ensure high service quality. The hardware utilized in each PoP is owned by DID Logic, and our SIP termination is routed to local carriers. RTP streams are transmitted over the lowest latency domestic IP links, avoiding the need to loop through aggregation sites. Additionally, DID Logic is affiliated with multiple Internet Exchanges (IXes) globally, granting us access to hundreds of networks through meet-me rooms and cross-connects, which further strengthens our service offerings. Our commitment to providing high-quality VoIP solutions positions us as a leader in the telecommunications industry.

RingLeader is revolutionizing communication for businesses and teams by introducing innovative technology known as "Crowdvoicing," enabling small enterprises and teams to effortlessly implement cloud-based phone services for durations as short as one day. Our SIP Trunking solution seamlessly integrates with any PBX hardware, making it ideal for larger organizations; in fact, many of our SIP clients can utilize our trunks without the need for extra hardware. As one of the rare NATIVE SIP providers globally, we believe that the best solutions are often the most straightforward ones. RingLeader empowers your business to communicate more intelligently, quickly, and efficiently—whether from a traditional office or a home setting—offering adaptable solutions tailored to the needs of small, medium, or large teams. Our services are backed by dedicated support from real people who answer calls personally. Ultimately, RingLeader provides the quickest and easiest method for organizations to enhance their communication strategies, fostering better connections with both colleagues and customers, regardless of their location. By prioritizing user experience and accessibility, we ensure that businesses can thrive in their communication efforts.

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.

Azure Confidential Computing enhances the privacy and security of data by safeguarding it during processing, rather than merely when it is stored or transmitted. It achieves this by encrypting data in memory through hardware-based trusted execution environments, enabling computations to occur only after the cloud platform has authenticated the environment. This method effectively blocks access from cloud service providers, administrators, and other privileged users. Additionally, it facilitates scenarios like multi-party analytics, where various organizations can collaboratively use encrypted datasets for joint machine learning efforts without disclosing their respective data. Users maintain complete control over their data and code, dictating which hardware and software can access them, and they can transition existing workloads using familiar tools, SDKs, and cloud infrastructures. Ultimately, this approach not only fosters collaboration but also significantly bolsters trust in cloud computing environments.

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 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.

Upgrading your on-premise business phone system through SIP Trunks not only enhances flexibility but also reduces telephony expenses. You can either modernize your current phone infrastructure or save enough to invest in a brand-new system. Most contemporary phone systems seamlessly integrate with SIP Trunking without any need for additional hardware. Our Partners collaborate closely to facilitate all aspects of setup and migration. Reduce your costs with lower rates for long-distance, toll-free, and international calls. By utilizing Intermedia's robust enterprise-grade data centers, you can safeguard your operations against local outages. Traditionally, customers incur costs based on a per-line charge for physical phone lines originating from their central office, leading to significant expenditures. The initial investment is substantial, and the costs associated with adding more lines for growth can be prohibitive. However, SIP Trunking allows phone calls to be transmitted over the internet, fundamentally changing the cost structure and flexibility of business communications. This shift not only streamlines operations but also positions businesses for future scalability and efficiency.

Google Cloud's Confidential Computing offers hardware-based Trusted Execution Environments (TEEs) that encrypt data while it is actively being used, thus completing the encryption process for data both at rest and in transit. This suite includes Confidential VMs, which utilize AMD SEV, SEV-SNP, Intel TDX, and NVIDIA confidential GPUs, alongside Confidential Space facilitating secure multi-party data sharing, Google Cloud Attestation, and split-trust encryption tools. Confidential VMs are designed to support workloads within Compute Engine and are applicable across various services such as Dataproc, Dataflow, GKE, and Gemini Enterprise Agent Platform Notebooks. The underlying architecture guarantees that memory is encrypted during runtime, isolates workloads from the host operating system and hypervisor, and includes attestation features that provide customers with proof of operation within a secure enclave. Use cases are diverse, spanning confidential analytics, federated learning in sectors like healthcare and finance, generative AI model deployment, and collaborative data sharing in supply chains. Ultimately, this innovative approach minimizes the trust boundary to only the guest application rather than the entire computing environment, enhancing overall security and privacy for sensitive workloads.