Alternatives to Sansay VSXi Software SBC

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.

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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.

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 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 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.

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.

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 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 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.

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 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.

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.

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.

Go Trunk offers a SIP trunking service that is both affordable and reliable, providing a comprehensive communications platform for businesses utilizing their own PBX systems. It supplies inbound phone numbers and facilitates the connection of outgoing calls to the public switched telephone network (PSTN). With Go Trunk, companies can access robust communication features, flexibility, dependability, and cost savings. Our blog dedicated to SIP trunking will keep you updated on the latest industry trends and innovations. The articles will serve as an introduction to SIP trunking, highlighting its advantages over traditional telecommunications methods like ISDN. Additionally, the posts will guide you in setting up your SIP trunking and maximizing the benefits of unified communications to gain a competitive edge in your industry. While there is a wealth of information available on VoIP, much of it tends to concentrate primarily on financial aspects. However, it is essential to explore the broader implications and advantages that come with adopting SIP trunking for modern businesses.

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.

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.

In 2013, Flowroute emerged as the pioneering software-centric carrier, introducing the Flowroute HyperNetwork™ to meet the rigorous demands of carrier-grade services while leveraging the flexibility, automation, and scalable nature of cloud technology. The Flowroute SIP Trunking solution allows for seamless integration with existing PBX systems or analog/digital telephone adapters in just a few steps. Customers enjoy the freedom to adjust their call capacity as needed, paying solely for the minutes they consume, which means there are no limits on concurrent calls with Flowroute’s inbound SIP trunking. This eliminates the hassle of capacity planning, as the system automatically generates new instances to accommodate increasing call volumes, ensuring that voice services can expand effortlessly. Additionally, Flowroute optimizes outbound call audio transmission through the most efficient routing path available, enhancing call quality while reducing costs for both local and international calls. Overall, Flowroute's innovative solutions provide unparalleled flexibility and efficiency in telecommunications services.

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.

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.

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.

Experience intuitive call accounting software featuring a user-friendly web interface that supports unlimited extensions and trunks. This solution allows for multi-site call reporting, accommodating various phone system vendors all within a single CALLTRAK application. Users can view real-time phone and line usage data, export detailed call accounting reports in Excel, PDF, Word, or HTML formats, and receive scheduled reports directly via email. The software is designed to be robust, reliable, and entirely user-maintainable, offering a flexible and customizable call accounting experience. It includes a billing solution capable of recovering additional charges such as line rental and voicemail services. With competitive pricing and one-click access to daily activity reports, users can tailor report templates to suit their specific needs. Additionally, the software provides telephone network analysis and trunk diagnostics to optimize line usage and identify any faulty lines, while also displaying average ring times and the number of calls answered, ensuring comprehensive insights into communication efficiency. Overall, this call accounting software enhances operational transparency and helps businesses manage their telecommunication expenses effectively.

We provide a sophisticated SIP trunking solution designed to enhance efficiency and deliver substantial cost savings. Comcast Business SIP Trunking operates on the nation’s largest VoIP network, ensuring reliable communication. This service comes with dedicated bandwidth, featuring a robust 100 Mbps connection to facilitate seamless voice and data transmission. The versatility of Comcast SIP enables various configurations tailored to your business needs, supporting 6 CCS and accommodating up to 800. You have the flexibility to set up trunks in a manner most suitable for your organization, whether it’s for inbound, outbound, two-way, single, or multiple trunk groups. Additionally, you can implement continuity features, such as overflow and failover, to guarantee that no calls are missed, regardless of high call volume or service interruptions. Furthermore, the service is bolstered by Comcast Business's Quality of Service (QoS) and round-the-clock expert support. In case of emergencies like natural disasters or power outages, you can easily reroute calls to a preselected number. Moreover, you can link caller information to display the name and number of incoming callers, enhancing your communication experience and efficiency further.

Crystal Quality is a comprehensive web-based multimedia recording platform designed for TDM, analog, and VoIP environments, accommodating a variety of telephony interfaces such as analog trunks/extensions, digital trunks (ISDN BRI/PRI, E1/T1), digital extensions, VoIP trunks/extensions, alongside discrete and trunked radios. Avdor CIS's Crystal Quality facilitates the management of diverse recording types, including those for call centers, contact centers, telephony, mobile devices, screen captures, SMS, and Skype for Business, all through an intuitive and aesthetically pleasing user interface. The CIS platform provides multiple flexible pricing options, making it simple to select the most appropriate plan tailored to your business requirements while maximizing benefits and cost efficiency. This innovative solution from Avdor CIS consolidates call recording and monitoring into a singular, exceptional offering that meets all your organizational needs seamlessly. Ultimately, Crystal Quality enhances operational efficiency by integrating various communication forms into one effective system.

Zentrunk offers a cloud-based SIP trunking service that ensures global connectivity and supports unlimited simultaneous calls with a flexible pay-as-you-go pricing model. You can link your VoIP setup within minutes, as we have carefully developed and fine-tuned our network infrastructure to eliminate the burden from your end. Our service guarantees low Post Dial Delay by routing your calls through a single hop to local carriers, while also providing essential features like Caller Line Identification (CLI) and Dual-Tone Multi-Frequency (DTMF) signaling, all without the inconvenience of out-of-region audio looping. With a resilient infrastructure that spans multiple regions and incorporates at least three local carrier connections in various countries, Zentrunk assures an impressive uptime of 99.95%. This reliability makes it an ideal solution for businesses looking to enhance their communication capabilities.

Juris Astra offers a comprehensive call centre solution designed for both inbound and outbound communications, featuring essential components such as a telephony automatic call distributor (ACD), interactive voice response (IVR), and a dialler system. Additionally, it provides optional functionalities for customer relationship management (CRM), sales force and marketing oversight, as well as case management to effectively manage sales leads, marketing efforts, customer complaints, service requests, and overall customer service challenges. This all-in-one solution enables the seamless handling of various communication channels, including telephony, email, web chat, fax, and SMS interactions, all in an automated fashion. The PABX system serves as the backbone for managing telephony connections, accommodating incoming lines from telecommunications providers like ISDN PRI or CO Analog PSTN trunks, alongside extension lines leading to agents or users. Among its standard features, Juris Astra includes caller privileges, Class of Service (CoS) for outgoing calls, support for hardphones and softphones, VoIP capabilities, and SIP integration. With its diverse offerings, Juris Astra ultimately positions itself as an essential tool for enhancing operational efficiency in customer engagement.

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.

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 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.

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.

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

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 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.

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.

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.

Switchvox is tailored for businesses similar to yours, offering enterprise-level features, a comprehensive pricing structure, and a design that has garnered awards, making it more than just another VoIP system for business phones. It provides a smart approach for your organization to leverage Unified Communications (UC) and collaboration tools at a reasonable cost. With functionalities such as chat, video conferencing, quick dialing, status updates, presence indicators, and customizable call rules, your team and clients can communicate more effectively. Switchvox stands out as a highly adaptable and scalable UC platform. It features smooth integration with IP phones, SIP trunks, and PSTN connections, as well as various collaboration and productivity capabilities, allowing you to choose from multiple deployment options. In educational environments, effective communication is crucial, and Switchvox delivers an accessible, user-friendly solution that goes beyond a standard phone system that merely handles calls. This platform ensures that schools can maintain seamless interactions and foster an engaging learning atmosphere.

CommsOffice Express serves as both a single and multi-site call management solution, continuously assessing how effectively your company or organization engages with current and prospective clients. This all-encompassing business call management tool integrates seamlessly with your telephone system and delivers immediate, real-time graphical insights regarding incoming and outgoing calls within your enterprise. The Today screen offers a swift graphical overview of system activity at any moment during the day or for any chosen historical date range. Users can customize this display from a variety of available charts and apply filters to showcase only the specific information they need. With 140 standard reports, CommsOffice Express equips users with extensive data, enabling reporting based on site, department, extension, trunk, DID/DDI, account code, or even tracking how frequently a particular phone number has been dialed or called. This flexibility ensures that organizations can tailor their reporting and analytics to suit their unique operational requirements.

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.

Leverage Verizon’s Voice-over-IP (VoIP) network through IP Trunking to provide your organization with affordable and adaptable communication solutions. Affordable voice services for businesses can distinguish between merely being advantageous and gaining a competitive advantage in the market. As companies navigate an increasingly interconnected landscape, IP Trunking can enhance team productivity, improve connections with clients and partners, and optimize expenses simultaneously. Organizations aiming to manage and streamline their IT expenditures will find significant benefits. This solution is especially beneficial for enterprises that handle a high volume of VoIP calls concurrently, including those involved in disaster recovery, seasonal retail, and call centers. Additionally, government entities looking to upgrade their infrastructures can fully capitalize on their unified WANs. IP Trunking facilitates voice communication over digital networks rather than relying on conventional telephone lines. This technology is designed to accommodate a large number of simultaneous connections, allowing organizations and agencies to better manage their costs while enhancing communication efficiency. Ultimately, embracing IP Trunking can serve as a strategic move towards modernizing communication frameworks across various sectors.

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.

SIP.US is an innovative cloud SIP trunking service tailored to enhance business communication through quick setup, adaptable management, and an array of powerful features. The intuitive self-service control panel empowers users to effortlessly oversee SIP trunks, acquire and manage phone numbers, activate or deactivate services, analyze and export real-time call data records (CDRs), and adjust billing settings, all within a matter of minutes. Additionally, SIP.US facilitates business SMS, offering a secure and immediate messaging platform for interactions with customers and clients. With extensive Direct Inward Dialing (DID) capabilities, businesses can easily search for and order both local and international numbers, most of which can be set up immediately. The service also supports Bring-Your-Own-Bandwidth, granting users the flexibility to select their internet provider and bandwidth, while being compatible with both G.711 and G.729 codecs. Furthermore, SIP.US ensures that businesses can scale their communication solutions efficiently, adapting to their growing needs without hassle.

We offer a top-notch wholesale voice and SMS service that includes SIP trunking, business SMS solutions, local DID numbers, and toll-free 800 numbers, tailored for the modern digital landscape. Our direct connections with leading Tier 1 voice carriers across Canada, the US, and globally ensure exceptional voice clarity. With high-quality A2P messaging capabilities to any global destination through reliable telecom partners, we provide a risk-free trial and a no-obligation consultation. With nearly twenty years of experience assisting over 300 businesses and resellers, we have established a strong reputation as a trusted "carrier's carrier." Recognizing that your communication infrastructure is vital to your operations, we pride ourselves on being the only service provider focused on delivering pristine TDM quality voice transport to and from the PSTN. Our business-oriented approach, combined with advantageous peering agreements and volume discounts, guarantees that our clients enjoy substantial cost savings from the very beginning of our partnership. In this rapidly changing industry, we remain committed to adapting and evolving to meet our clients' needs effectively.

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

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.