VOIP IX is a commercial Internet Exchange.

Our mission is to overcome challenges which are experienced on a daily basis by the VOIP industry such as: latency, quality and security.

Our offer differs from other Internet Exchange as we apply top priority to the VOIP traffic of our members.

VOIP IX offers:

• Dedicated Network: Exclusively optimized for VoIP traffic transport
• Ultra-Low Latency: Always at the forefront of our priorities
• Multi-location Connectivity: Broaden your reach seamlessly
• Stringent Security Protocols: Ensuring every communication is secure
• Advanced Telemetry: Track latency, jitter, MoS, and more to any endpoint

Yes they do!

More and more companies host their PBX on Cloud Service Providers' infrastructure.

By joining VOIP IX Cloud Service Providers can therefore offer an unmatched experience to their customers running a VOIP infrastructure.

That's a direct connection from the cloud to VOIP operators and providers.

Our presence is listed here

The list of our members can be found here.

Our peering policy can be found here.

You can send your request here.

VoIP, or Voice over Internet Protocol, is a technology that allows voice communications and multimedia sessions to be delivered over the internet or other IP-based networks. Instead of using traditional analog phone lines, VoIP converts voice data into digital packets and transmits them over the internet.

Here are some key points about VoIP:

1. Cost-Effective: VoIP services often offer cheaper long-distance and international calling rates compared to traditional phone services. Additionally, many VoIP services offer flat-rate plans allowing unlimited calls to certain areas, further reducing costs for businesses and individuals.
2. Flexibility: VoIP users can make and receive calls using a variety of devices, including dedicated VoIP phones, computers with headsets, or even mobile devices with VoIP apps.
3. Features: In addition to basic calling features, many VoIP services offer a suite of advanced functionalities, such as voicemail-to-email, video conferencing, call forwarding, and more.
4. Quality: While early VoIP services were sometimes criticized for subpar call quality, advances in technology have greatly improved VoIP call clarity. Nowadays, with a good internet connection, VoIP calls can be as clear as, or even clearer than, traditional phone calls.
5. Portability: With VoIP, users can move their VoIP device or phone to a different broadband network connection, and their phone number and configurations typically move with them. This is especially beneficial for businesses or individuals who travel or relocate frequently.
6. Reliance on Internet: One downside is that VoIP requires a stable and high-speed internet connection. If the internet goes down or if there's significant latency or packet loss, it can affect call quality or availability.

Popular services like Skype, Zoom, and WhatsApp use VoIP technology to enable voice and video calls over the internet.

Overall, the adoption of VoIP has been growing rapidly due to its cost savings, flexibility, and array of features. Many businesses and consumers have either supplemented or entirely replaced their traditional phone lines with VoIP.

A VoIP phone is a hardware or software-based telephone designed to use voice over IP (VoIP) technology to send and receive phone calls over an IP network, such as the internet, rather than the traditional public switched telephone network (PSTN).

There are two main types of VoIP phones:

1. Hardware-based VoIP Phones: These look like traditional telephones but are equipped with specialized hardware to support IP technology. They connect directly to your router via Ethernet or wirelessly and are configured specifically for VoIP communication.
2. Software-based VoIP Phones: Often referred to as "softphones", these are software applications installed on a computing device, such as a PC, laptop, tablet, or smartphone. Softphones use the device's internet connection and built-in audio hardware (or external microphone and speakers) to place and receive VoIP calls. Examples include Skype, Zoom, and the voice-calling feature of WhatsApp.

Key features of VoIP phones include:

• Codec Support: VoIP phones use codecs to convert voice into data packets and vice versa. Some common codecs include G.711, G.722, and G.729.
• Call Features: Many VoIP phones come with advanced call handling features, such as call transfer, call hold, call conferencing, voicemail, caller ID, and more.
• Network Configuration: These phones often have capabilities to connect to different types of networks, be it via Ethernet, Wi-Fi, or even cellular networks when using softphones on mobile devices.
• HD Voice: Many modern VoIP phones offer high-definition (HD) voice capabilities, providing clearer audio quality compared to traditional phone calls.
• Integrated Directories: VoIP phones can integrate with directory services, allowing users to access contact lists or directories to quickly place calls.
• Power over Ethernet (PoE): Some hardware-based VoIP phones can receive power and data through the same Ethernet cable, eliminating the need for a separate power adapter.
• Security Features: Given that VoIP calls traverse the internet, VoIP phones often incorporate security features such as encryption and secure boot to protect calls from eavesdropping and tampering.

VoIP phones, both hardware and software varieties, have gained popularity in businesses and homes because of the cost savings, flexibility, and enhanced features they offer when compared to traditional landline phones.

SIP, or Session Initiation Protocol, is a signaling protocol used for initiating, maintaining, modifying, and terminating real-time sessions that involve video, voice, messaging, and other communications applications and services between two or more endpoints on IP networks.

Here are some key points to understand about SIP:

1. Initiating and Managing Sessions: SIP can be used for various types of communications, such as voice and video calls, instant messaging, and multimedia conferences. It's commonly associated with and utilized in VoIP (Voice over IP) systems but is not limited to just voice.
2. Components:
   • User Agent (UA): The endpoint in a SIP communication. This could be a physical device like a VoIP phone or a software-based application like a softphone. User agents can function as clients (UAC) initiating SIP requests or servers (UAS) responding to requests.
   • SIP Server: These manage and facilitate SIP communication. There are several types, including:
     • Registrar: Records the locations of User Agents.
     • Proxy Server: Routes SIP requests to the user's current location.
     • Redirect Server: Informs the sender of the user's new address when they've moved.
   • Session Border Controllers (SBCs): These are devices that control the initiation and termination of calls, manage call routing, handle signaling, and provide security functions.
3. URI Addressing: Much like how emails use email addresses, SIP uses SIP URIs (Uniform Resource Identifiers) to address users. A SIP URI looks somewhat like an email address, for example, sip:username@domain.com.
4. SIP Messages: SIP employs various methods or messages, such as INVITE (to initiate a call), BYE (to end a call), ACK (to acknowledge call setups), and REGISTER (to register a user agent with a SIP server), among others.
5. Interoperability: SIP was designed to be modular, meaning it can work with other protocols to provide a complete multimedia experience. For example, while SIP handles session setup and termination, the actual data transport (like voice or video streams) is often managed by other protocols, such as RTP (Real-time Transport Protocol).
6. Security: SIP can be secured using techniques like SIP over TLS (Transport Layer Security) for encryption and SIP authentication for verifying the identity of endpoints.
7. Standardization: SIP is a standard protocol developed by the IETF (Internet Engineering Task Force) and is defined in RFC 3261.

In the realm of telecommunications, SIP has become one of the main protocols used for VoIP and other real-time, multimedia communications because of its flexibility, scalability, and wide industry support.

An IP PBX (Internet Protocol Private Branch Exchange) is a phone system that uses IP data networks to manage call switching, route calls, and handle other messaging. In essence, an IP PBX is a PBX system that uses the internet or other IP networks to handle voice (and often video and other media) communications.

Here's a deeper look into IP PBX:

1. Traditional PBX vs. IP PBX: Traditional PBX systems manage calls between landline telephones and are often more hardware-centric. IP PBX systems, on the other hand, manage calls between IP-based networks and can handle both traditional landline calls and VoIP (Voice over IP) calls.
2. Components: An IP PBX consists of one or more SIP (Session Initiation Protocol) phones, an IP PBX server, and optionally a VoIP gateway to connect to existing PSTN lines.
3. Functionality: The IP PBX server functions similarly to a proxy server. SIP clients, being either softphones or hardware-based phones, register with the IP PBX server, and when they wish to make a call, the IP PBX connects the call. The IP PBX has a directory of all phones/users and their corresponding SIP addresses, which allows it to connect internal calls or route external calls through a VoIP service provider or a traditional phone line.
4. Features: Modern IP PBX systems offer a variety of features, including:
   • Call routing and transferring
   • Voicemail
   • Call queues and auto-attendants
   • Time-based call routing
   • Conference calling
   • Call forwarding, hold, and mute
   • Integrated messaging (like SMS or chat)
   • Call recording
   • Detailed call logs and analytics
5. Flexibility & Scalability: One of the biggest advantages of an IP PBX is the ease of installation and scaling. As it's largely software-based, adding new lines or extensions often doesn't require much physical installation. Additionally, because it operates over the internet, users can connect from anywhere, making it especially useful for businesses with remote or distributed workforces.
6. Cost-Effective: IP PBX systems often provide cost savings, especially for organizations that make a lot of long-distance or international calls. By routing calls over the internet through VoIP providers, companies can often get lower rates.
7. Unified Communications: Many IP PBX systems are part of broader unified communications solutions, integrating voice, video, messaging, and even file sharing into a single platform.

It's worth noting that, as with all internet-based systems, IP PBX systems require a stable and high-speed internet connection to maintain call quality. Proper setup, including measures like Quality of Service (QoS) configurations, can help ensure consistent performance.

SIP Trunking is a technology that uses the Session Initiation Protocol (SIP) to provide connectivity between a private phone system (like an IP PBX) and the public switched telephone network (PSTN) via the internet or another IP-based network. In simpler terms, it allows phone systems to make and receive calls using the internet.

Here's a more detailed look into SIP Trunking:

1. Replacing Traditional Lines: SIP Trunking can replace traditional analog phone lines, Primary Rate Interfaces (PRI), and other older technologies. Instead of using physical lines, SIP Trunks are virtual phone lines that exist over the internet.
2. Scalability: One of the major benefits of SIP Trunking is its scalability. Businesses can easily add or remove lines based on their needs, often without the need for physical installation or waiting for a provider to add or remove physical lines.
3. Cost Savings: Using SIP Trunks often results in cost savings for businesses. The costs for long-distance and international calls are typically much lower than with traditional phone services. Additionally, the need for physical infrastructure is reduced, further saving on costs.
4. Unified Communications: SIP Trunking can be part of a unified communications strategy, where voice, video, data, and other communication methods are integrated. This can streamline communications and improve efficiency.
5. Reliability: Since SIP Trunking uses the internet, it's essential to have a stable and high-speed connection. However, with proper setup and redundancies (like failover SIP Trunks), businesses can achieve high reliability and uptime.
6. Features: SIP Trunking can come with various features, including Direct Inward Dialing (DID), which allows callers to reach a user directly, caller ID, encryption for security, and more.
7. Requirements: To use SIP Trunking, a business usually needs:
   • An IP PBX that supports SIP.
   • A SIP Trunking provider to offer the service.
   • A high-speed and stable internet connection.
   • Appropriate security measures, like firewalls and Session Border Controllers (SBCs), to ensure the safety and quality of calls.

In summary, SIP Trunking provides a modern, scalable, and cost-effective solution for businesses to manage their telephony needs, bridging the gap between traditional phone systems and the digital world of the internet.

A VoIP interconnection refers to the process and infrastructure through which different VoIP (Voice over Internet Protocol) networks or services connect and exchange traffic. This allows users on one VoIP service or network to call users on another, seamlessly, as if they were on the same network. The concept is analogous to how different internet service providers interconnect to allow users from one ISP to access websites or services hosted on another ISP.

Here are some points related to VoIP interconnection:

1. Purpose: VoIP interconnection enables end-to-end communication between users of different VoIP service providers. This is essential for the global reach of VoIP services and ensures that a user on one VoIP service can call a user on any other network, including traditional telephony networks.
2. Interconnection Points: These are the physical or virtual points where two VoIP networks connect. They involve shared equipment, IP addresses, and ports to facilitate the transfer of VoIP traffic.
3. Protocols: The Session Initiation Protocol (SIP) is the most common protocol used for VoIP interconnections. However, the actual voice data is typically carried by the Real-time Transport Protocol (RTP).
4. Settlements and Agreements: Just like traditional telecommunication providers have agreements about rates and traffic handling, VoIP service providers often have peering agreements or settlement agreements to define the terms of their interconnection. This can include technical specifications, financial settlements, and traffic handling policies.
5. Challenges: Interconnection can come with challenges such as:
   • Interoperability: Ensuring that different VoIP systems can work together without issues.
   • Quality of Service: Ensuring that voice traffic is prioritized and that the call quality remains high.
   • Security: Protecting the interconnected networks from potential threats like denial-of-service attacks, fraud, or eavesdropping.
   • Regulation: In some regions, VoIP interconnection might be subject to regulatory requirements, especially when connecting with traditional telephony networks.
6. VoIP Exchanges or IX: Some centralized platforms or hubs facilitate VoIP interconnection between multiple providers. These can simplify the process, especially for smaller VoIP providers, by offering a single point of interconnection to multiple other networks.
7. Gateway to PSTN: One critical form of VoIP interconnection is the connection between VoIP networks and the traditional Public Switched Telephone Network (PSTN). This allows VoIP users to call regular phone numbers and vice versa. This connection is typically facilitated through gateways that convert between the digital signals of VoIP and the analog signals of the PSTN.

In essence, VoIP interconnection is fundamental for the broad adoption and success of VoIP technology, ensuring that it's not just an isolated network but a fully integrated part of the global telecommunication ecosystem.

DID stands for Direct Inward Dialing. It's a feature offered by telephone companies for use with their customers' PBX (Private Branch Exchange) system. Essentially, DID allows outside callers to dial directly to a specific extension on a PBX without the need for a human operator or receptionist.

Here are some key points to understand about DID:

1. Virtual Numbers: In the context of DID, the numbers provided by the telephone companies are often called "virtual numbers" because they don't have a dedicated physical phone line. Instead, they are mapped to specific extensions or devices within the PBX.
2. Cost-Efficiency: Before the introduction of DID, businesses needed individual phone lines for each employee or department that wanted a direct number. With DID, a company can have fewer physical phone lines than direct numbers, saving costs while still offering direct access to employees.
3. Integration with VoIP: In modern telephony, DID numbers often get integrated with VoIP (Voice over IP) services. This means that even if a call originates from the traditional PSTN (Public Switched Telephone Network), it can be directed to a VoIP endpoint through a DID number.
4. Global Reach: Many VoIP service providers offer international DID numbers. This means a business can have a local phone number in another country, allowing customers from that country to call them at local rates, even if the business's actual location is elsewhere.
5. Use Cases: DIDs are commonly used in call centers, where individual agents can have direct numbers. They are also popular with businesses that want direct numbers for various departments or high-profile employees.
6. Number Portability: Many regions and countries allow for DID number portability, meaning businesses can keep their DID numbers even when switching service providers.

In essence, DID provides a mechanism for businesses to offer a more personalized and direct connection for their clients and partners without the overhead of numerous dedicated lines. It streamlines the call-routing process and can enhance the professional image of a business.

SS7, which stands for Signaling System No. 7, is a telecommunications protocol suite that defines how network elements in a public switched telephone network (PSTN) exchange information over a digital signaling network. This information allows the setup and tear-down of phone calls, routing of calls, and various other services.

Here's a more detailed overview of SS7:

1. Role in Telecommunications: SS7 is primarily used in traditional voice telephony to establish and control call connections. It also supports various non-voice services.
2. Components: The SS7 network is composed of signaling points, which can be:
   • Service Switching Points (SSP): They are switches that originate, terminate, or switch calls.
   • Signal Transfer Points (STP): These are packet switches that route received SS7 messages to their proper destination.
   • Service Control Points (SCP): They are databases that provide information necessary for advanced call-processing features, like 800 number translation or mobile number location.
3. Services Supported: Beyond basic call setup, management, and termination, SS7 also supports:
   • Number Translation: Useful for services like toll-free (e.g., 800) numbers.
   • Local Number Portability: Allows customers to keep their number when switching service providers.
   • Short Message Service (SMS): In mobile networks.
   • Prepaid Billing Systems.
   • Roaming Abilities for Mobile Networks: SS7 helps determine the location of a mobile user and deliver calls/messages to them.
4. Security Concerns: The SS7 protocol, being several decades old, was designed before security threats over telecommunication networks were a significant concern. This has led to vulnerabilities that can potentially be exploited by attackers to intercept calls and messages or get the location of a mobile device. Over the years, there have been calls to improve the security features of SS7 or transition to more secure protocols.
5. Relation to Modern Protocols: In the context of newer telecommunication and VoIP (Voice over IP) systems, SS7 is often interfaced with newer protocols like SIP (Session Initiation Protocol). Gateways can translate between SS7 for the PSTN and SIP for IP-based networks.
6. Decline and Successors: With the rise of IP-based networks and services, the significance of SS7 is gradually declining. Protocols like Diameter (often used in mobile networks) and SIP are becoming more prevalent.

In summary, SS7 has been foundational in the telecommunication industry for call and message handling in traditional phone networks, but as technology progresses, newer protocols are starting to take its place.

A cloud-based IP PBX, often referred to as a hosted PBX or virtual PBX, is a modern approach to business telephony wherein the entire phone system is hosted in the cloud, as opposed to having physical hardware on-premises. It leverages IP (Internet Protocol) technology to manage call routing, features, and functionalities over the internet.

Here's a breakdown of cloud-based IP PBX:

1. No On-Premises Equipment: Unlike traditional PBX systems that require physical equipment (like servers and switchboards) to be installed at the business location, a cloud-based IP PBX system hosts all the necessary infrastructure in data centers, managed by the service provider.
2. Accessibility: Since the system is cloud-based, users can access their business phone system from anywhere with an internet connection. This makes it highly beneficial for businesses with remote or distributed teams.
3. Scalability: One of the standout features of a cloud-based system is scalability. Businesses can easily add or remove users or features based on their needs without worrying about capacity constraints or physical hardware.
4. Cost-Efficiency: With no need for on-site equipment and its associated maintenance, businesses often find cloud-based solutions more cost-effective. They typically operate on a subscription model, which includes updates, maintenance, and customer support.
5. Features: A cloud-based IP PBX can offer a wide array of features similar to, or even exceeding, traditional systems. This includes call forwarding, voicemail, auto-attendants, call recording, conferencing, integrated messaging, and more.
6. Reliability: Reputable cloud PBX providers invest in redundant infrastructure to ensure high uptime. If one data center faces issues, the traffic can often be rerouted to another, ensuring continuous service.
7. Security: While transmitting voice over the internet does raise security concerns, many cloud PBX providers offer robust security measures, including encryption, fraud detection, and secure access controls. Additionally, they often have dedicated teams to monitor and address potential security threats.
8. Integration: Many cloud-based IP PBX systems can integrate with other cloud-based business tools, such as Customer Relationship Management (CRM) software, team collaboration tools, and more. This allows for unified communications and streamlined business processes.
9. Updates and Maintenance: With a cloud-based system, the service provider manages updates and maintenance. This ensures that businesses always have access to the latest features and security patches without having to worry about manual updates.

In essence, a cloud-based IP PBX offers businesses a modern, flexible, and feature-rich phone system without the need for significant capital expenditure or on-site technical expertise. It aligns well with the broader trend of businesses moving their IT infrastructure and services to the cloud.

An Internet Exchange (often abbreviated as IX or IXP for Internet Exchange Point) is a physical infrastructure where Internet service providers (ISPs), content delivery networks (CDNs), and other network providers meet to exchange internet traffic between their networks. This direct exchange of traffic, often called "peering," allows data to be transferred more efficiently and often with reduced latency and cost.

Here's a more detailed overview of Internet Exchanges:

1. Purpose: The primary purpose of an IX is to allow networks to interconnect directly, bypassing the need to route traffic through third-party networks. This results in faster, more efficient, and often cheaper data exchange.
2. Facility: An IXP is typically hosted in a data center. It consists of network switches, routers, and other infrastructure that facilitates the exchange of data between member networks.
3. Peering: Networks at an IX establish peering agreements, which are mutual agreements to exchange traffic. These agreements don't typically involve monetary exchange for the transit itself, though there might be costs associated with connecting to the IXP or for the port speeds desired.
4. Benefits:
   • Reduced Latency: By directly connecting to other networks at an IXP, data can take a shorter and more direct path.
   • Cost Savings: Peering can reduce the need to pay upstream providers for internet transit, which can be a significant expense for larger ISPs.
   • Resilience: With multiple peers, networks can handle the failure of one or more connections without significant disruption to their service.
   • Increased Bandwidth: Direct connections at IXPs can handle larger amounts of data than typical transit connections.
5. Route Servers: Many IXPs have route servers that facilitate the exchange of routing information using the Border Gateway Protocol (BGP). By connecting to a route server, a participant can automatically establish peering sessions with multiple networks without having to set up individual BGP sessions with each one.
6. Regional Importance: Many countries or regions have their own IXPs, ensuring that local internet traffic can be efficiently exchanged without having to be routed internationally. This can significantly speed up in-country or in-region internet communications.
7. Growth of Content Providers: In recent years, major content providers like Google, Netflix, and Facebook have become significant participants in IXPs, as these allow them to deliver their content more efficiently to end-users by peering their content delivery networks directly with ISPs.

In essence, Internet Exchanges play a pivotal role in the global internet infrastructure, promoting efficient data exchange, reducing costs, and improving the overall internet experience for end-users.

You need a BGP ASN which is usually assigned by a RIR, one IPv4 block (/24 at least) and one IPv6 block (optional).
You then need to establish a cross connect between your equipment and the equipment of the IX.
Once the physical part is achieved, you can start peering with other members and the route-servers via BGP.

BGP (Border Gateway Protocol) is the standard protocol used to exchange routing information between different autonomous systems on the internet. It's responsible for determining the best path for data transmission and ensuring the scalability and stability of the global internet.