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Simplifying Regulatory Compliance and Reliability for Power Utilities with Voxtelesys' VAST Flow Builder

Simplifying Regulatory Compliance and Reliability for Power Utilities with Voxtelesys' VAST Flow Builder

Business Solutions
IT Integration
Call Center
Simplifying Regulatory Compliance and Reliability for Power Utilities with Voxtelesys' VAST Flow BuilderManaging customer interactions effectively is crucial for power utilities, especially in a highly regulated and reliability-focused environment. Clear and efficient communication is critical to complying with regulations and ensuring reliable service. Properly handling phone calls is a vital part of this process, and that's where the call flow builder from Voxtelesys comes into play. Learn More
VAST Flow Builder is Here: Unlock the Power of Workflow Automation

VAST Flow Builder is Here: Unlock the Power of Workflow Automation

IT Integration
Call Center
CCaaS
VAST Flow Builder is Here: Unlock the Power of Workflow AutomationIn the world of modern telecommunications, staying ahead means constantly innovating and optimizing how you manage customer interactions.We’re thrilled to announce that the VAST Flow Builder, a tool designed to take your workflow automation to the next level, will officially launch on September 16th. If you’ve been searching for a solution that rivals the capabilities of Twilio Studio but offers even greater flexibility, customization, and affordability, look no further. Learn More
Get Ready for VAST Flow Builder: Revolutionizing Workflow Automation

Get Ready for VAST Flow Builder: Revolutionizing Workflow Automation

Business Solutions
IT Integration
Call Center
Get Ready for VAST Flow Builder: Revolutionizing Workflow AutomationIn today’s fast-paced business environment, automating and optimizing workflows is critical to maintaining a competitive edge. We are excited to introduce the VAST Flow Builder. This tool will make your complex telecommunication processes more manageable. Learn More

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3CX Version 20

3CX Version 20

Call Center
SMB
PBX
3CX Version 20Take advantage of our offer: No setup fees will be charged for upgrading to 3CX V20 with Hosting by Voxtelesys until March 2024! - 2 Core, 2 GB All 3CX's hosted by Voxtelesys come standard with a minimum of 2vCore and 4GB's of memory, so no worries here. - Sufficient Disk Space needed. Ensure a minimum of 5 GB of free disk space - The source list must remain unaltered for a successful upgrade; any modifications will result in failure Remove any additional source lists. If you are utilizing Microsoft Azure, verify by checking "cat /etc/apt/sources.list.d/microsoft-prod.list." Learn More
3CX's Latest Release: Geo-Routing Headers Take the Lead in Dynamic E911 Integration

3CX's Latest Release: Geo-Routing Headers Take the Lead in Dynamic E911 Integration

3CX
911
Calling
3CX's Latest Release: Geo-Routing Headers Take the Lead in Dynamic E911 Integration3CX is leading the telecommunications industry with its new release, v20, which features an innovative integration of Dynamic E911. The main change in this update is that 3CX has decided to use geo-routing headers. This move simplifies the process and enhances the reliability and efficiency of emergency call routing. Learn More
Host Your 3CX with Voxtelesys

Host Your 3CX with Voxtelesys

Business Solutions
Call Center
Hosted
Host Your 3CX with VoxtelesysEffective communication is the cornerstone of success in today's fast-paced business environment. As organizations strive to enhance their telecommunication infrastructure, 3CX emerges as a leading CCaaS solution, offering flexibility, scalability, and powerful features. Partnering with Voxtelesys, a renowned name in telecommunications, provides premium hosting and support for your 3CX setup, ensuring seamless, secure, and superior business communication. Why Choose 3CX? 3CX is an open-platform, software-based PBX system that delivers voice calls, video conferencing, live chat, and SMS. It's designed for businesses of all sizes, helping to reduce communication costs, improve customer experience, and boost productivity. The Voxtelesys Advantage offers unmatched reliability, optimized performance, enhanced security, scalable solutions, and expert support, ensuring your 3CX system effectively addresses voice communications' unique demands effectively. With Voxtelesys, transitioning to or upgrading your 3CX system is seamless, providing a robust, reliable, and efficient communication system that is essential in the digital age. Hosting your 3CX with Voxtelesys gives your business a competitive edge, transforming how your organization connects, collaborates, and thrives. Learn More
Home / Learning Hub / Blogs / Understanding The Language of Telecom Part 3FAQs
Understanding The Language of Telecom Part 3
SIP/VoIP
Explain It

In parts one and two of Learning the Language of Telecom, we looked at some of the key terms that call center owners and administrators need to understand about the traditional public switched telephone network, or PSTN. In this post, we’re going to guide you through the landscape of internet telephony, so that you understand the basic differences and can talk the talk when the conversation turns to VoIP, SIP, and IAX.



The Rise of Packets

When you call your friend to say, “I am going to eat lunch at the park after taking my dog to the vet,” you take it for granted that they’re going to hear it correctly. At the heart of every phone call that is carried across the public switched telephone network (PSTN) is a calibrated transmission of sound between two parties on one line, connected by circuits and switches that span the globe. When you speak into the phone, your voice is converted into signals which are transmitted across the network and then converted back into audio by the person you’re calling. The key here is that all those signals are transmitted and received in order and that you don’t have to worry that your friend is going to hear “I am going to eat my dog at the park after taking lunch at the vet.”

But the internet doesn’t carry information in a strictly linear process. Instead, it delivers packets of information. Imagine mailing ingredients for a recipe to a relative, and sending each one in an envelope (one with salt, one with pepper, one with chili powder, etc.). The idea is that when those envelopes are delivered, your relative could open them up, empty them in a pot, and start cooking. The order wouldn’t matter if all the packets reached their destination.

Conducting phone calls on the internet, or Voice over Internet Protocol (VoIP), required a new generation of network technology to ensure that each packet of information created when a caller speaks is sent and received in the right order and that they’re delivered in half of the time that it takes to blink your eye (150 milliseconds).



Protocols

While the traditional definition of protocol refers to customs and etiquette, computer protocols refer to the rules surrounding the format of information exchanged between two parties. This is complicated by the fact that, instead of sharing one line, Internet exchanges require two streams, one for each party. Each protocol was designed to solve the problem of efficiently slicing and dicing the information and enclosing them into packets, and delivering those packets quickly and flawlessly.

You may wonder what has driven such complicated innovation, but the primary reason is simple: VoIP offers cost savings that are an order of magnitude lower than using the PSTN.



SIP

SIP stands for Session Initiated Protocol and is used to establish, modify, and terminate internet phone calls. It is a widely accepted and increasingly popular protocol, due primarily to its flexibility. Since SIP is dedicated to starting, maintaining, and ending a call, it helps provide a measure of reliability that users expect when using the phone.

What’s important to understand is that it isn’t actually carrying the information across the network. It relies on the Realtime Transport Protocol, or RTP, to transfer the voice or data packets between two ports, one for each stream. SIP devotes a third port for signaling, whether it is to begin a phone call, terminate a call, or add additional parties to the call.

So, if your call center uses SIP and you have 50 agents making calls, they will require 100 ports, plus one port for signaling (which is shared across all users).



T38 FAX

The T38 protocol defines how a fax should be sent over a digital network. A fax cannot be sent in the same way voice packets are transmitted. The Fax was originally designed to work on analog networks and special considerations must be made when switching over to a VoIP based phone system. In particular, the use of an ATA (Analog Telephony Adapter) that supports the T38 protocol is necessary for older equipment. Many VoIP providers support the T38 protocol and will readily provide service that ensures your FAX reaches its destination.



IP PBX: Internet Protocol Private Branch Exchange

As opposed to a traditional private branch exchange (PBX), an IP PBX is a private branch exchange that switches calls between VoIP users on local lines while letting all users share a fixed number of external phone lines. SIP trunking provides a virtual connection between VoIP users and the PSTN, allowing them to make direct calls to any phone on the PSTN without traditional phone lines by connecting to a hosted PBX system.

Putting these terms together, a call center operator needs to know that:

  • An IP PBX can switch calls between a VoIP user and a regular telephone user.
  • Using SIP, the PBX can route inbound calls from a business’ local, toll-free, or even international number.
  • An IP PBX doesn’t need to be tied to a physical location and can be cloud-based to support call centers or agents on multiple sites.
  • They integrate with existing systems, such as the Customer Relationship Management system (CRM).

When evaluating on-premise vs. hosted IP private branch exchanges, keep in mind that an on-premise IP PBX requires a strong network infrastructure. A hosted IP PBX system, on the other hand, is delivered via the cloud. A third party manages the IP PBX, including set-up, maintenance, upgrades, and monitoring bandwidth, while guaranteeing quality and uptime.



Conclusion

There are many protocols beyond SIP that are used in a VoIP or IP Telephony network, including H.323, RTP, RTCP, TLS, SRTP, IAX, MGCP, and UNISTIM. Remember, even though SIP is only one of the many protocols used in a VoIP network the term SIP is used by carriers and customers interchangeably with VoIP.

With this brief introduction to the language of telephony, you should be better prepared for the conversations ahead. To learn more about finding the right VoIP solution for your call center, contact Voxtelesys today.

 

Read more articles in this Explain It! series:

Part 1: Helping business Owners Understand The Language of Telecom

Part 2: Understanding The Language of Telecom: PSTN, Exchanges, PBX

 

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Illustration of a brain, gears, and telephone terms

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