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Navigating 10DLC Compliance with Voxtelesys: Everything You Need to Know

Navigating 10DLC Compliance with Voxtelesys: Everything You Need to Know

10DLC
Compliance
SMS
Navigating 10DLC Compliance with Voxtelesys: Everything You Need to KnowIn business messaging, ensuring compliance and maintaining high-quality customer communication is more critical than ever. One of the most significant developments in this space is 10DLC (10-digit Long Code) for SMS messaging, which enables businesses to send messages through local phone numbers. Voxtelesys is here to guide you through the process, ensuring your business adheres to all the necessary regulations and maximizes the benefits of 10DLC. Learn More
Case Study: Revolutionizing Call Flow Automation for Non-IOU Utilities with VAST Flow Builder

Case Study: Revolutionizing Call Flow Automation for Non-IOU Utilities with VAST Flow Builder

Calling
IT Integration
Business Solutions
Case Study: Revolutionizing Call Flow Automation for Non-IOU Utilities with VAST Flow BuilderEfficient and adaptable call flow automation is paramount in the evolving utility sector. As non-IOU utilities face increasing regulatory demands and pressure to enhance service reliability, leveraging cutting-edge technology becomes essential. Enter the VAST Flow Builder—a revolutionary tool to simplify and streamline complex telecommunication processes, empowering utilities to maintain a competitive edge. Learn More
Case Study: Enhancing Emergency Communication for East Coast IOU Utilities with Voxtelesys Solutions

Case Study: Enhancing Emergency Communication for East Coast IOU Utilities with Voxtelesys Solutions

IT Integration
Messaging
10DLC
Case Study: Enhancing Emergency Communication for East Coast IOU Utilities with Voxtelesys SolutionsFor major East Coast IOU (Investor-Owned Utility) utilities, communicating quickly and effectively during weather emergencies and peak load events is critical. With the growing frequency of extreme weather conditions and increasing energy demands, utilities must have robust communication systems to keep customers informed and maintain grid stability. Voxtelesys offers a suite of solutions, including VoxVoice VoXML, VoxSMS, and Voice 10DLC SMS/MMS Email, designed to deliver large-scale, reliable communication when needed most. Learn More

Popular Blogs

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
Case Study: Enhancing Emergency Communication for East Coast IOU Utilities with Voxtelesys Solutions

Case Study: Enhancing Emergency Communication for East Coast IOU Utilities with Voxtelesys Solutions

IT Integration
Messaging
10DLC
Case Study: Enhancing Emergency Communication for East Coast IOU Utilities with Voxtelesys SolutionsFor major East Coast IOU (Investor-Owned Utility) utilities, communicating quickly and effectively during weather emergencies and peak load events is critical. With the growing frequency of extreme weather conditions and increasing energy demands, utilities must have robust communication systems to keep customers informed and maintain grid stability. Voxtelesys offers a suite of solutions, including VoxVoice VoXML, VoxSMS, and Voice 10DLC SMS/MMS Email, designed to deliver large-scale, reliable communication when needed most. 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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