COVID-19 has pushed companies to work remotely, forcing them to ask themselves, How do we stay connected when we have to work apart?

One answer is to get the help of SIP trunking. A SIP trunk is like a virtual phone line, enabling you to connect voice calls over the internet to anyone with a phone, anywhere in the world.

As employers change their remote work policies to rely on video conferencing and VoIP-enabled calling, they need to make sure their SIP trunking partner provides the best call quality possible, since call quality is a big factor in whether remote workers can successfully work with their peers and clients throughout the world.

A quality SIP trunking provider helps you offer your customers inbound and outbound landline or mobile calling options no matter where they’re working. Your provider should do this by acting as your liaison to public telecommunications services and allowing you to do two key things:

• Provision global dial-in phone numbers — Call participants should be able to join a conference by dialing a phone number specific to their region, no matter where the other members of the call are located.
• Support dial out to any PSTN number — Call participants should be able to dial any public switched telephone network (PSTN) number and add the recipient of that call to a conference.

In the coming months, employees and companies will rely more and more on remote workers to get things done. Make sure that your company is ready to meet the demand and exceed expectations with the help of a SIP trunking provider that offers high call quality at a reasonable cost.

When to look for a new SIP trunking provider

Today’s video conferencing market gets increasingly competitive every day. Teams are more dispersed than ever, and organizations are looking for a product that allows them to easily and affordably connect with their employees and clients.

If your current SIP trunking service doesn’t match up to the quality of the rest of your product, it’s time for a switch.

There are three main reasons video conferencing service companies decide to switch SIP trunking providers:

• They want better call quality. The switch to a new SIP trunking provider who uses quality-based routing (QBR) instead of least-cost routing (LCR) gives companies and their customers better call quality than a company’s existing provider. It’s easy to fall for the low price point of carriers who offer LCR, but that cost savings can come at the expense of call quality, customer support, reliable coverage, and other key features.
• They plan to support inbound and outbound calling in new areas. A global SIP trunking provider extends inbound and outbound calling services to nearly anywhere in the world.
• They need more support than their existing provider offers. A new SIP trunking provider gives companies the technical and strategic support they need to provide excellent service to customers.

You need top-notch support, sound quality, geographic coverage, and flexibility. The right SIP trunking provider will work with you to ensure you — and your customers — have the connections you need to succeed.

What makes an ideal SIP trunking provider

Make sure your SIP trunking provider offers exactly what you need and doesn’t make you pay for what you don’t. Consider eight key factors:

1. Call quality

Customers care most about the quality of their calls. Your SIP trunking provider should provide access to local numbers via Tier 1 carrier networks. Your provider’s connections should also use the G.711 codec to support the landline-like quality customers demand.

To ensure the highest possible call quality, Plivo’s Zentrunk SIP trunking service terminates calls through one-hop local carriers for a low post-dial delay (PDD) to minimize the time between when customers dial conference numbers and when they’re connected. We use intelligent QBR to ensure no calls are dropped and audio isn’t delayed.

2. Global reach

Look for a SIP trunking provider that has connections with carriers in the regions where you operate. Though you may only require narrow geographic coverage now, partnering with a provider that offers restricted coverage will make it more difficult to scale as your business expands.

Your SIP trunking provider should have at least three carrier connections in major areas and support local, national, and toll-free numbers. Zentrunk offers outbound call service to nearly every country in the world, and inbound call service with phone numbers in more than 70 countries and counting.

Additionally, we offer points of presence (PoP) in seven locations across five continents to ensure low latency. When you partner with Zentrunk, your service also includes caller line identity (CLI, a.k.a. caller ID) support to customize the appearance of incoming calls.

3. Service reliability

Even the most robust carrier networks fail at times, so your provider must have multiple carrier connections and redundant infrastructure.

To ensure high availability, Zentrunk serves video conferencing providers by working with at least three local carriers in all major regions to offer SIP trunking services and 99.95% uptime.

4. Flexible pricing

Pay-as-you-go pricing and low upfront setup costs allow you to change phone numbers, adjust services, and add volume and IP access control lists without inflating costs.

Zentrunk’s outbound calling starts at $.0065 per minute and inbound calling at $.0025 per minute. Additionally, quality SIP trunking providers like Zentrunk offer volume discounts that make it quick and cost-effective to scale.

5. Interoperability

Adding SIP trunking shouldn’t take an army of engineers and hours of troubleshooting.

Ensure your provider’s services work with your existing VoIP infrastructure. Companies that partner with Zentrunk enjoy interoperability with standard soft switches and IP PBXs, so you can seamlessly and immediately integrate PSTN connectivity without having to do a costly, time-consuming infrastructure overhaul.

6. Instant provisioning

The needs of your company will evolve over time, and you need a SIP trunking provider that can grow and change with you. Instant provisioning of phone numbers ensures unlimited global scalability and the fastest setup possible.

With Zentrunk, you can immediately scale to more than 70 countries by instantly provisioning direct inward dialing (DID) phone numbers from our inventory of more than 5 million numbers.

7. Responsive support

Your SIP trunking integration should be simple to implement and maintain. But in the event something does go wrong, you need to have adequate support from your provider.

Zentrunk offers customers 24/7 customer support to make integration easy, even if you’re layering our SIP trunking solution onto your existing PBX system. We’re there to answer your questions, support your business objectives, and ensure you get the most out of our services.

8. Infrastructure security

Your SIP trunking technology upgrade should make your IT landscape and customer service offerings more sophisticated, not more complex. You should be able to easily segment and manage traffic using separate trunks with unique SIP domains.

As a global, modern SIP trunking provider, Zentrunk offers a robust set of features, including:

• Encrypted trunks with Transport Layer Security (TLS) and Secure Real-time Transport Protocol (SRTP).
• Fraud protection and alerts to automatically detect and notify you of fraudulent activity.
• A self-service portal to enable quick access and management of analytics, logs, and SIP trunks.
• IP authentication so you can securely access your SIP trunks using IP or credentials.
• Separated trunks to allow you to easily segment and manage each trunk with a unique SIP domain.

You’ve built your reputation on offering quality video conferencing services. Make sure your new SIP trunking provider measures up to your — and your customers’ — standards.

How Zentrunk helps you support customers when they need you

With Zentrunk, you have access to more than 1,600 carrier networks across the world and connectivity in more than 150 countries. Plus, our SLA-guaranteed uptimes of 99.95% help you avoid costly service issues.

If you’re looking to quickly add capabilities to your video conferencing services without causing a surge in costs, sign up for free and start testing Zentrunk or connect with an expert for a free consultation.

You can now send and receive MMS messages in the United States and Canada on all MMS-enabled Plivo long code phone numbers. With Plivo MMS, businesses now have a new channel to use to communicate with their customers, and the ability to create engaging visual experiences with pictures, audio, and video.

Our early MMS customers are using the new feature in several ways:

• Marketing campaigns: To improve open rates for marketing campaigns using rich media such as pictures and videos.
• Last-mile logistics: To identify personal information and shipping details using pictures and videos shared during delivery.
• Collecting account information: To collect account-related information such as pictures and receipts using MMS.

Sending and receiving MMS messages

Sending and receiving MMS messages is like sending and receiving regular text messages on the Plivo platform. You can make a POST request to the message resource URL or an API request to specify the image link in the Media URL parameter.

For more information, check out our API reference documentation.

Our MMS API includes many powerful features:

• Multiple sender types
  Send and receive MMS using any number type (long code, toll-free, short code) across the US and Canada.
• Variety of formats
  Support for pictures (JPEGs and GIFs), emojis, audio (MP3), and video (MP4) files.
• Built-in multimedia storage
  Plivo provides built-in multimedia storage for both inbound and outbound MMS messaging so you can access and reuse media.
• Intelligent picture resizing (coming soon)
  The MMS API intelligently resizes images to meet carrier guidelines and to ensure delivery across different carrier networks.
• Automatic MMS converter (coming soon)
  Automatic conversion of multimedia messages into SMS messages when sending messages to MMS-incompatible phone numbers to ensure better deliverability.

With our server SDKs and ready-to-use code samples, it’s simple to incorporate sending MMS message in your applications.

MMS is available for the US and Canada on long code phone numbers, and US short codes and toll-free numbers are both on the roadmap. Pay-as-you-go pricing for the US starts at $0.016 to send and $0.008 to receive, with volume discounts on committed spend as you scale your usage.

From our product design to our documentation, everything we do is designed to help you simplify your business communications, including the way you get set up, how rapidly you deploy, and how you manage day-to-day operations. That driving force applies to the Plivo console as well.

That’s why we’re launching an update to the Plivo console that includes changes to both the information architecture and UI. With this update:

• All product applications appear in a left navigation menu
• All components such as endpoints, call logs, alerts, and geo permissions nest within the left menu
• Links to documentation and support pages and billing and account information is consolidated at the bottom of the panel

In addition, we’ve standardized the interface across our products, creating a unified system that gives users a more comfortable workspace that’s faster and easier to navigate.

• Streamlined navigation panel: We’ve moved the navigation to a collapsible side-panel format. This new layout makes it easy to move between product workspaces and creates more room for working on your applications.

• More intuitive billing and account sections: The new format cleans up the billing and account sections and creates two separate areas: one for all things related to finance, including payments and invoices, and one for all things related to account security, such as users, subaccounts, team, and user credentials.

• Optimized view of application configuration: Now you can see at a glance all the details of your application configuration — phone numbers, all associated URLs for both voice and SMS, and SIP endpoints — without having to click through multiple layers to verify or modify the application configuration.
• URLs for quick editing: We’ve taken a couple of steps out of item editing. Instead of going to an entity list (Applications, Endpoints, Phone Numbers) and scrolling through a list of items, now you can navigate directly to an entity URL and edit.
• Easier UI for configuring phone numbers: We’ve eliminated multiple steps so that you can see the phone number configuration in the same panel where you see Your Numbers, and you can modify or attach a number to an application from the same screen.

• Simpler phone number identity documentation: Some countries require address and identity documentation before you can use a specific phone number. We’ve added two new features in the console that simplify the documentation process. First, we’ll tell you upfront about whether the documentation is required. Second, when you buy the number, you’ll have the option to attach the documentation.
• Product overview featuring analytics: We’re updating the analytics dashboard to give every product an overview page that will present a snapshot of product-specific analytics.

As always, we look forward to hearing your feedback about these changes.

Plivo makes it simple for businesses to send SMS text messages.  With a couple of lines of code, your application is ready to start sending messages to more than 190 countries. On average, our customers send out their first text message within 15 minutes of signing up.

The complexities of SMS messaging may surface once you start sending a high volume of messages. At that point you may face challenges such as managing country-specific restrictions, per-source-number SMS limits, and daily SMS volumes. You may have to write additional software logic to manage such complexities to ensure high SMS delivery rates.

To help our customers improve SMS delivery and go live faster for high-scale messaging use cases, we’re releasing Powerpack for SMS, a bundle of features that makes message communication at scale more effective and reliable. Powerpack ensures that your SMS messages get delivered at whatever scale you’re sending them, to any country.

How Powerpack helps

Powerpack provides several features that help high-volume messaging customers.

Number pools for easy scaling

With Powerpack, you construct a pool of source numbers optimal for your daily volume and peak send rate. Your messages are automatically distributed across this source number pool. All you have to do is specify the Powerpack’s unique identifier in the Send SMS API request.

You can add more numbers to your Powerpack number pool in seconds, right from your Plivo console. This lets you scale infinitely without having to make any code changes.

Sticky sender for a consistent experience

To ensure that your end users always receive messages from the same source number, Powerpack establishes a sticky mapping between the recipient’s number and the source number used from the Powerpack for the first text message to that recipient. This stickiness enables two-way conversations by keeping the conversation thread on the recipient’s handset intact.

Smart queuing for carrier compliance

Powerpack maintains an internal rate-limiting queue for every source number in the pool and sends messages to downstream carriers at a rate that’s compliant with industry-defined best practices. Source number-specific rate-limiting queues, in conjunction with Powerpack’s message distribution logic, ensure you meet your overall volume and throughput requirements while staying compliant with per-number limits.

Local connect for higher open rates

Powerpack prioritizes source numbers in the pool that are from the same geographic region as the destination number. The highest priority is given to numbers of the same area code, followed by those in the same state, then country.

For example, when sending an SMS to a San Francisco number that starts with area code 415, Powerpack will use a number in the pool that begins with +1-415. If no +1-415 numbers are found in the pool, Powerpack will use an available California state number to send the message.

The Powerpack management UI in the console allows you to provision number pools and all the other features with a few clicks.

Getting started with Powerpack

Getting started with Powerpack is easy. Log in to the Plivo console and click to Messaging > Powerpack. Follow the instructions on the console to create a new Powerpack.

To send messages with your newly created Powerpack, invoke Plivo’s Send SMS API with the powerpack_uuid request parameter set to the Powerpack’s UUID.

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import plivo

client = plivo.RestClient('<auth_id>','<auth_token>')
message_created = client.messages.create(
    powerpack_uuid='<powerpack_uuid>',
    dst='<destination_number>',
    text='Test Message'
)

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require 'rubygems'	
require 'plivo'	

include Plivo	

client = RestClient.new("<auth_id>","<auth_token>")
response = client.messages.create(	
  nil,	
  [destination_number],	
  'Test Message',	
  powerpack_uuid: '<powerpack_uuid>'	
)	
puts response

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let plivo = require('plivo');
let client = new plivo.Client('<auth_id>','<auth_token>');

client.messages.create({
  powerpackUUID:"<powerpack_uuid>",
  dst:'<destination_number>',
  text: 'Test Message',
},).then(function (message_created) {
  console.log(message_created)
})

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require 'vendor/autoload.php';	
use Plivo\RestClient;	

$client = new RestClient('<auth_id>', '<auth_token>');	
$message_created = $client->messages->create(
    [  
            "powerpack_uuid" => "<powerpack_uuid>",
            "dst" => "<destination_number>",
            "text"  =>"Hello, this is a sample text",
            "url"=>"https://<yourdomain>.com/sms_status/",
    ]
);
print_r($message_created);

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package com.plivo.api;

import com.plivo.api.models.message.Message;
import com.plivo.api.exceptions.PlivoRestException;
import java.io.IOException;
import java.util.Collections;

class Example {
  public static void main(String[] args) throws IOException, PlivoRestException {
    Plivo.init("<auth_id>", "<auth_token>");
    Message.creator("<powerpack_uuid>","<destination_number>"), "Test Message").create();
  }
}

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package main

import (
	"fmt"

	"github.com/plivo/plivo-go/v7"
)

func main() {
	client, err := plivo.NewClient("<auth_id>", "<auth_token>", &plivo.ClientOptions{})
	if err != nil {
		fmt.Print("Error", err.Error())
		return
	}
	client.Messages.Create(plivo.MessageCreateParams{
		PowerpackUUID: "<powerpack_uuid>",
		Dst:           "<destination_number>",
		Text:          "Test Message",
	})
}

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using System;	
using System.Collections.Generic;	
using Plivo;	

internal class Program	
{	
    public static void Main(string[] args)	
    {	
        var api = new PlivoApi("<auth_id>","<auth_token>");	
        var response = api.Message.Create(	
            powerpack_uuid:"<powerpack_uuid>",	
            dst:new List<String>{"<destination_number>"},	
            text:"Test Message"	
        );	
        Console.WriteLine(response);	
    }	
}	

Your message will be sent via one of the phone numbers in the Powerpack number pool.

For more information, visit our Powerpack documentation.

Pricing

Powerpack is enabled on your account at no extra cost. You pay only for actual text messages sent and received.

Upcoming enhancements

We’re actively working on additional Powerpack features. Some of the enhancements planned for the coming months include:

Alphanumeric sender IDs: Many countries allow using brand names and keywords as sender IDs. Soon, customers will have the option to configure Powerpack to use alphanumeric sender IDs for SMS messages to these countries.

Powerpack management APIs: We’re working on APIs to create new Powerpacks and manage number pools programmatically.

Content optimization features: We’re also working on features to better manage the content of your text messages based on country-specific support for characters sets and message length.

Not yet using Plivo? Getting started takes just five minutes. Sign up today.

Voice call quality may be impacted by many factors, and troubleshooting voice call issues requires technical expertise. It can be a time-consuming task, given that voice calls from an API platform often involve multiple call legs and precise orchestration of the call flow with the help of REST APIs, XML, and API callbacks.

Plivo’s new Call Insights service provides detailed statistics, actionable insights, and powerful analytics for voice calls from the Plivo Voice API platform.

Call Debug UI

In this release, we’re introducing a new Call Debug UI with a host of features to help you troubleshoot call setup and call quality issues.

Call state change timeline: The UI displays call initiation time, ring time, answer time, and end time information for every call, along with other basic details such as call type (VoIP or carrier), direction, and billing info.

Suspected issues and Plivo quality score: The panel shows a list of call quality issues suspected on the call, identified based on audio quality statistics collected for calls. Each call is also assigned a rating from zero (the worst) to five (the best) based on an analysis of audio quality metrics such as packet loss, jitter, and latency.

Call relationships: Conversational use cases often involve two or more parties bridged together with a distinct call leg connecting each party to the call. This section lists all of the call legs that were ever connected with a given call, so you can visualize the overall call flow and isolate the source of call quality issues on calls involving multiple parties.

Call hangup cause and hangup source: Why a call hung up and who hung it up are important clues in debugging call issues. Our descriptions of call hangup reasons and hangup sources help you understand why a call was terminated.

Call origination and termination details: We display comprehensive information about the complete routing path taken by Client SDK and carriers calls.

Client SDK calls now include the geographic location of the endpoint and rich metadata about the device used to place the call.

Information about carrier calls includes the media server location of Plivo’s infrastructure and the partner carrier, along with location and end operator details based on the from and to numbers.

Audio quality stats: We measure audio quality via several metrics — network jitter, packet loss, audio level, and network latency (round trip time). High jitter on an audio stream can lead to distorted audio on the receiving end, and high network round trip time (RTT) can result in delayed audio reception. Audio quality stats, per stream and for every call leg, are helpful in isolating the root cause of audio quality issues.

The information from Call Insights makes troubleshooting call issues easier and faster — and to make debugging issues using this data easier still, we’ve created a troubleshooting guide for common call issues as part of our documentation.

Proactive alerts on Plivo client SDKs

Plivo client SDKs for web browsers, iOS, and Android can now monitor call quality and network conditions in real time and send alerts when quality is low. The SDKs emit events for an array of network conditions that might result in a subpar experience on an ongoing call. By altering the application experience based on these events, you can turn a bad situation around for end users.

Call Insights is in public beta. While in beta, Call Insights is available to all Plivo customers at no extra cost. After general availability, premium features will available at a modest cost.

If you’ve made or received a phone call using a mobile app, such as WhatsApp, you’ve used Voice over Internet Protocol (VoIP). A mobile application enabled with VoIP lets you make and receive calls using mobile data instead of the device’s cellular network.

However, VoIP apps have to maintain a persistent connection with a server to receive incoming calls, which means constant communication between the app and the server in the background to keep the connection alive even when the app isn’t in use. Maintaining a persistent connection consumes battery power — but if you kill the app, it loses its connection to the server and cannot receive incoming calls.

Push notification is the solution

Using push notifications is a better way to handle incoming calls on your mobile app. With push notifications, smartphone users can receive an incoming call even when their phone is locked or an app isn’t running in the background.

Plivo mobile SDKs are built to provide a great VoIP experience in your mobile apps, and VoIP push support comes “out of the box.” Your voice-enabled app simply registers with Plivo to receive incoming calls and Plivo manages the rest.

Benefits of push notifications

Push notifications offer several benefits:

• An app can wake up when VoIP push occurs, saving battery power.
• VoIP push goes straight to the app for processing, improving efficiency.
• VoIP push is considered a high-priority notification and delivered to an app without delay.
• An app launches automatically when it receives a VoIP push, removing the requirement to keep the app running in the background.

Push notifications with Plivo mobile SDKs

The Plivo mobile SDKs for Android and iOS come integrated with push notifications. They use Apple Push Notification service (APNs) for iOS and Firebase Cloud Messaging (FCM) for Android to send notifications to an app.

Apple and Google issue push credentials for iOS and Android apps to receive incoming call alerts when push notifications are enabled on the app. Managing multiple push credentials becomes tricky when you’re building apps on different platforms. You’re likely to use unique push credentials for apps in different stages of development — alpha, beta, or general release. If you’re managing several apps with push notifications enabled, you need to generate credentials for each app.

You can manage push credentials on the Voice > Mobile Push Credentials screen of the Plivo console so that Plivo can deliver incoming calls to mobile apps.

To learn more about Plivo mobile SDKs and see how you can incorporate voice calling into your mobile apps, check out our SDK documentation for Android and iOS or get in touch with our team.

SIP trunking can provide cost savings for telephony customers, but pricing should be only one part of the equation when choosing a telecom provider. Any business looking for a new VoIP solution should also consider factors such as reliability and call clarity.

Buyers should be particularly wary of any provider that promises least-cost routing (LCR) to deliver lower prices. LCR works by being carrier-agnostic. For each call, carriers are ranked from lowest to highest price, and the call is attempted over the lowest-cost network. If the call can’t be completed over the lowest-cost carrier, the call is then attempted over whatever carrier ranks as having the next-lowest cost.

While LCR can make SIP trunking pricing appealing and works for some businesses, some LCR users pay in the form of lower overall call quality, because not all carriers provide the same level of service. It’s important to understand how a SIP service provider balances the two key factors of cost and call quality.

The challenges of using LCR

If a SIP trunking provider uses LCR, its calls may be routed to whatever network provides the lowest rate for a particular time and location, regardless of connection quality. By focusing only on cost, LCR users may experience challenges such as:

1. Audio cut-out: Since SIP trunking providers that use LCR don’t factor in quality, calls might be routed over networks that experience audio cut-outs, known as “jitter,” in which phone calls sound jittery or where words seem to be coming in out of order. SIP trunking providers route calls over the internet and the actual voice sound is transmitted as packets of data, so if a connection is unreliable, some data packets might come in out of order or not at all, which leads to jitter.
2. One-way audio: Similar to jitter, those who are on calls made via LCR can experience one-way audio if voice packets aren’t properly transmitted. When this occurs, it can result in only one party being able to hear the call, while the other party hears nothing and might think that the call dropped or the other person hung up.
3. Dropped calls: Again, by not prioritizing quality, those who are on calls made via LCR can find their calls dropped by unreliable networks.
4. Latency: Have you ever experienced a delay between the time someone on a call says something and the time the other person hears it? It makes having a conversation difficult, and it’s a problem that can occur with LCR when networks don’t transmit data fast enough. With good carrier quality, there should be no delay.  
5. High post-dial delay: LCR providers may try to connect over a network that’s at capacity and have to try again over the next-lowest-cost network. While this happens relatively quickly behind the scenes, you don’t want to waste precious seconds waiting for calls to connect every time, particularly at high-volume call centers.
6. Lack of support for automated operating systems: LCR providers may not be able to connect to a number that uses dual-tone multi-frequency (DTMF) signaling for interactive voice response (IVR), such as an automated operating system at a customer service call center that asks callers to enter information using a Touch-Tone keypad so calls can be routed to the appropriate department. Some LCR systems pick carriers based only on cost, rather than looking into whether the underlying carrier can support DTMF.
7. Lack of Caller ID retention: Calls transmitted using LCR don’t always retain calling line identification (CLI), which means the recipient might get a call from a number that displays as “Unknown.” As we all know, calls from an “unknown” number rarely gets answered. Even if the call displays a number, it may be a random number, not the one that the caller wants to display. That means that if a customer tries to call back based on the number displayed by the LCR carrier, they may not reach the original caller.

Selecting the right carrier

To avoid these pitfalls that accompany LCR, companies should look for SIP trunking providers that have these five attributes:

1. Media and signalling PoPs near where you make calls: Having points of presence (PoP) available close to the regions where you intend to make calls helps ensure that calls connect clearly and quickly, without the latency issues that can plague LCR.
2. Multiple local carrier connections: Similar to the benefits of having strong PoP coverage, having multiple local carrier connections improves routing efficiency because it eliminates latency and call routing.
3. Guaranteed DTMF support: To avoid the problems that can occur with an LCR that can’t connect to automated operating systems that use DTMF, look for a carrier that guarantees DTMF support.  
4. CLI retention: Make sure your carrier provides CLI retention so you can avoid the problems that can stem from not displaying the right phone number.
5. High availability with promised uptime: What good is a carrier if you can’t make calls due to network downtime? Look for a SIP trunking provider that can deliver consistently. Zentrunk, for example, promises 99.95% uptime.

Quality should be the priority

While LCR provides the benefit of cost savings, both SIP trunking pricing and SIP trunking quality need to be considered together, with quality coming first. That’s because “saving money” on ineffective calls often isn’t really saving money at all, due to lost productivity and missed opportunities from unreliable calls.

One benefit of a cloud-based SIP trunking provider like Plivo is that we work with multiple local carriers and can route your calls efficiently. Rather than ranking carriers solely by cost and routing accordingly, our intelligent routing system measures call quality and creates a quality score to reorder the priority of carriers and make calls through the highest-quality carrier available. Using self-learning algorithms, Plivo continuously searches for the best-quality route for calls.

Focusing on highest quality rather than the lowest cost helps ensure that SIP trunking as a whole is worthwhile for your business.

To better understand how to select the right SIP trunking provider beyond looking just at SIP trunking pricing, see our blog post on 10 factors to consider. To learn more about SIP trunking and see how Zentrunk can help with your communications needs, get in touch with our team.

Plivo’s cloud communication platform lets you send text messages globally via web and mobile apps. With our new feature, geo permissions for SMS, we’re putting more control of your SMS traffic into your hands. Geo permission offers a simple way to control which countries you want to enable or disable for SMS messages, without the need for developer support or code base changes.

SMS geo permissions is a tool to reduce the risk of SMS fraud and abuse. We suggest you maintain international geo permissions to protect your business from sending unwanted messaging.  

Advantages of maintaining international geo permissions

Here are some of the top reasons for you to consider managing your SMS traffic by geographic region.

• Prevent SMS fraud: Though it’s unlikely, your Plivo Auth ID or Auth Token may get compromised. If that happens and you have geo permission enabled, you can reduce the potential for a surcharge in SMS fees for messages sent to high-rate destinations.
• Reduce nonserviceable destinations: Among the 195 countries across the globe and more than 240 country codes, there are many places where your text messages could end up. If your web or mobile app caters to a specific geography or country, you may want to block SMS traffic to other countries and aligning your SMS spend with where your customers are concentrated.
• Avoid incorrect destination numbers: An incorrect or incomplete country code can land your text message in an unexpected part of the world. Some recipients’ phone numbers come from forms, and some form applications are not equipped to automatically geotag form inputs. Consider enabling geo permissions if you don’t validate user input using code.
• Improve customer segmentation: If you segment your SMS traffic using subaccounts, you can use geo permission to customize the reach for each subaccount. If a particular subaccount caters to a specific region, you can disable text messages to other destinations.  

Simple to set up, easy to maintain

Plivo customers can send messages to all countries, or with a click of a button disable messages to all countries except the ones where your audience is focused. By default we enable traffic to the US, Canada, Australia, India, and the country where your account is registered. To modify your geo permissions, navigate to Messaging > Geo Permissions on the Plivo console.

What happens if you send a text message to a disabled country?

If you try to send a text message to a country for which sending is disabled, Plivo will block it and you will not be charged for the message. On SMS logs, the message state will be “failed” and the error code will be 450.

Visit our documentation for more information about geo permissions.

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The global collaboration economy demands that voice conferencing providers seamlessly connect people for a reliable communication experience. Here are three ways Plivo optimizes our platform to provide clear, reliable conference calling.

1. Uniform customer experience with global coverage

When your customers get on a conference call, you want to provide them a hassle-free experience with easy connections and clear conversation quality. Plivo’s conference solution fits your needs at every stage:

• Coverage: We have a comprehensive network of Tier 1 carriers in more than 190 countries with voice and SMS to more than 1,600 global carrier networks.
• Call quality: In another post, we covered the steps we’ve taken to build our infrastructure with globally located data centers and failover mechanisms. Our optimized global routing minimizes latency, jitter, and packet loss.
• Proactive testing:  With the help of quality-based routing and distributed test nodes, we continuously test and improve call quality.
• Redundancy: We maintain redundancy and auto-failovers across all layers of our carrier integration system. This setup protects our customers from downstream risks such as carrier outages and network failures.
• Direct connections with SLAs: Because we maintain premium direct routes in more than 190 countries, we guarantee timely delivery to any mobile network in the world. We don’t use aggregators — we work directly with all of the major telecom operators in every region, which helps us deliver better quality and resolve issues rapidly.  

2. An architecture that improves call quality and minimizes call latency

We have seven points of presence (PoP) locations spread across five continents — in California, Virginia, Frankfurt, São Paulo, Mumbai, Singapore, and Sydney. We established these PoPs to minimize latency and optimize global routing.

3. Capabilities to derive analytical insights

In communications, network flaws like packet loss, jitter, and latency can reduce call quality. Having data about your network traffic is the only way to monitor performance to inform real-time decisions.

Plivo’s analytics let you track metrics such as latency, jitter, packet loss, and Mean Opinion Score (MOS), which in turn lets you assess call quality, enables faster debugging of issues, and provides real-time detection of network problems.

Plivo can provide feedback after every conference call to help you and your customers troubleshoot issues. Imagine being able to debug or troubleshoot and determine whether poor call quality was due to a congested network connection or a bad headset.

One more thing

Our goal at Plivo is to simplify business communications. Features like direct relationships with carriers, optimized PoP infrastructure, and real-time analytics are just a few of the ways we help make it super simple for you to integrate voice communications into your applications. We also understand that you may have questions or need support, so we offer all of our customers round-the-clock support at no extra cost.

Connect with us and let us answer any questions you have.