All About Voice Services – Changing the Working World

This is no joke. As of 2017, nearly 4 million people in the United States are working remotely in some capacity. With social distancing and pandemic prevention now at the forefront of every company culture, this number is expected to increase further. 

As a result, traditional telephones have become rarer as voice service technology continues to grow in popularity. To help your business thrive in this new environment, let’s cover the basics of this critical emerging factor in the workplace:

What Are Voice Services? 

A voice service uses the internet and an IP network to route vocal conversations through digital channels. The services are also referred to VoIP (voice over internet protocol) and VaaS (voice as a service). 

Businesses use these technologies to place and receive calls through their internet connection instead of through traditional telephones. VoIP systems can be used in a variety of ways, including: 

  1. A VoIP adapter and a telephone: You can use VoIP with a landline phone so long as you have a VoIP adapter. These adapters are either plugged into the router or phone socket; you might hear these referred to as ATAs (analogue telephone adaptors). An ATA works by taking an analog signal, converting it to digital, and then sending it across the Internet.
  2. Using a computer: There are several different programs that you can use to make voice calls over the Internet, including Google Talk, Apple FaceTime and Skype.
  3. Call on your smartphone: It’s also possible to use your smartphone to make voice calls using the Internet. Apps which allow you to do this include FaceTime, FaceBook Messenger, Google Meet, WhatsApp, and many more. 
  4. IP Phones: These phones look the same as conventional telephones, yet the hardware is a little different. Instead of an RJ-11 connector, these phones are built with RJ-45 Ethernet connectors. All you have to do is link these phones to your router and they possess all the software and hardware to make IP calls.

How Do Voice Services Work? 

VoIP converts voice to a digital signal. This signal is then compressed and transferred over the internet to the receiving endpoint where the data is uncompressed. 

A standard VoIP configuration uses an SIP server and a desk phone, storing your data in the cloud. It’s simple to manage a VoIP system straight from your online dashboard. Using your dashboard, you can add new contacts, forward calls and adjust your calling preferences. Depending on the VoIP solution you select, there may be additional features like instant messaging, calendar integrations, and more. 

Why Should a Business Use VoIP? 

There are many benefits of VoIP, especially for businesses. A few of these include:

1 .Reduced expenses: Using a VoIP system allows businesses to make significant savings because VoIP systems are far cheaper to operate when compared to a landline. Firstly the phone bill itself will be significantly lower, and secondly, you may require less hardware.

2. Supports remote working: When you switch to a VoIP set up, your staff can connect to your business telephone network while working from home. In the modern business world, more professionals are working from home than ever before. Some studies have even indicated that remote working can increase staff productivity.

3. Improved accessibility: Using a VoIP system, businesses can make calls from any location, including calls on-the-go. When you’re no longer restricted to a single location, you’ll ensure that your business is more accessible, flexible and productive.

4. Scalability: With VoIP, you get a system that can grow with your company. There’s no need to invest in a dedicated line or more hardware as your company gets bigger. A VoIP system is scalable; all you have to do is adapt your preferences.

What affects call quality?

To get the most out of your voice services solution, you must invest in a high-speed internet connection. Poor or subpar connections may dilute call quality and cause issues with the VoIP system’s overall performance. 

When using voice services, the sound is converted into what are called “packets”. There are thousands of these packets, all of which are transferred over the Internet during the call. Several factors have an impact on packet transport, and this affects session quality. The following issues may arise:

1 . Packet loss: Also called “data loss,” this happens when the network drops a few packets because the routers are congested. It could also be caused by jitter buffer resulting in discarded packets. When this happens, words or syllables may be missing from the conversation. 

2. Jitter: Jitter means that the arrival rate of the data packets is inconsistent. It’s caused by route changes, a timing drift, or congestion on the network. If the Jitter time exceeds 30ms, then packet drops may occur, negatively impacting the call-quality. Jitter buffer discard means that packets were dropped due to delays.

3. Latency: Latency is sometimes referred to as delay; it means how long it takes voice packets to get to their end destination. The latency is measured using milliseconds. Generally speaking 150ms or under is considered acceptable, anything higher than this may compromise the quality of the call.

How Does SIP Trunking Factor Into All of This?  

Session initiation protocol, or SIP, refers to the way that a VoIP call is achieved. SIP technology creates, modifies, and ends sessions within an IP network. These sessions could include conference calls with multiple parties or two-way calls. 

Think of SIP as an application protocol for creating real-time video or audio. The protocol functions by sending messages between different SIP address which could include voice calls, instant messages, or videos. 

The SIP functions are facilitated through the SIP trunk, which you can think of as a digital replacement for the analog telephone line. With SIP trunks, your provider can connect multiple channels to the PBX for international, long-distance, or local calls over the internet. With this type of system in place, you’ll experience zero restrictions on the amount of simultaneous calls.

Conventional business phone systems consist of the PBX and the PRI lines. The PBX offers call management, including voice mail and auto attendants. The PRI functions by linking calls to the Public Switched Telephone Network. Next, the calls are routed to the correct endpoint. When SIP trunks are used, the IP-based PBX accesses the data network, (as opposed to the PRI phone lines). The voice data is sent across the web to connect to the Public Switched Phone Network.

How Can SIP Improve Voice Services? 

SIP functions by linking packets of voice data to telephone switching systems; using robust processing features to improve the quality of your calls. Conventional telephone lines can only match the quality of the line itself and will decrease according to the distance. 

Instead of relying on a physical location, SIP trunks can achieve and maintain optimal call quality despite distance. Quality is maintained because SIP transmits packets of data using the shortest digital path possible. SIP trunks also use buffers to minimize issues with jitter and latency.

VoIP services can often experience issues such as jitter, packet loss or latency. Fortunately, SIP systems can help to remedy all of these issues. Packet loss usually occurs due to network congestion. SIP services can determine the precise amount of traffic that packets will require, dedicate task-specific bandwidth, and  improve the handling of network traffic. SIP can determine precisely the amount of traffic that the voice packets will take. Using this data the SIP can separate the packets from the other internet traffic.

SIP can help to minimize the latency by reducing both jitter and packet loss by using buffering systems to hold the data so that the delay cannot interrupt the actual call. Jitter happens when voice packets fall out of sync. For instance, later and earlier messages may deliver out of order. Again, SIP trunks can correct these by using buffering systems, and ensuring that packets are not lost.

How is SIP encoded?

SIP calls need to be encoded before they are sent over the Internet, which is why speech needs to be transferred as data. First, the audio signals are converted into codecs; from here, they are sent from the SIP endpoint to the final destination. For most audio calls, SIP will use codecs such as G729 or G.711.

The G7.11 codec does not compress the voice data when transferring the audio signals. There is no loss of quality; the codec uses bandwidth to achieve the highest quality possible. The G.729 operates on reduced quality because it does compress the voice data. Usually, this codec is used in situations where bandwidth is unavailable or restricted.

SIP.US Can Take Your Voice Services to the Next Level 

SIP.US is a SIP trunk provider to analog/digital telephone adapters and IP-PBX systems. We provide unlimited rate plans and operate SIP trunks using your existing Internet connection. We offer all of this with an industry-leading Tier-1 network, simple setup, and a robust self-service SIP control panel. 

By using our SIP trunking solutions, you can help streamline your digital communications for any and all workplace needs. As the working world becomes more remote, it’s best to be prepared.