The technology of voice communication over the Internet, or VoIP telephony for call centers, enables regular phone calls as well as the transmission of text messages and video communication over the Internet. The technology offers unparalleled flexibility, cost-effectiveness, and advanced features. In this article, we will explore the most common questions regarding IP telephony in the context of contact centers.
What Telephony is Needed for a Call Center?
Implementing VoIP for a call center allows addressing numerous weaknesses in processes related to handling inbound and outbound calls. Opting for this technology in a contact center brings the following advantages:
- Synchronization with any information system (CRM, CMS, etc.)
- High reliability and uninterrupted operation, minimizing service disruptions
- Ability to work from any location with internet connectivity
- Ease of scaling up or down based on call volumes and business needs without significant infrastructure changes
- Even distribution of workload among call center operators, considering each employee’s skills and call processing speed
- Real-time analytics of agent performance and overall call center operations for data-driven decision-making
- Call recording, analytics, and automatic routing, conference calls, instant messaging exchange, IVR voice menus, mail, redirection, queues, and other features for enhanced customer interaction.
Even without considering the usefulness of additional features that positively impact overall productivity, VoIP for call centers has a significant advantage over traditional phone communication – a substantial reduction in call costs.
How is VoIP Different from Traditional Phone Service?
Traditional telephony relies on analog signals transmitted over dedicated copper lines, whereas VoIP uses the internet to transmit voice data in digital packets. VoIP for call centers ensures cost-effectiveness by leveraging existing internet infrastructure, reducing the need for dedicated phone lines.
What is SIP (Session Initiation Protocol)?
SIP telephony is a voice communication technology that enables phone conversations over the internet using the standard SIP (Session Initiation Protocol). Specialized equipment receives the signal (interlocutor’s speech), encodes it into a digital format, compresses it, and sends it over the internet.
The SIP network protocol contributes to the adaptability and scalability of VoIP in contact centers. SIP-based systems seamlessly integrate with other communication tools and applications, creating a unified environment for collaboration.
How to Connect VoIP for a Call Center?
For VoIP to function, a stable and fast internet connection is required. This involves dedicated channels with a bandwidth of at least 96 Kbp/s per line in each direction.
In general, VoIP connection is facilitated by the service provider. For example, Global Bilgi offers VoIP connection for call centers or offices based on the Sirius cloud solution.
Why Choose Global Bilgi for VoIP?
The Global Bilgi cloud contact center offers numerous advantages for businesses of any size and industry. Our corporate cloud contact center platform is a web application that unifies all communication channels in one interface (VoIP, web chats, messengers, email) and provides in-depth analytics of agent and client actions.
Our specialists will configure the connection for calls according to any scheme allowed by VoIP:
- Computer to computer
- Computer to phone and vice versa
- Web phone (Web Phone)
- Softphone (Software Telephone
Additionally, we can integrate VoIP with other systems:
- Callback forms
- IVR systems
- Call tracking
- Call recording
- Dialer
- Online chat
- CRM systems
- Toll-free number connections
All of this is available in a user-friendly interface that allows complete control over the company’s communication system. VoIP based on the Sirius cloud platform from Global Bilgi provides incredible opportunities for your business. Simply contact our sales department, and we will prepare an individual offer for you!
Equipment Required for VoIP in a Call Center
Until recently, the standard equipment for VoIP in a call center was IP phones—stationary (wired or wireless) devices that transmit calls over the internet. A more modern solution is a Softphone.
Softphone (Software Telephone) is specialized software designed for installation on a PC, laptop, or tablet. The software allows displaying call statuses, placing calls on hold, recording them, keeping statistics, analyzing data, or automating dialing. Developers typically offer packages with various features. Softphones, with their extensive capabilities, are a common solution for call centers and technical support lines.
One of the advanced solutions is WebRTC technology, enabling the reception and processing of calls in a single browser window without the need for additional software installation on the user’s device. In this case, the software for making and receiving calls through the web browser is called a Web Phone. The Sirius contact center platform from Global Bilgi operates on this principle.
What is WebRTC?
WebRTC (Web Real Time Communications) is a technology for establishing real-time communication over the internet. It enables the streaming of audio, video, and graphic data online. With WebRTC, any compatible browser becomes a working tool for the call center—just open the page and follow the web link. No need for additional plugins, modules, or third-party programs.
How Does WebRTC Work?
The WebRTC workflow involves four key stages:
- The first user opens a web page working with WebRTC.
- The browser requests permission to access the user’s microphone and webcam (if video communication is involved).
- The browser generates and sends a text file, known as an SDP (Session Description Protocol) packet, containing the description of the established connection: video, audio, graphics, codecs, browser parameters, etc.
- The second user’s browser receives the SDP packet, generates, and sends a similar one.
After the browsers exchange data, a stable connection is established between them, supported by network protocols.
WebRTC technology is developed on the principles of open-source software, making it freely deployable for any relevant service. It is ideal for use in cloud call centers, allowing making and receiving calls from a web browser without the need to download and install a softphone.
What Internet Speed is Needed for VoIP?
The main internet parameters affecting VoIP for call centers are network speed and bandwidth.
Bandwidth for VoIP depends on the number of simultaneous calls. For high call quality, the minimum internet speed for VoIP should be around 90-100 Kbps. This value needs to be multiplied for each user. For example, if you use 10 VoIP phones, you’ll need approximately 1000 Kbps of bandwidth.
Additional Internet Parameters for VoIP
VoIP for call centers can work on both wired and wireless connections (Ethernet cable or WiFi, 4G/5G mobile networks). However, in practice, wired internet is most commonly used in call centers. Certain requirements must be met for quality voice communication:
Internet Speed for VoIP
For quality voice communication, the internet speed should be around 100 Kbps (for one channel). When operating a call center with CRM, built-in functions like auto-dialer, mailer, online chats, and video communication, a speed of 512 Kbps is recommended due to the presence of service data transmitted along with telephony data.
Ping for VoIP
Ping (network delay) is a parameter that helps evaluate the connection speed between your computer and another device on the network. The lower the Ping (measured in milliseconds), the better the connection. When designing VoIP for a call center, it’s essential to ensure that the maximum Ping does not exceed 150 ms.
Jitter for VoIP
Jitter is the deviation or delay in delivering data packets over the network, measured in milliseconds (ms). In IP telephony, especially in call center operations, high jitter negatively impacts call delay, interruptions, echoes, and distortions in audio and video. As long as jitter does not exceed 20 ms, these problems are unlikely.
Network Losses for VoIP
Another negative phenomenon in VoIP is packet loss. Since packets transmitting voice data are not resent, their loss in the network leads to short pauses. Frequent packet losses in the communication channel worsen voice clarity and sometimes result in complete communication failure.
For VoIP in a call center to ensure good communication quality, an acceptable packet loss level should be within 0.5%.
How to combine VoIP and mobile communication in a call center?
Integrating VoIP into a call center and mobile communication is easy when using FMC technology. FMC (Fixed Mobile Convergence) is a technology that combines mobile communication networks and the Internet network, through which IP telephony data is transmitted. Essentially, it is a hybrid PBX that, unlike a virtual telephone exchange, operates with two communication methods—VoIP and GSM. The hybrid nature provides an additional advantage: if the internet connection is lost for any reason, all employees will remain connected.
FMC (Fixed Mobile Convergence) – how it works
Technically, it looks like this: each SIM card is assigned its internal number on the operator’s virtual PBX. Essentially, the SIM card in the mobile phone becomes an internal subscriber. It can be called using a short dialing code from another similar SIM card or from an office phone. Call routing is done at the server level.
Unlike IP telephony and connecting SIP clients on a mobile phone, FMC works through the mobile channel (GSM). This eliminates issues with interruptions due to unavailable or slow mobile internet. Each device on the network receives a shared working or fixed corporate number. This way, call center employees can make calls from their FMC devices, and clients will see the company’s shared numbers.
Advantages of FMC for call center operations
The main advantage of FMC technology for call centers is flexibility and scalability when connecting—one device can be used for all communication methods, various networks (VoIP, mobile communication, Wi-Fi networks), and simultaneous connection of a large number of devices. Additionally, the use of FMC provides the following advantages:
- General cost reduction, as there are no charges for roaming, call redirection, calls between corporate network subscribers, and other services.
- In the case of remote work for agents, calls will be automatically forwarded to their mobile number, and the call will be made by the corresponding PBX. If the company has offices in multiple cities or countries, this provides a significant gain in communication quality and costs.
- No direct dependence on the quality of internet connection or mobile signal.
- Regardless of the type of PBX (physical or virtual), communication quality remains high.
For call center managers, FMC offers a specific advantage—call control, including calls made with a mobile phone. Call analytics and reports allow improving customer service quality.
FMC is a modern and high-quality telephony solution for call centers, which you can use based on the cloud platform Sirius from Global Bilgi.
What is a virtual phone number?
A virtual phone number is a number that can be used for calls, SMS, and other forms of communication over the Internet, without being tied to a physical device. The main difference between a virtual number and a regular one is that it is not linked to a SIM card and can be used on different devices simultaneously. Internet connectivity is sufficient for operation.
Advantages of a virtual number for call center operations
Virtual numbers have become an integral part of VoIP in call centers. They enable the mobility of business communications, allow the use of IVR menu services, facilitate call redirection, establish schemes for distributing inbound and outbound calls, receive faxes, and much more. When using VoIP in a call center, virtual numbers offer the following advantages:
- The ability for operators to work remotely, receiving calls through redirection from the office PBX or via a cloud service.
- Recording and listening to calls in a personal account.
- Easy connection and effective handling of multi-channel lines.
- Call distribution in queues.
- IVR menu simplifies sorting inbound calls and facilitates navigation for clients.
- Simple and convenient cost control with detailed call breakdown in the personal account.
- In case of complex issues, unusual situations, or the need for additional consultations, calls can be transferred to the required expert.
- After completing a conversation with an operator, the client hears a sound message inviting them to rate the service quality.
- In the personal account, you can configure notifications for various events via SMS or email.
How to get a virtual number?
There are several main ways to obtain a virtual number:
- Purchase a virtual number from a mobile operator or VoIP provider.
- Rent from specialized services that provide virtual numbers for a specific period.
- Use mobile apps that allow obtaining a virtual number directly on the device.
When choosing a method, it is essential to consider the usage term, required functionality, and the reliability of the service provider.
How much does VoIP for call centers cost?
As mentioned, VoIP for call centers has a significant advantage over traditional telephone communication—a substantial reduction in call costs. The cost of connecting VoIP in a call center depends on many factors, such as the number of operator workstations, the need for integration with other systems (IVR, dialer, online chat, quality assessment system, analytics modules). The best option is to acquire an all-in-one solution—a cloud contact center platform that saves costs and provides quality service.
What is a cloud contact center?
A cloud contact center is a convenient and economical solution for organizing VoIP in a call center. Using the SaaS model, a cloud contact center allows the rapid implementation of an in-house or remote sales department, customer service call center, and quick scalability.
The Sirius cloud contact center platform from Global Bilgi
Drawing on our experience in contact center outsourcing and customer experience management, we have developed a comprehensive solution for a cloud contact center with corporate capabilities for businesses of all sizes and industries.
Integrate all contact center channels on one platform
Software for the call center from Global Bilgi will help organize an efficient multi-channel contact center with VoIP, web chat, messengers, integration with social networks, and outbound calls through automated dialing software. It enables the creation of complex IVR trees and provides detailed real-time analytics on a single web interface.
The Sirius cloud contact center platform from Global Bilgi is your ticket to a modern, efficient, and fully integrated call center with VoIP support.
