What is IP Telephony (Internet
Traditional phone systems, such as landlines, mobile phones or smartphones, allow us to communicate at a distance. So does IP (Internet Protocol). Yet, instead of using regular analog transfer through a telephone line, voice is converted into digital information and delivered over the Internet. It allows you to make calls online, with no need for limiting hardware solutions.
Using IP telephony can make your business much more efficient. All you need is a stable network, quality internet connection and a reliable provider.
What is IP Telephony?
IP telephony arrived on the market in 1995, causing massive changes in the telecommunication industry. Today, carriers are using IP either for a part or all of their services.
It became the general transport for data communication.
IP telephony replaces traditional landline phone networks and telephone systems. It does not require any hardware and thus no regular maintenance. All you need is a stable internet connection, strong network setup and a one-time download of an app for any device through which you can make phone calls.
Just imagine Skype, Viber, Facebook Messenger, Google Voice, WhatsApp – these are examples of applications that use IP telephony and allow you to make calls for free over the Internet. People use these apps all over the world to talk with friends and family. Now, businesses can as well take advantage of call center software features.
It makes communication with customers and clients much more effective.
How Does IP Telephony Work?
IP is a general term for the technologies that use internet protocols to transmit digital signals over the Internet and enable telephone communication.
The protocol defines how signals should travel over the network. Similarly to HTTP (HyperText Transfer Protocol), it sets how data is to be transmitted, formatted, and displayed in web servers, as well as web browsers.
IP works by scanning and consequently identifying speakers’ analogue voice signals and turning it into a digital version of these signals through PSTN (Public Switched Telephone Network).
After transformation, digital signals are transmitted through a broad line in a data format, reconfiguring a phone conversation. Signals are then directing the data via an IP system into the network and out to be accepted by the receivers’ web network.
In other terms: The IP telephony is taking the phone conversations and routing them through an IP system across a cable into the network – then out – and lastly into the internet connection.
To replace the classic phone line and the phone network, the whole system uses the IP and LAN (Local area network) for transmission of calls and other information over the office network, as well as the provider’s network.
Difference Between VoIP and IP Telephony
It is not completely incorrect when you consider IP and VoIP terms as synonyms. They both use a LAN system for connecting to the internet via router. However, there are some differences between IP Telephony and VoIP (Voice over IP). Main one is the type of data traffic.
IP telephony is simply a way of transforming analogue signals into digital ones. It’s a concept that stands behind all phone systems. IP can also transfer data signals for fax, messaging, computer systems, printers and more. This makes IP telephony efficient, since you need to operate only one internet network for all communication.
VoIP voice system is designed to transfer voice signals from analogue to digital and transport the signals through internet connection exclusively. It’s used for online calling and voicemail systems. VoIP is basically a transport vehicle of digital voice signals.
We can say that the VoIP system is a subcategory of IP telephony. While IP is a complete concept, VoIP is a mode, transmitting voice in order to implement this concept.
Therefore, IP telephony and VoIP work together – they offer cheap or free calls and add more features to telephone communications.
Our IP Cloud PBX is an example of the IP telephony and VoIP concept implementation.
Here is a brief insight into VoIP voice system:
6 Most Used IP Telephony Protocols
For sending voice data over the internet, you need a way to compress and decompress data, since phones stream them in real time and focus on human voices.
This is why IP telephony solutions rely on many open-source protocols that transfer data from the phone to the service provider. Which protocol IP telephony uses depends on how a phone system is set up, as well as at VoIP provider.
Here are 3 most common protocols:
Yet, in this blog, we will also introduce those that are not that widely mentioned.
SIP is an abbreviation of Session Initiation Protocol – a signaling protocol used for establishing a session between two or more participants, modifying this session and eventually, terminating it.
An advantage of SIP is that it strongly resembles HTTP. In both cases, messages are text-based and a request-response mechanism eases the troubleshooting process.
Role of the SIP messages is to describe the identity of call participants and how they can be reached over an IP network. SIP finds out the type of media channels that are going to be established for the session and how media engines will reach each other.
Once the exchange of setup messages finishes, the media is switched using another protocol. Typically, it is RTP, which we will discuss later.
SIP protocol is designed to be extensible. It is better suited for keeping up with a modern market and technical needs of the IP telephony industry.
Such as SIP, H.323 was designed for the setup, managing and terminating of a media session. It was made relatively early, which gave it an advantage – not only it defined the basic call model, it covered the supplementary services that were needed to address business communication expectations with relevant standards.
Though, a disadvantage of H.323 is that it’s a binary protocol, which makes it a bit more technically challenging. Its features need more time in order to be defined, standardized and implemented. It ended up almost completely replaced by SIP.
RTP, Real-time Transport Protocol, features a packet format for the transmission of audio and video across the internet. It is used mainly in entertainment and communication systems which involve streaming media – video teleconference applications, television services, web based push-to-talk features, as well as telephony.
This protocol is mostly implemented together with another one – RTCP, about which we will talk next. While RTP carries the media stream – audio and video, RTCP monitors transmission statistics and QoS (Quality of Service), and helps to synchronize multiple streams.
RTP is also used in conjunction with SIP which assists in setting up the connection across the network. Real-time Transport Protocol is certainly one of the foundations of IP telephony and VoIP systems.
It stands for Real-Time Transport Control Protocol. As we explained above, it works hand in hand with RTP.
While RTP delivers the actual data, RTCP sends control packets to participants in a call. Its primary function is to give feedback on the quality of service that is provided by RTP.
RTPC therefore transports information and statistics. Software can use this data to control QoS parameters. Real-Time Transport Control Protocol, as well as RTP, doesn’t provide any flow encryption or authentication methods, yet these mechanisms may be implemented by the next protocol – SRTP.
The meaning behind SRTP is a Secure Real-time Transport Protocol. It was published in 2004, as an extension profile of RTP. It adds more security features, like message authentication, confidentiality and replay protection, mostly intended for IP telephony / VoIP communication.
SRTP uses security methods – authentication and encryption – for minimizing risks of attacks. It implements AES (Advanced Encryption Standard), and default encryption cipher. SRTP features are optional, you can individually enable or disable them.
This protocol is flexible and easily accommodatable for new encryption algorithms.
Session Description Protocol is a standard of defining the multimedia communication sessions type for session announcements and invitations. It’s commonly used in streaming video conferences and VoIP applications.
In a session setup process, there are two endpoints participating. Each of them sends an SDP to inform the other of its specifications and capabilities.
Therefore, SDP by itself doesn’t deliver any media. It simply limits itself to the negotiation of a compatible set of media exchange parameters. The media streams are handled by a different protocol.
In other words – SDP protocol is a declaration, by a media endpoint, of its receiving specification and capabilities.
A typical declaration would tell us:
- which IP Address is prepared to receive the incoming media stream
- which port number is listening for the incoming media stream
- what media type the endpoint is expecting to receive (typically audio)
- which protocol the endpoint is expecting to exchange information in (typically RTP)
- which compression encoding the endpoint is capable of decoding (codec)
5 Benefits of IP Telephony
There is no doubt that IP telephony, therefore also VoIP, is taking over the telecommunication market. More than 79 % of United States businesses are using VoIP phones for at least one location.
Let’s take a look at the most significant advantages of IP telephony.
#1 It’s Cheaper Than Landlines
As mentioned, by using IP telephony technology instead of traditional phone systems, companies can use only internet connection to make phone calls with customers.
This fact assists companies in lowering operational spendings and phone call fees. In opposition to analogue solutions, the cost of IP telephony in business is consistent and easily predictable. Switching to a VoIP plan can save you up to 60 % costs, compared to a business landline plan.
Only charges are from the internet provider. There is no need to pay to the telephone companies, as you no longer need analogue lines. Also, long distance and international calls are less expensive with IP telephony solutions.
#2 It’s Easy to Integrate it with CRM
IP telephony means an advance in telecommunications systems, as it allows for the convergence of various systems into one. By leveraging IP telephony, you can integrate it with an existing infrastructure – combining data from your CRM, e-commerce, helpdesk and accounting solutions in a single unified system – call center software.
The possibility to centralize data provides more effective, consistent customer service that can be used across multiple channels. That boosts the ability of your agents to get information faster, and consequently increases employee productivity.
IP telephony also improves means of communication with your clients. You can reach them exactly where they are, therefore improve customer satisfaction.
#3 It’s Scalable and Simple
Unlike with traditional analogue phone systems, IP telephony makes scaling your line truly easy. Since IP has much less restrictions, it allows full flexibility.
You can quickly and easily add new lines or remove unnecessary ones, which enables you to scale up and down however you need. There is also no need to pay separately for incorporating other phone lines.
The deploying of new lines doesn’t require any special technical skills. It can be done easily via an online interface. No programmer needed, since IP telephony doesn’t have a complexity of proprietary software or graphical user interface.
#4 It Allows for Remote Work
People can easily get used to doing their job from home, assuming they have the right technology. IP telephony gives you a seamless solution on how to work fully remotely, since there is no need for hardware. Therefore, there is nothing that ties your agents to one location.
By choosing an effective cloud-based calling software, such as CloudTalk, they can be wherever in the world while you can monitor them via features like record phone calls or monitor agents’ performance.
#5 It Offers Advanced Features
It’s clear that hardware-based call centers have their limitations. Basically, they can only dial and pick up phone calls. In contrast, businesses that are using IP telephony systems can do much more than just calling.
IP is revolutionizing the way of communication with clients and customers, due to a rich portfolio of advanced features.
To mention some, you can make use of an advanced IVR tool with wide possibilities of customization. Workflow automation allows you to streamline business processes by automating unnecessary tasks.
You can also easily follow your call center statistics, track contact history of your clients or create customized tags and notes right during phone calls. Automatic callback ensures that you never miss a call. Skill-based routing can route the phone call to agents who have the best overview in a specific topic.
In CloudTalk, we offer all of these features and more. Use one of the most advanced call center software on the market, and make the most of your IP telephony experience.