08Jun

SIP Vs VoIP: Functionalities, Similarities and Differences

June 8, 2021 Sagar Malam 62

Enterprises around the world are fast moving from traditional phone lines to cloud-based internet telephony. This is because cloud technologies support more cost-effective and hardware-independent communication options.

Internet telephony has also become a go-to choice for many businesses due to its increased reliability and accessibility – and VoIP and SIP are providing the foundations for IP technology.

However, in this new age of telephone technologies, voice represents only part of a modern organization’s communication needs. Starting from the 1990s, VoIP (Voice over Internet Protocol) has gained attention and traction, and become one of the most successful voice communication tools across the world. VoIP has enjoyed soaring popularity as it can seamlessly support a growing global mobile workforce. Tech watchers predict it may overtake traditional public switched telephone networks in the coming years.

But VoIP isn’t the only cloud-based technology that businesses must consider to cater to their unified communication needs. SIP (Session Initiation Protocol) is also an important contributor to cloud-based business communication.

Although both VoIP and SIP refer to internet telephony, they are technically very different. While VoIP is an Internet-powered alternative to landline phone calls and is a voice-oriented technology, SIP is a multimedia communications protocol and an industry-standard method used to enable VoIP calls between devices.

In this blog, we will look at how VoIP and SIP function independently of each other and also how they complement each other and provide the best results when they are used together.

What is VoIP?
What is SIP?
SIP Vs VoIP: What are the Differences
How SIP and VoIP Work Together
What works best: VoIP, SIP, or VoIP with SIP
VoIP or SIP: What’s the Right Fit for your Business

What is VoIP?

Unlike traditional phone lines or public switched telephone networks (PSTN), VoIP refers to any telephone communication over the Internet. VoIP is also referred to as IP telephony, internet telephony, or voice over broadband. This technology uses the internet as a channel to digitally transmit voice signals to another telephone or endpoint. VoIP is deployed in conjunction with mobile or browser-based applications, through an Internet Telephony Service Provider (ITSP).

For basic voice functions, enterprises need a VoIP-enabled phone and VoIP software, to make and receive calls over the internet. It’s turned into a popular choice for organizations that want to remove the clutter of landline phones in their office. With a VoIP setup, they get to retain an official line for their business.

What is SIP?

While VoIP refers to a type of phone call, SIP refers to the protocol used for setting up these calls. SIP is a communication protocol that enables VoIP and transmits VoIP calls, by governing the establishment, termination, and other essential elements of a phone call.

SIP can transmit information between two or more endpoints. With SIP, enterprises are not limited to voice communication only and can expand into video conferencing, instant messaging, media distribution, and other applications. Most IP phones that you will find in organizations are SIP-compatible.

Besides being fast, flexible, and scalable to meet specific business demands, one of the main benefits of SIP over other VoIP protocols is its ability to interact intelligently with other protocols. This enables it to adapt its signaling protocol to match that of the client system.

SIP Vs VoIP: What are the Differences

Comparing VoIP to SIP is akin to comparing apples with oranges. While VoIP describes any internet-based telecommunication system, SIP is a specific type of VoIP deployment.

VoIP refers to any call made over the internet and depends on data connectivity to transmit voice information packets, rather than PSTN. On the other hand, SIP is a protocol used to initiate, maintain, and terminate multimedia system communication sessions in VoIP applications. SIP protocols enable the management of voice, video, and electronic messaging applications.

Let us look at some key differences between the two technologies:

VoIP best supports voice communications between single parties. On the other hand, SIP supports a broader set of applications, including multimedia conversations between multiple parties, including voice, video, and instant messaging. SIP systems allow users to make voice and video calls online, most often for free.
A major difference between VoIP and SIP is their scope. VoIP is not a discrete technology, but a family of technologies that are used within modern telecommunication networks. SIP is a signaling protocol used within the umbrella of VoIP technologies.
VoIP handsets and SIP handsets vary in terms of devices – although a SIP handset can also function as a VoIP handset. VoIP handsets, or VoIP phones, must be connected to a computer to make or receive calls. SIP phones work without a computer – requiring only a modem. This makes SIP phones more like traditional phones.
VoIP hosts handle traffic differently from SIP systems. SIP systems are processed by individual systems that are located on every user’s system. This peer-to-peer system can handle large amounts of data and traffic as its users have access to more bandwidth and resources. On the other hand, VoIP hosts direct traffic from one central network. This can result in a system overload when there is a large amount of traffic, impacting its overall performance. However, a single network system also has its benefits – as fewer points of communication result in stronger security.

How SIP and VoIP Work Together

The relationship between VoIP and SIP is similar to websites and HTML. A website is a collection of pages that can be accessed using a web browser. HTML is a language used to build websites. However, websites can also be created using other languages like Java.

Likewise, VoIP is a method used for sending voice calls over the internet – covering any type of voice communication ranging from Skype and FaceTime to enterprise phone services. SIP is one of the many protocols that can be used to implement VoIP. Most phone systems used by organizations today rely on SIP, as it is considered an industry-standard today.

Despite their fundamental differences, VoIP and SIP overlap in many ways. Softphones, also called SIP telephones as they use SIP technology, and they connect devices to VoIP networks. Softphones integrate using a computer or a cloud-based phone system, making phone numbers independent of any single device. This is their most distinctive feature when compared to traditional landlines. However, VoIP adapters allow phones that were traditionally used with landlines to incorporate the functions of softphones as well.

When SIP and VoIP come together, they offer several advantages to enterprises:

VoIP and SIP allow organizations to consolidate technologies via unified communications (UC). Communication methods that support SIP can integrate with other business cloud applications, such as customer relationship management (CRM) systems. This way, users can share data via SIP-supported platforms.
As part of an organization’s UC solution, VoIP and SIP help consolidate administrative tasks associated with video, voice, and instant messaging.
SIP plays a big role in supporting VoIP and other unified communications across the globe. In 2019, SIP trunking technology held the largest share of the global $15 billion VoIP market. With the VoIP market estimated to double by 2025, SIP will get a big share of the pie.

What works best: VoIP, SIP, or VoIP with SIP

VoIP and SIP may work well together, but they’re not mutually inclusive. If you’re using only voice communications then VoIP doesn’t need SIP to function. There are many internet protocols that enable voice communications, though SIP is the most widely used.

Besides voice, SIP also enables text and video communication. Any enterprise on the lookout for an all-around UC provider will go for VoIP with SIP. Using only VoIP can limit their options, while SIP allows them to scale their communication as per their business needs. Also, as SIP can interact with other protocols, it means your VoIP won’t develop a snag while trying to connect with other systems.

More and more organizations are finding that, sooner or later, they have to upgrade to using VoIP with SIP – it’s emerging as the most popular option. Using this tech combo, they can integrate different types of multimedia channels into a single system, to transmit voice, video, and conference calls, and create seamless UC suites.

A big reason for SIP going mainstream is that it supports diverse communication capabilities. It takes VoIP’s capabilities and adds the power of video and data transfer to it. SIP has you covered if you:

– Need a network that provides seamless long-distance video conferencing.

– Need a convenient way to send files and documents to a co-worker.

SIP builds on a VoIP system’s strengths, helping it to boost productivity. It’s also easy to implement and manage, as long as your IT department is familiar with the protocol.

The benefit of using VoIP-only services is that it has portable components and easy-to-manage payments. If an enterprise requires only voice communication support, they might find it more cost-effective to use a protocol other than SIP with their VoIP system. However, this can limit their future communication growth.

SIP is a protocol that is compatible with cloud applications and provides more growth opportunities. Industries are increasingly adopting advanced technologies to stay competitive and to scale up quickly. As organizations seek better unified communication strategies, they are turning to SIP services to provide this opportunity.

In a final analysis, the VoIP Vs SIP debate is more about what solution works best for you, rather than which one comes with greater advantages. If an organization is looking at consolidating its communication technology, integrating SIP with VoIP services could be a game-changer.

VoIP or SIP: What’s the Right Fit for your Business

For organizations seeking a unified communications approach, the most important question is whether voice-only VoIP phone service best serves their business needs, or if a SIP implementation – with its UC capabilities – is a better fit.

Factors that help organizations decide whether UCaaS (Unified Communication as a Service) is the right match for them include their budget and a desire to adopt multimedia communications tools. If they’re already using collaboration apps or technologies, which are a component of UC, they have a strong case for SIP adoption. For many businesses, moving directly to SIP offers cost efficiency and a boost in productivity.

So should organizations go for hosted VoIP services or SIP trunking? It depends on a number of factors, including:

Business needs of an organization

Hosted VoIP services give businesses very little control over features, customization, and security. The opposite happens with SIP – you’re in charge of everything. There are scores of VoIP services available for businesses to opt for these days. So finding a service provider who can meet their business needs is easy. But if a hosted VoIP does not offer the end-to-end communication needs of an organization, then SIP trunking is the obvious choice.

Budget

SIP trunking needs a heavy upfront investment – as it requires an IP PBX and other hardware components to be purchased – although it does prove cost-effective in the long run. Therefore, small and medium businesses may do a double-take about the large spend. Hosted VoIP services, on the other hand, are easy on the pocket and fit well with any budget.

Availability of in-house expertise

If an organization opts for hosted VoIP services, they don’t need an IT team to manage the system – everything is handled by the vendor. Professional service providers provide seamless troubleshooting and support services. SIP trunks, on the other hand, will require in-house IT expertise to deploy and maintain the phone system.

Long-term communication goals

Before zeroing on a communication technology, enterprises must answer some key questions. How do you see your business growing in the near future? What changes in communication methods do you anticipate in the long term? Hosted VoIP services can scale fast. But if you are aiming at dramatic organizational changes, then they might not be able to keep pace. Businesses that are looking at a sharp growth trajectory need better control over their phone systems – which SIP trunking can provide.

Conclusion

As modern businesses communicate using a combination of emails, text, video conferencing, and cloud-based IM platforms, they require a lot more from their communication services provider than just voice calls.

Organizations that are looking at boosting their communication strategies by moving to VoIP should consider each of the ways they can deploy it, including SIP trunking, and opt for the one that provides the greatest benefits to them. If your customers appreciate the benefits that SIP and VoIP bring, they are likely to give a go-ahead to modernize their communications platform.

21Apr

What is SIP and How Does It Work?

April 21, 2021 Maulik Shah 79

All conversations need an introduction and a reliable framework to initiate them. Session Initiation Protocol (SIP) does this job in the digital world. It’s a seamless way to begin and end online conversations.

SIP is a signaling protocol for IP-based telephony applications. It provides the control layer for communications like the establishment and release of a voice call. This communication technology does more than just connect agents to their clients. It allows users to manage and modulate every call that enters their system. What’s more, SIP is content-agnostic. Although SIP was initially designed for voice communication, today it is also used for establishing sessions for instant messaging, video conferencing, SMS, file transfers, and more.

SIP is essentially used for internet telephony and disseminating multimedia between two or more endpoints. For instance, a user can start a telephone call with another user using SIP, or initiate a conference call with multiple participants.

The SIP protocol was designed to be simple, with a limited set of commands. Since it is text-based, anyone can read a SIP message that is passed between two endpoints in a session. The protocol is also flexible and comes with great depth. It was designed to set up real-time multimedia sessions between different participant groups. For instance, apart from facilitating regular telephone calls, SIP can be used to set up video and audio multicast meetings, or instant messaging conferences.

Since SIP is an open standard, it has sparked enormous interest in the telephony market. Marketers of SIP-based phones are witnessing huge growth in this sector.

In this blog, we will get a macro as well micro view of what SIP is, how it works, how it compliments VoIP, and why businesses should invest in this technology:

What does SIP do?
SIP Vs VoIP and how SIP adds value to VoIP.
Features of SIP.
How SIP enables a VoIP session call flow.
How SIP works during a voice call.
Benefits for businesses.

What does SIP do?

SIP is not voice, not video, and not data – the protocol is not restricted to any one media. Also, while it is mostly applied to VoIP, it’s not exclusively a VoIP protocol as well.

SIP launches and ends IP communication sessions that could either be voice calls between two users or a team video conference. It initiates a session by sending messages – in the form of data packets – between two or more identified IP endpoints, also known as SIP addresses. Every SIP address is linked to a physical SIP client – like an IP desk phone – or a software client – like a softphone, for instance.

During these sessions, SIP does not decode or transfer any information. Therefore, the protocol can be used for video conferencing and instant messaging besides making internet voice calls.

SIP’s primary function is to set up calls and end them once they finish. This process involves 5 parts:

User location: This tells the protocol where the end system that will be used for the call is located.
User availability: This tells the protocol whether the party that is being called is available to take it.
User capabilities: This determines the media that will be used for the call.
Session setup: This involves connecting a call from one user to the other.
Session management: This involves the running of the call session – for example: transferring and ending a call.

SIP Vs VoIP: How they differ

Since both SIP and VoIP (Voice over Internet Protocol) are both internet-based telephone service technologies, it is common to confuse between the two. The key differentiator between the two is that while VoIP is a type of telephone service, the SIP protocol is used to commence and end communication through VoIP or any other technology.

If your business only requires voice communication, you might want to invest in technologies other than SIP with your VoIP system. However, organizations are rapidly adopting advanced technologies in order to stay competitive and scale rapidly. With its superior unified communications platform and strategies, SIP provides businesses with this opportunity.

Finally, businesses must consider what solutions work best for them. If an organization is considering consolidating its communication technologies, then integration of SIP and VoIP services would be a high-value option for them.

How SIP adds value to VoIP

A primary reason for SIP’s soaring acceptance as a go-to telephony technology is that it supports a wide range of communication platforms. Besides being easy to implement and manage, SIP builds on an enterprise’s VoIP system strengths and fills the tech gaps, allowing them to turn more productive.

SIP adds value to VoIP in various ways:

If a business needs a network system that will provide crystal clear images in long-distance video conference calls, SIP will do the job seamlessly.

If an organization is looking for an easy way to send files and documents to its employees and outside its office, SIP offers the functionality it needs.

Because of SIP’s flexibility, enterprises, as well as service providers, use it to deliver VoIP services. SIP even builds on VoIP’s capabilities by adding video and data transfer potential to it.

SIP enables network operators to gain more control over their services. For instance, not only does SIP set up the initiation of contact between devices, it also allows users to add more parties to a call, or switch between communication methods.

Features of SIP

SIP can handle a diverse range of media technologies as the protocol is not restricted to any limited media types. Let’s look at some features of this protocol:

Direct connection between private and public telephone networks:

One of SIP’s main benefits is that it provides a direct connection between private or local telephone systems and public telephone networks. This way, individuals and businesses don’t need legacy phone lines to connect calls.

Provides end-to-end authentication and encryption:

VoIP networks opt for SIP as a call control protocol because of its open and extensible properties. However, the integrity of call signaling between two sites is a priority for businesses, to counter eavesdropping concerns. SIP provides this by supporting comprehensive end-to-end and hop-by-hop authentication, as well as end-to-end encryption.

Supports both multicast and unicast communication:

Users in a SIP session can communicate using multicast or unicast relations, or even a mix of the two. SIP supports session descriptions that allow participants to accept a common set of compatible media types. It also facilitates user mobility by proxying and redirecting requests to the location where the user is at the moment. SIP is not restricted to any one conference control protocol.

Additional capabilities can be added easily:

SIP doesn’t make any presumptions about the underlying transport protocol. It can, therefore, be easily extended with additional capabilities and is actively used for many media applications. This also makes SIP an enabling technology for providing innovative new services that integrate multimedia with internet services like e-mail, instant messaging, and the worldwide web.

Asynchronous event notifications:

In addition to real-time services, SIP can be used for asynchronous event notifications, such as automatic callbacks, message-waiting indicators, SIP proxy statistics changes, presence information, session changes, buddy lists based on presence, and more.

Unlike other telephony systems, SIP is non-numeric:

Telephony is usually associated with numeric addresses. In SIP, an end station has a SIP URI (which is a form of URL) that recognizes it and is used in the SIP protocol. Because phones generally have numeric keypads, they translate what you dial into a SIP URL.

SIP is text-based:

The SIP Protocol is text-based and has a notable resemblance to the HTTP protocol. The text-based messages and SIP’s request-response mechanism make for easy troubleshooting. Also, it is easy for anyone to read and decipher the text-based SIP messages passed between the endpoints in a session.

SIP leaves fewer margins for error:

Because SIP isn’t affected by external factors, there are lesser things that can go wrong with it. While traditional phone lines can be affected by bad weather, SIP trunks use the internet and can have back-ups set up to ensure that even if your internet connection goes, you aren’t offline for too long.

Big cost benefits:

SIP works out to be much cheaper than traditional phone lines. Small to large businesses that switch to SIP-enabled VoIP phones reduce the cost of their local calls and save on international calls as well. Once the initial setup is done, the only thing that businesses pay for is the Wi-Fi that you use to place these calls and your unlimited SIP trunk subscription.

How SIP enables a VoIP session call flow

Now that we have understood the dynamics of how SIP works, let’s look at how a SIP call flow establishes a VoIP call. There are four parts that together establish a call:

Registration:

When a user agent – let’s suppose it’s a softphone – launches, it has to register with a SIP server so it can be found by other users. This happens by using the SIP Register request message. The message provides the location details through the To and From SIP URIs. Besides the message, additional binding can be provided through the contact field.

Establishing a call:

This is where the real work happens – in the following steps:

SIP invite request: The SIP invite initiates an attempt to establish a call. This is sent from the caller to the SIP server where it finds the call receiver’s number. In larger networks, the SIP server may need to consult other SIP servers if this user number is not local. Once the number is located, the invite is forwarded.

SIP response 100: A message is sent from the SIP server to the number of the end-user in order to confirm the invite request.

SIP Response 180: This message is a confirmation that the invite has been received and the user agent is alerting the user.

SIP Response 200: A 200 response is sent to confirm a call when it is answered by the end-user. Also, the user agent sends a message with the VoIP call parameters. This message results in an exchange and negotiation of VoIP call parameters.

SIP Ack Request: In the final lap, the caller confirms the call by sending an Ack request to the number he has called.

VoIP Call:

The VoIP call is transmitted between the user agents using RTP (real-time transport protocol), which is also used for sending audio and video data over Internet Protocol networks. Besides this protocol, RTCP (RTP control protocol) is used to provide statistics and control for the RTP transmissions.

Terminating a call:

When a user ends a call, a SIP Bye request is sent. The call can be ended from either side of the line. The termination is confirmed by the other user agent with a SIP 200 status code response.

How SIP works during a voice call

Before voice information is sent over the internet, it needs to be encoded in codecs. This enables audio signals to be translated into data. There are two primary types of codecs that are used: G.711 and G.729. Let’s take a step-by-step look at how SIP connects voice calls:

Real-time transport protocol (RTP) is used to transport packets of encoded audio data. RTP comes with its own protocol application layer which facilitates the movement of audio and video data in real-time.
In the next step, the RTP and SIP packets are transported by protocols at the transport layer. This is done either by using a Transmission Control Protocol or a User Datagram Protocol. User Datagram Protocols are considered to be better suited for transporting VoIP calls.
In the final step, as SIP is a media-independent application, another application layer protocol – known as the Session Description Protocol (SDP) – works with SIP. SDP indicates the type of media that the SIP client sending the call and the client receiving the call can support.

Benefits of SIP for Businesses

Enterprises today are trying to figure out how to support a host of new unified communications applications in an era of new devices, mobile connections, and sophisticated end-users. In such a scenario, how does SIP benefit organizations?

Customer engagement is a top priority for sales agents, and this is where SIP stands out the most. Using a text-based medium to transcribe messages, this VoIP function enables both agents and clients to run through the messages exchanged during every session, and makes it smoother for both developers and network managers to find and correct mistakes.
Whether it’s through their computers or mobile phones, clients can efficiently communicate with agents through SIP. Since its appearance, SIP has played a key role in the success of internet telephony as a critical tool for call centers in providing customer service.
This VoIP function has played a significant role in helping companies become cost-efficient in their communications. You could be located anywhere in the world SIP has made it possible to make free audio or video calls.
Telecom major AT&T has confirmed that SIP “can reduce access costs, improve bandwidth utilization, and bring down operational expenses. Enterprises using SIP can save significant money by converging their voice and data networks, particularly if they are covering multiple locations.” Gartner estimates that SIP trunking will reduce telecom expenses by 50% in the US.
Since SIP has become a well-accepted telephony standard, it has helped enterprises eliminate telecom complexities, accelerate innovation and avoid vendor lock-ins. Using SIP, technology organizations can construct diverse multi-vendor networks that meet their functional and financial needs.
SIP is a valuable tool for companies that have employees working in locations across the world, and need to collaborate over long distances. As SIP facilitates fast and powerful multimedia sessions, it’s an ideal solution for such businesses.
Businesses deploy SIP to boost efficiency and increase ROI within their organizations. For example, instead of having several platforms to manage video, telephony, and other services, they can use SIP to manage them all with one interface.
SIP calling ensures that an enterprise is always available and on-call for their customers to contact them.

Conclusion

SIP has become one of the most common protocols to be used in VoIP technology, as this application layer protocol works in conjunction with other such protocols to control multimedia communication sessions over the internet. SIP is fast gaining momentum as being a protocol of choice to facilitate audio calls, video conferencing, and instant messaging in next-generation networks, to boost efficiency and cost-effectiveness for businesses.

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