All You Need to Know About the Differences between VoIP and SIP

　　With its increasing accessibility and cost-effectiveness, Internet telephony has become a useful technology for businesses. It is reliable and offers a host of calling and conferencing features. While both VoIP and SIP are digital communication channels, it is good for users to understand what exactly they are, how they differ, and which suits their requirements better.

　　Business communication today goes far beyond voice calls. Although phones are still critical to connect with clients, vendors and stakeholders, the workforce in any organisation also uses emails, instant messaging apps and video conferencing tools for more effective exchange of ideas and data. Such needs are increasingly driving businesses to adopt flexible IP-based transport medium and communication protocols.

　　If you are considering new phone systems and wish to know which between SIP and VoIP is a better technology for your enterprise, here are some points that will help you make a choice:

　　SIP and VoIP: Not the same ideas

　　Session Initiation Protocol (SIP) is an overlay protocol riding over an IP network which can be used in any telephony application.

　　Voice over IP (VoIP) is the name given to the technology that establishes voice conversations over an IP medium. Essentially, VoIP uses SIP as a protocol to establish the communication. This communication can be between device-to-device (CUG), app-to-app (CUG) and from app or device to any PSTN phone.

　　A critical difference between VoIP and SIP is their scope.

　　VoIP is a broad term that may be used to refer to any voice call made over the Internet instead of traditional phone lines. It needs data connectivity and not a Public Switched Telephone Network (PSTN) connection to transmit voice packets. VoIP & SIP are used interchangeably with terms such as digital telephony, IP telephony, Internet telephony, broadband telephony, voice over broadband and IP communications. All of these describe the fact that VoIP is used to transmit voice signals to another telephone.

　　SIP is a powerful communication protocol that enables calling on IP-based phone lines. It is just one of the signalling protocols used to enable VoIP. A SIP trunk is digital and can be plugged into an Internet Protocol private branch exchange (IPBX) to start communication operations.

　　Technically, SIP defines the messages sent between endpoints, and it manages the establishment, termination and other essential constituents of a call. SIP can be used to transmit data between two or multiple endpoints.

　　In addition to voice calls, SIP also enables instant messaging/online chats, video conferencing, media sharing and other web-based applications.

　　If you are connecting an IP-PBX to a PSTN provider to route PSTN calls, this connection is termed as SIP Trunk. It then depends on the law of the land, which defines the transport medium of this connection. In India, if you are connecting to a BSO, the transport medium has to be private in the form of MPLS or P2P link. If you are connecting to a VoIP provider, the transport medium can be public Internet and can only be used for outgoing international calls.

　　Devices used for communication

　　SIP and VoIP handsets are different, too, although it should be mentioned here that a SIP phone is also a VoIP phone.

　　To make or receive a voice call, you must connect your VoIP phone to a powered (active) computer with an Internet connection or an IP-PBX. In contrast, SIP phones – wireless and wired and provided by a basic service operator (BSO) or VoIP providers – can operate without a computer registering directly to provider switches and platforms. VoIP provider also uses SIP as a connectivity protocol for connecting to SIP-enabled PBXs, Routers and Edge devices like Session Border Controllers.

　　Traffic handling procedures

　　When SIP is provided by a BSO in India, the last-mile delivery is changed to Ethernet towards Customer Premises Equipment. The transport layer has to be as MPLS or P2P connection to form the network layer. The session layer is created by defining SIP parameters on either side. PSTN calls handled over this connectivity have to conform to the PSTN numbering scheme of that country. Calls handled at the BSO switch then ride on upstream private interconnects to other local, NLD and ILD providers.

　　Whereas for VOIP trunks, the transport layer can be over the Internet connection to form the network layer and SIP to create the session layer. Call standards are decided by the VOIP provider and usually ride on upstream public interconnects to other providers.

　　Choosing between VoIP and SIP for your business

　　While VoIP can be availed via any Internet telephony service, SIP is a specific protocol. As per your business priorities, you can choose VoIP in isolation (deployed through any IP technology) or VoIP with SIP.