Session Initiation Protocol (SIP) is a flexible protocol used in VoIP platforms to initiate and manage call sessions. SIP is considered a call initiation protocol, but it can do much more than initiate voice calls. SIP also has enhanced functions that make it powerful throughout communications environments.

SIP technology also enables SIP trunking, a service that allows on-premise PBX networks to connect to VoIP services for outbound calls rather than being restricted to traditional telephony networks. 

Read on to learn more about SIP and how it enables essential functions for VoIP communications.

In both traditional and IP telephony, network engineers distinguish between two phases of a voice call:

1. Call setup: This phase includes all the details necessary to connect two phones to each other. 
2. Data transfer: Once connected, the data transfer phase moves voice packets between the two endpoints. 

SIP is classified as a call setup protocol that operates on the application layer. SIP was designed as a general-purpose protocol that enables real-time multimedia communication. 

This vital protocol was designed for simplicity. SIP is text-based and modeled after HTTP’s response and request model. The text-based nature of the protocol enables easy debugging and reviewing of error logs. Additionally, SIP can operate on IPv4 or IPv6 and can use either TCP or UDP. In most applications, SIP uses IPv4 and UDP. 

SIP handles several functions that make it a core protocol in VoIP services. SIP needs flexibility and versatility to meet the demands of VoIP communications platforms. Engineers consider SIP to have five distinct significant functions:

1. User location and registration: Endpoints leverage SIP proxies to register locations, and SIP uses this information to determine which endpoints will participate in a call, including video calls. 
2. Session management: SIP can terminate calls, transfer calls, or change call parameters, such as adding another user.
3. Session setup: SIP is used to agree on specific session attributes used by both the called party and the calling party, including telling the called party’s endpoint that it needs to ring. 
4. User availability: Is the called party available to answer the call? SIP can tell the calling party that the user they’re calling is available or unavailable.
5. User capabilities: SIP supports multimedia communication, so both endpoints must agree on the codec that needs to be used, such as the voice codec.

These functions are used throughout most VoIP calls to enable multimedia and dynamic communication. 

VoIP and SIP may seem like competing technologies, but they are part of the same family of technologies enabling IP-based communications. VoIP is the umbrella term for a collection of protocols, including SIP. SIP is the call setup protocol that enables other VoIP protocols. 

SIP trunking is a specific service that leverages SIP technology to connect traditional PBX (Private Branch Exchange) telephony networks to VoIP services. Doing so allows for more concurrent calls, limited only by available bandwidth, and enables cost-effective long-distance and international calling. 

SIP trunking can be thought of as a virtualized version of analog phone lines. This virtualization allows on-premise PBX networks to interact with VoIP services and removes the dependence on traditional telephony services.

SIP empowers businesses with several meaningful benefits that are often attributed to VoIP, but are actually enabled by SIP. Some of these benefits include:

• Easily scalable: SIP systems are entirely Internet-based. As broadband has penetrated most of the world and fiber optic cables have replaced copper wires, SIP can scale as needed to meet the demands of large enterprises and small businesses alike. 
• Cross-device compatibility: Since SIP works over the Internet, it can operate on mobile devices, laptops, desktops, and tablets. Employees can use any device they need, and SIP will dynamically understand which device needs to ring when receiving a call.
• Baked-in redundancy for reliability: If a traditional telephony network has an outage, there generally isn’t any redundancy, and businesses need to wait for the network to undergo repairs. Conversely, SIP calling leverages geo-redundant cloud-computing technology for substantially improved reliability. This redundancy provides improved reliability for VoIP services over traditional telephony services.

SIP is a core protocol in VoIP that enables many of the advanced features that have led VoIP communications to the forefront of modern business communications.