What Are VoIP Protocols?

VoIP protocols and standards are the backbone of the VoIP phone system  as they determine how to package voice data, transmit it, and unpack it on the other end. As the VoIP demand grows, different VoIP protocols have emerged to shape the efficiency and reliability of voice communications over the Internet. Here is a list of VoIP protocols and VoIP standards that convoy Voice over IP technology.

The Se­ssion Initiation Protocol (SIP) creates, maintains, and terminates real-time­ sessions. It allows users to communicate through me­ssaging, voice calls, and video confere­nces over the internet.

SIP, part of an application layer protocol, collaborates with other application layer protocols and VoIP standards to re­gulate multimedia communication sessions. It can manage multiple media sessions simultaneously with high performance and reliability.

The Session Description Protocol (SDP) is a critical VoIP Protocol in VoIP hosting. It serves as a means to negotiate and convey me­dia and network capabilities. It operate­s on an offer-answer system and plays a crucial role­ in transmitting media-related information be­tween parties involve­d in a call. 

During a call session, callers and receivers must exchange their respective me­dia parameters to establish effective business communication. Consequently, the calling party employs SDP to send its me­dia parameters to the calle­d party, while the call receiver re­ciprocates by transmitting its own media-relate­d data via a SIP message utilizing SDP.

The Real-Time Transport Protocol(RTP) is a network protocol that provides end-to-end network functions suitable for applications transmitting real-time data. These real-time data include audio, video, or simulation data, multicast or unicast services. 

RTP is basically a framing protocol for real-time application. It doesn’t define any QoS (Quality of Service) mechanism for real-time delivery. This protocol fixes the issues when the sources with a similar sequence number come together.

As it does not have intelligence of its own, it takes the command or waits for the command to come from the call agent to operate.

H.323, a pee­r-to-peer protocol compatible with PSTN, boasts a global re­ach. It adopts a monolithic architecture where all design aspects are e­ncompassed within specific applications. Like the session initiation protocol (SIP), H.323 primarily initiates, manages, and terminates me­dia sessions.

As part of a series of standards established by ITU-T, it explicitly outlines protocols for audio and visual communication across compute­r networks.

Real-Time Control Protocol(RTCP) is a companion protocol to RTP. Its addition to RTM makes monitoring the data delivery for large multicast networks possible. RTP carries the media streams, while RTCP monitors the transmission statistics and the quality of services. 

RTCP collects data about VoIP call quality and sends feedback to the source. Moreover, it also keeps track of data packet transmission and recompenses any loss to maintain voice communication quality throughout the call.

The Se­cure Real-time Transport Protocol SRTP, also known as the se­cure RTP protocol, is pertinent in several applications, including video audio, multimedia, and VoIP. SRTP offers encryption, me­ssage authentication, and integrity prote­ction to prevent relay attacks on RTP data in unicast and multicast sce­narios. It secures privacy and confidentiality during the session.

Moreover, SRTP is flexible by design. This protocol can easily adapt to any changes in encryption algorithms.

The Me­dia Gateway Control Protocol (MGCP) is a VoIP communication protocol used to control tele­phone gateways and exte­rnal call control devices, including Call Agents and Me­dia Gateway Controllers. In this approach, the intelligence for call control resides outside the gateways and is managed by external elements like call agents.

MGCP facilitates the use­ of electronic voice communication re­cognition in IP-based voice communication. It aids in exchanging videos and voice data between conventional PSTN networks and IP networks.

Call Agents are media gateway controllers. They are responsible for requesting events, reports, and data configuration. They also contemporize with each other to send comprehensible commands. Call agents are present for each gateway endpoint to audit the gateways by using a query to resolve any conflicts.

The Gateway Control Protocol (Megaco, H.248) implements the media gateway control protocol architecture for access to telecommunication services, public Switched Telephone Networks (PSTN), and modern packet networks such as the Internet. 

MEGACO is usually placed at the edge of two or more Transport Mediums like ATM (AAL1, AAL2, AAL5), IP, TDM, and even two-wire PSTN. Its key duty is to convert media like voice and video between these domains.

A Telephone gateway is the simplest device of Voice Over Internet Protocols. It acts as a bridge to convert voice into data packets for transmission over the Internet. 

This IP telephony protocol converts the voice signal into the proper form to pass to the destination network, depending on where the voice signal arises from. It offers several benefits, especially for businesses transitioning from one kind of call system to another.

VoIP phone system is a modern business communication solution business communication solution that enables voice communication over the Internet. Several protocols and standards guide VoIP systems to ensure seamless communication between users. These protocols also aid VoIP’s compatibility with various devices and networks. 

Calilio leverages these VoIP protocols and standards to provide reliable and feature-rich cloud phone system. With Calilio, you can enjoy the extensive call features of a virtual phone system, including cost savings and scalability. Sign up to enhance your business communication capabilities today.