The VoIP protocols provide guidelines and instructions that direct the software in this procedure, and thus VoIP calls can be made. In comparison with traditional landlines, a business VoIP phone system offers such advantages as custom routing, HD voice, analytics, and other advanced options.
The VoIP technology is based upon a few communication protocols which are necessary to aid in transmitting the audio data online. VoIP protocols support the functionality and structure of all processes of the VoIP process.
In this article, you will find everything you need to know about VoIP protocols, the most significant, their functionality, and the necessity.
What is a VoIP Protocol?
A VoIP protocol is a collection of regulations and specifications which determine how the software systems used by users to connect, make and sustain VoIP calls. To enable users to connect on VoIP call on the internet, their software systems have to first marshal fundamental information: identification of each party, codecs to be used, channels to be used, type of media to be transferred, etc. Next, both of them will have to actively transmit and receive multimedia data throughout the call session.
What do VoIP Protocols Do?
VoIP protocols entail order, sequence, and guidelines regarding the manner in which software systems ought to format and transmit data to create and stream a VoIP link. VoIP communication needs a number of features in order to support internet-based calling: to make a real-time session, to register participated users, define the types of media that will be transferred, to transmit and receive the media in fact, and so on. VoIP protocols instruct the software systems on the steps to take, and the messages to send to ease this process.
The VoIP single phone call consists of a number of different protocols, some functioning concurrently and others functioning in a specific sequence.
VoIP functions dealt with by protocols include:
Transport: Transport protocols create reliable end to end communication in such a way that both parties are able to transmit data in the course of the conversation. Data reception is also confirmed in these protocols, to make sure that the packets are being received by the destination.
Connection management: Connection management protocols provide a call or a session between user endpoints.
Signaling: Signaling protocols locate each endpoint, and IP address, and dial the participating parties, negotiate the codecs that the call will use, as well as call controls such as mute and transfer.
Description of media: Media description protocols are the ones that organize which kind of media is going to be transmitted in the session and that include the audio and video and other forms of data.
Media: Media protocols deal with the real-time transmission of data that is actually transferred in a VoIP call such as audio and video.
Security: Security controls are used to verify the identity of the user, maintain access control and encrypt session data.
Software developers have dozens of VoIP protocols to choose, including proprietary and open-source protocols. There are numerous protocols that have the same functionality and are replaceable- e.g., SIP has been replacing H.323 over the past ten years -and other protocols depend on each other to work.
Most Common VoIP Protocols
These are the most widespread VoIP protocols. The following are the majority of protocols used in conjunction with each other and are usually found in the VoIP platforms of the present day.
Common VoIP protocols:
i. Session Initiation Protocol (SIP)
SIP is the most widespread VoIP protocol and is a signaling protocol that creates, sustains, and ends a connection between all participants of a VoIP call. SIP determines the participants of the call and then establishes the format and sequence of messages between the participants-invites and ringing. After SIP has already placed the call between the two endpoints, the active media stream is replaced by another protocol, such as RTP.
ii. Real Time Transport Protocol (RTP)
RTP is a transport protocol that transmits audio and video media in real-time between the endpoints in a VoIP call. RTP comes in to stream audio data in the active call once SIP sets the VoIP call. Virtually all VoIP and video-conferencing systems use RTP in live media communications, such as web-based software, such as WebRTC, which are web-embedded.
iii. RTP Control Protocol (RTCP)
RTCP is used in collaboration with RTP to give quality of service (QoS) and statistics of packet-delivery to multimedia data. RTCP measures and reports data to the parties of the call on the number of packets, the number of packets lost, and the round trip delay time. This data can be used in troubleshooting to find out the data transmission bottlenecks of the phone system software, and poor connectivity.
iv. Secure Real Time Transport Protocol (SRTP)
SRTP is a security protocol that operates with RTP, parallel to the encryption of data, authentication of messages, and integrity, and replay attack prevention. SRTP is optional, although it uses RTP. Each of the features of SRTP can be switched on and off by the users.
v. Session Description Protocol(SDP)
SDP is a signaling protocol that is used in conjunction with SIP and transmits fundamental information within the call between the users. SDP transmits such information as the name and the purpose of the session, its start and finish time, the kind of media that will be involved in the session, the endpoint port numbers, the codecs being used etc.
vi. Media gateway control protocol (MGCP)
MGCP is a transport protocol that manages the media gateway on the Internet and the public-switched telephone network (PSTN). Other VoIP calls involve the use of the internet and the PSTN based on a cable, and this necessitates the use of media gateways that can bridge the gap between packet-based audio data and a circuit-based audio signal, which can be supported by the PSTN. MGCP manages such gateways on any VoIP calls and endpoints involving PSTN phone lines.
vii. H.323
Another protocol for transmitting data packets across an IP network, the predecessor of SIP is H.323, which is a system specification comprising multiple protocols that create a session to facilitate data packet transmission. H.323 has protocols that exchange registration, call signaling, and open the channel of a VoIP call to take place.
Although certain VoIP and video communications software continue to use H.323, a majority VoIP platforms have changed to SIP-which offers identical functionality with a less complex installation.
Additional or Not So Popular Protocols
The following protocols are not as widespread as the previous ones. Such protocols may have a specialized usage with reduced popularity, or be more of a supplement to VoIP calling. Others of the following protocols are old and have gradually been replaced by the protocols above.
i. XMPP and Jingle
XMPP (Extensible Messaging and Presence Protocol) is an application-layer protocol that was initially created to transport instant messaging, presence detection data, and contact list information across the Internet. This feature has been incorporated in most other forms of applications, which include VoIP, video conferencing, messaging and file transfer.
Jingle is an XMPP extension and signaling protocol, which is used to enforce the transit of instant messaging, file sharing, and other forms of structured data into VoIP and video calls. This media is prepared and delivered by Jingle and is streamed by RTP when it is used during the session.
ii. Inter-Asterisk Exchange (IAX)
IAX is a VoIP telephony protocol, SIP substitute, and offers VoIP telephony (VoIP) over the Asterisk software-based personal exchange (PBX). Whereas the majority of VoIP systems employ SIP, MGCP and RTP, IAX employs a single stream of data and port number to transmit session signaling and media.
It is a system that makes the process of cloud-based telephony easy in some aspects. Although Asterisk has been increasingly popular, and resorts to approximately 16 percent of VoIP systems, it remains less popular than SIP and the above-mentioned protocols.
iii. H.248 (Megaco)
H.248 is one component of the MGCP and it allows the media gateway controllers to communicate with media gateways, to ensure that the gateways can convert audio between the signal based PSTN and packet based IP networks.
Nevertheless, H.248 does not support the communication between media gateway controllers. It consequently relies on MGCP and other protocols in forming a full system that links endpoints in a VoIP call.
iv. H.320
H.320 recommendation consists of several protocols that facilitate the narrow-band visual telephone systems, especially the video-conferencing and video phones, to send audio and video media through the PSTN. The recommendation informs and details communication modalities, types of terminals, and call control facilities that enable videoconferencing to operate using landline.
The protocols of H.320 are only applicable to the media of ISDN based networks which supports video over cable landline, it also specifies that of nowhere narrow band audio signals-bit rates between 64 and 1920 kbit/s. H.320 is barely used these days because of the little popularity of ISDN and the fact that the current day data transmission methods are significantly faster.
v. H.324
H.324 is a recommendation that offers the standard of low-bitrate multimedia communication in the traditional analog telephone lines. Like H.320, H.324 is applicable to voice, audio and data that is carried through the landline. It provides specifications of the technical needs of low-bitrate destinations and terminals to participate in multimedia communications on the PSTN.
vi. Skinny Client Control Protocol (SCCP)
SCCP is a transport protocol, which is proprietary to Cisco, which works in the same way as MGCP, which translates media between the analog PSTN and the IP network of packets. SCCP is created to work on Cisco endpoint hardware, e.g. Cisco VoIP phones.
It is also a signaling protocol, which registers and interconnects endpoints, such as SIP. CCP is proprietary and SIP is open, which has made SIP far more popular among VoIP services.
Final Words – VoIP Protocols
To sum up, the important protocols of VoIP provide a powerful insight into the power of VoIP in changing the way business communication takes place. VoIP has unmatched convenience and cost-saving in the face of wide accessibility and low demands in terms of resources. Organizations can enjoy tremendous advantages of this technological wonder that is ever-evolving and using powerful technologies such as SIP, RTP, SDP, H.323, MGCP, and SBC.
FAQs – VoIP Protocols
Q1. What is the distinction between SIP and RTP?
VoIP calls are set up and controlled by the signaling protocol SIP and sent by the media protocol RTP.
Q2. What is the most popular VoIP used?
VoIP signaling is the most popular protocol currently SIP.
Q3. Is VoIP secure?
VoIP is safe, provided it is applied properly with the help of such protocols as SRTP or TLS to encrypt messages.
Q4. What is the bandwidth needed by VoIP?
VoIP bandwidth needs will depend on the codec employed, however, most common bandwidth requirements per call lie in the range of 30kbps to 128kbps.
Q5. What is the benefit of VoIP usage?
VoIP is cheaper, more flexible and versatile than other traditional phone systems.
Q6. What are the typical issues of VoIP?
Problems like network problems, low quality of service and security loopholes are among the problems linked with the use of VoIP.
Read More : How To Set Up VoIP For Business in 2025
