Voice over Internet Protocol is swooping businesses across the world with the promise of providing a reliable unified communications package. VoIP providers claim that they can easily outshine traditional circuit-switched networks. But is it time to put your traditional phone to the side?
It just might be. VoIP deployments use special protocols to allow its different components to work between communicating devices to enable incoming calls using internet connection. It generally makes the VoIP system adhere to international standards, ensure seamless video and voice packet transfers, as well as prevent VoIP security issues at their core.
Let’s start with a quick explanation of the main sets of ‘rules’ followed by the packet-switched networks:
Major VoIP Protocols
SIP
SIP or Session Initiation Protocol is the standard for signaling and controlling interactive VoIP communication sessions.
A session is simply a call between two or more VoIP endpoints. An endpoint is any device that can receive and send voice and multimedia content over the internet. This includes VoIP phones, mobile devices and even laptops.
VoIP device manufacturers have mandated the use of SIP as it is based on client-server architecture. It is considered a popular protocol and a better alternative to the competing H.323 protocol due to it being:
- More modular
- Better to scale
- Internet compatible
- Highly flexible
SIP is a part of the application layer responsible for creating and terminating voice calls with one or more participants or endpoints. It is designed to be independent with limited commands. SIP provides VoIP service with high reliability, performance, and scalability as it can handle the ever expanding range of multimedia sessions.
RTP
Real-time Transport Protocol is the network standard that supports the transmission of audio signals or videos in real-time during a phone call.
It enables VoIP calls because it allows real-time data transfer on IP-based networks. Once the voice is encapsulated into IP, RTP is triggered for the mixing, sequencing, and time-stamping. It works together with SIP, UDP, and RTCP.
RTP supports the different formats of files such as MPEG and MJPEG. It can code multimedia data streams like audios and videos, establishing itself as the communication link for transmitting audio and video streams for the IP telephony protocols.
RTP packets have a header that contains information related to the packet’s contents, including version and sequence numbers, a unique ID for the sender, time-stamp, and the format of the data being transmitted.
Real-time Transport Protocol is involved in resolving conflict, which happens when two sources have the same sequencing number. This is how service providers enhance synchronization across VoIP networks.
RTCP
RTCP or Real-time Transport Control Protocol is RTP’s companion, which is why it provides out-of-band statistics and information for every RTP session, helping with VoIP traffic and quality control.
It’s used for monitoring data as it sends digital signals that can control the transmission and quality of data. This instant messaging allows the recipients to send feedback to the source. RTCP also monitors packet loss and compensates for any jitter delays to improve overall voice communication.
RTCP is the feedback mechanism enabled during VoIP calls as it is often used to deliver RTP and recover streamed data. It has five types of messages, each serving a different function. These are Sender Report, Receiver Report, Source Description Message, Bye Message, and Application Specific Message.
SRTP
Secure Real-time transport protocol is mainly known as the extension profile of the RTP. It adds more transport layer security features to minimize the risk of attacks. It provides confidentiality, integrity, and replay protection of the media files.
It includes encryption, authentication, and security features for VoIP control signaling, and provides a framework for RTP and RTCP streams.
It supports origin authentication and can easily accommodate new encryption algorithms. It is guaranteed to be secured for both unicast and multicast applications.
H.323
H.323 is one of the oldest standards for VoIP and is mainly used to run video conferencing. It’s based on binary language which is a major difference between it and SIP that uses a text-based format. It describes protocols for the provision of audio-visual sessions and depends on RTP and RTCP.
H.323 allows the transfer of media data over packet networks and covers the supplementary service needed for multimedia communications. One of its strongest advantages is its interoperability, allowing solutions produced by different companies to connect and operate with each other seamlessly. H.323 is based on four components to deliver multimedia services. These are Terminals, Gateways, Gatekeepers, and Multiple Control Units.
What Other Protocols and Standards Should I Know?
Telephone Gateway
When utilizing VoIP, you’ll need interfaces to convert telephony traffic and enable both radio and telephone users to communicate with each other. It converts the IP voice traffic from the node to the signaling format to communicate and translate voice streams used for traditional phone lines. Telephone Gateway basically converts the voice signal into the proper format for the receiving device.
MGCP
Media Gateway Control Protocol handles and controls communication protocols applied in VoIP. It creates a centralized gateway administration and only executes commands from the gateways. MGCP controls telephone gateways from call agents and media gateway controllers. It signals each port in the gateway to establish connections with IP Telephone gateways.
Call Agent
Call Agents are also known as Media Gateway Controllers and control the media gateways, which is a simplified definition of all they do. These agents are responsible for requesting events, reports, and configuration of data, and they also synchronize with each other to send comprehensible commands.
Different call agents are present for each endpoint on the gateway. They can audit gateways by using a query to resolve any conflicts.
H.248 or MEGACO
H.248 implements MGCP to provide VoIP telecommunication services across a converged internetwork. It provides complete and scalable controllers to control media gateways to support multimedia streams across IP networks. This is why it’s considered a complementary of the H.323 and the SIP.
SCCP
Skinny Client Control Protocol is a Cisco proprietary signaling protocol for providing:
- Extended routing
- Flow control
- Segmentation
- Connection orientation
- Error correction facilities
SCCP is easy to understand as it’s lightweight with a defined set of messages to be observed and implemented. This is why it is designed to exchange signaling messages between client and server during calls, as it has much easier syntax and requires less processing power.
SDP
Session Description Protocol is a format for describing multimedia communication sessions and WebSocket transports. It is also used for the initiation and announcement of sessions. SDP is a short structured textual description that conveys the name, purpose of the session, the media protocols, and codec formats to help participants join or gather info about a particular call.
Final Words
SIP and H.323 are currently the most widely used protocols. However, there’s a growing number of alternatives, and the future looks very promising for modern VoIP communication. Keep up with the voip-info.org to ensure you are up to date with the evolution of different protocols and VoIP telephony in general.
April 7, 2023
Phone System 