Did you know many of us, use VoIP on daily basis? Surprised! Yes, every time we use our phones, tabs, laptop, or pc to make calls using Skype, Facebook Messenger, Whatsapp, or Teamviewer, we are using VoIP applications. Voice over Internet Protocol is a technology that transports voice communication and multimedia sessions over an IP network. RTP or Real-Time Transport Protocol helps in structuring the data packets to deliver at incredible speed across the internet followed by reassembling into an easy flow stream that is suited for delivering voice and multimedia calls naturally. It serves as a basis for voice-over IP and without RTP, VoIP cannot be imagined. Thus RTP is utilized in interactive audio and video conferencing. However, if you want to learn in detail what is the purpose of RTP in VOIP, we suggest do not skip any part of the upcoming sections in the article.
Practically, RTP is based on a multitude of protocols. It uses UDP (User Datagram Protocol) protocol in IP architecture. However, using UDP protocol for summarizing the RTP packets includes constraints at the error correction level. Consequently, it discards the lost or damaged packets. RTP protocol supports the series and combination of audio and video instead of fairness of the transported data.
An important function of RTP is assuring a steady way to transfer data subjected to real-time constraints. It inserts time markers and sequence numbers to audio-visual and other multimedia streams. It also controls the destination position for the arrival of RTP packets and also detects the type of information that has to be transferred. RTP cannot reserve network resources, provide network reliability, or ensure delivery time. RTP lays the foundation for VoIP and often clubs together with SIP (Session Initiation Protocol) for connecting across networks.
The evolution of VoIP Telephony
During the early times from the 1880s till the 1960s, all the calls traveled along a single wire or series of wires in form of a smooth continuous electrical signal. If you had to call from one city to the other, you required a manual operator who personally connects you to the concerned city through a wiring route by rearranging the jack plug. Later the physical connections were replaced with digitally switched exchanges with the presence of new transistors.
With time advancement and the advent of the internet, data has now started being distributed into series of packets loaded with instructions that are important for switching and bandwidth allows all the packets to pave their way through internet traffic congestion. Since these packets travel really fast, it is not easy to navigate through internet traffic and as a result, the packages don’t arrive in the similar order and fashion as they were sent. All these packets must be marked at their appropriate place on the stream for easy reassembling.
What is Real-Time Transport Protocol (RTP) in Telecom?
RTP is an internet protocol framework used for exchanging audio and video over the internet on IP-based formats in real-time. Therefore, it is featured in IP-based telephone systems. Its first standardization dates back to 1996 in RFC 1889 by the IETF’s Audio-Video Transport Working group. Real-Time Transport Protocol is primarily used in internet telephony software. RTP is dependent on network characteristics hence unable to ensure real-time multimedia data delivery but provide all the resources to control data for its arrival in the best form.
RTP couples with control protocol (RTCP) to control data for multicast communication and maintain the QoS (Quality of Service) parameters during data transfer.
The architecture of RTP is composed of 3 key components- the synchronization source which represents an actual source of data; a translator that pushes ahead of the RTP packets and if required tweaks coding of the data to be transferred; and a mixer combines the data streams of different sources and advance them to a new data stream. The composition of RTP can further be explained as consisting of the following:
- Sequence number- It is used to identify the packets that are lost during the transfer.
- Payload recognition- It describes the media encoding to modify it to adjust with the internet variations.
- Frame indication- It marks the beginning and end of the frame
- Source identification- It is used to identify the source of the frame.
- Intermedia synchronization- It organizes timestamps to identify and compensate the delay jitters within data streams.
At the application layer, the RTP packet is constructed which is then transferred to the transport layer for delivery. So RTP packet data is the initiation of every part constructed by the application even a small fragment of RTP media data. It also informs the receiver about the information reconstruction and packeting of codec bitstreams. The least size of any RTP header is 12 bytes.
The header can be divided as follows:
- Version- shows the protocol version
- Padding(p)- shows the presence of extra padding bytes
- Extension(X)- signifies that an extension header is present between the header and payload data
- CSRC count(CC)- consists of total CSRC identifiers present
- Marker(M)- signifies the deployment in a profile specific way at the application stage
- Payload type (PT)- showing the payload format and its interpretation by the application
- The Sequence number- for every RTP packet is increased. The receiver uses it to identify loss of data and respond to order delivery
- The Timestamp- this is installed by the receiver to replay the received sample at the right time and interval
- SSRC- this synchronization source determines the source of the stream.
- CSRC- it acts as a contributing source of RTP packets that contributes towards the merged stream produced by the mixer
- Header Extension- It is indicated by the extension field and is optional.
The RTP header also includes different information about the data format and transport.
By now, you would have found the answer to your question – what is the purpose of RTP in VoIP? We can easily understand as VoIP utilizes Real-Time Transport Protocol to send real-time streams of data across networks for easy audio-visual transmission through IP networks. RTP is usually combined with other different protocols such as RTCP, UDP, and SIP to connect a voice or video call or fax across the internet. These combinations even encounter internet traffic, adjusting them to signal delays and prevent any delay jitter that can degrade the QoS during voice or video conferencing. So it is not wrong to say we have come far away since the 1880s!