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VOIP Service RTP Protocol

RTP Protocol
VoIP data packets live in RTP (Real-Time Transport Protocol) packets which are inside UDP-IP packets.

Firstly, VoIP doesn't use TCP because it is too heavy for real time applications, so instead a UDP (datagram) is used.

Secondly, UDP has no control over the order in which packets arrive at the destination or how long it takes them to get there (datagram concept). Both of these are very important to overall voice quality (how well you can understand what the other person is saying) and conversation quality (how easy it is to carry out a conversation). RTP solves the problem enabling the receiver to put the packets back into the correct order and not wait too long for packets that have either lost their way or are taking too long to arrive (we don't need every single voice packet, but we need a continuous flow of many of them and ordered).

The Real-time Transport Protocol (or RTP) defines a standardized packet format for delivering audio and video over the Internet. It was developed by the Audio-Video Transport Working Group of the IETF and first published in 1996 as RFC 1889.

It was originally designed as a multicast protocol, but has since been applied in many unicast applications. It is frequently used in streaming media systems (in conjunction with RTSP) as well as videoconferencing and push to talk systems (in conjunction with H.323 or SIP), making it the technical foundation of the Voice over IP industry. It goes along with the RTP Control Protocol (RTCP) and it's built on top of the User Datagram Protocol (UDP) (in OSI model).

According to RFC 1889, the services provided by RTP include:
  • Payload-type identification
  • Sequence numbering
  • Time stamping
  • Delivery monitoring
The protocols themselves do not provide mechanisms to ensure timely delivery. They also do not give any QoS guarantees. These things have to be provided by some other mechanism.

Also, out of order delivery is still possible, and flow and congestion control are not directly supported. However, the protocols do deliver the necessary data to the application to make sure it can put the received packets in the correct order. Also, RTCP provides information about reception quality which the application can use to make local adjustments. For example if a congestion is forming, the application could decide to lower the data rate.

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