Real-time Transport Protocol

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 which was made obsolete in 2003 by RFC 3550. Real time transport protocol can also be used in conjunction with RTSP protocol which enhances the field of multimedia applications.

RTP does not have a standard TCP or UDP port on which it communicates. The only standard that it obeys is that UDP communications are done via an even port and the next higher odd port is used for RTP Control Protocol (RTCP) communications. Although there are no standards assigned, RTP is generally configured to use ports 16384-32767. RTP can carry any data with real-time characteristics, such as interactive audio and video. Call setup and tear-down for VoIP applications is usually performed by either SIP or H.323 protocols. The fact that RTP uses a dynamic port range makes it difficult for it to traverse firewalls. In order to get around this problem, it is often necessary to set up a STUN server.

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 RTCP and is built on top of the User Datagram Protocol (UDP). Applications using RTP are less sensitive to packet loss, but typically very sensitive to delays, so UDP is a better choice than TCP for such applications.

According to RFC 1889, the services provided by RTP include:

Payload-type identification - Indication of what kind of content is being carried
Sequence numbering - PDU sequence number
Time stamping - allow synchronization and jitter calculations
Delivery monitoring
The protocols themselves do not provide mechanisms to ensure timely delivery. They also do not give any Quality of Service (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 supported directly. 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.

RTP was also published by the ITU-T as H.225.0, but later removed once the IETF had a stable standards-track RFC published. It exists as an Internet Standard (STD 64) defined in RFC 3550 (which obsoletes RFC 1889). RFC 3551 (STD 65) (which obsoletes RFC 1890) defines a specific profile for Audio and Video Conferences with Minimal Control. RFC 3711 defines the Secure Real-time Transport Protocol (SRTP) profile (actually an extension to RTP Profile for Audio and Video Conferences) which can be used (optionally) to provide confidentiality, message authentication, and replay protection for audio and video streams being delivered.

Packet structure
+ Bits 0-1 2 3 4-7 8 9-15 16-31
0 Ver. P X CC M PT Sequence Number
32 Timestamp
64 SSRC identifier
96 ... CSRC identifiers ...
96+(CC×32) Extension header (optional).
96+(CC×32) + (X×((EHL+1)×32)) Data


The RTP header size is minimum 12 bytes.

Ver.
(2 bits) Indicates the version of the protocol. Current version is 2.
P
(1 bit) Used to indicate if there are extra padding bytes at the end of the RTP packet.
X
(1 bit) Indicates if the extensions to the protocol are being used in the packet.
CC
(4 bits) Contains the number of CSRC identifiers that follow the fixed header.
M
(1 bit) Used at the application level and is defined by a profile. If it is set, it means that the current data has some special relevance for the application.
PT
(7 bits) Indicates the format of the payload and determines its interpretation by the application.
SSRC
Indicates the synchronization source.
CSRC
Contributing source ID.
Extension header
Indicates the length of the extension (EHL=extension header length) in 32bit units, excluding the 32bits of the extension header.

Potential further development of RTP & RTCP
The Real-time Transport Protocol (RTP) and the Real-time Transport Control Protocol (RTCP) are commonly used together. RTP is used to transmit data (e.g. audio and video) and RTCP is used to monitor QoS. The monitoring of quality of service is very important for modern applications. In large scale applications (e.g. IPTV), there is an unacceptable delay between RTCP reports, which can cause quality of service related problems.

For more information read about problems and potential further development of RTCP

To reduce the size of the IP, UDP and RTP headers, Compressed RTP (CRTP) was developed and specified in RFC 2509. It is primarily used for reliable and fast point-to-point links, but it can be problematic in other applications. Therefore, Enhanced CRTP (ECRTP) was defined.

Especially in VoIP over wireless applications, headers are significantly larger than the payload. The RObust Header Compression (ROHC) specified in RFC 3095 seems to be an increasingly deployed method for better efficiency. There are currently two ROHC profiles defined for the compression of IP/UDP/RTP traffic. The original definition in RFC 3095, and a recently published RFC 5225.


Mathematical background
The equations for RTCP protocol are explained in section I. and II.A in the Optimization of Large-Scale RTCP Feedback Reporting in Fixed and Mobile Networks paper.

Structure of RTP/RTCP applications
RTP/RTCP protocols are commonly used to transport audio or audio/video data. Separate sessions are used for each media content (e.g. audio and video). The main advantage of this separation is to make it possible to receive only one part of the transmission, commonly audio data, which lowers the total bandwidth.