Understanding RTP and RTCP Protocols
Translated by
ScribdTranslationsUnderstanding RTP and RTCP Protocols
Translated by
ScribdTranslationsThe RTP/RTCP protocols
September 2015
1. Introduction : Qu'est ce que RTP et RTCP ?
2. RTP (Real-time Transfer Protocol)
3. RTCP (Real-time Transfer Control Protocol)
4. Planned use of RTP and RTCP
5. Format of headers and their content
6. The RTCP header
7. How is RTCP used in relation to RTP?
8. Over which protocols do RTP and RTCP operate
9. How is the type of flow conveyed?
Introduction: What are RTP and RTCP?
The spread of computers, combined with the availability of inexpensive audio/video hardware, as well
that the availability of higher speed connections has sparked interest in using the Internet to
send audio and video, types of data that were traditionally reserved for networks
specialized for this purpose, and for several years now audio and videoconferencing have become a
common practice. But the very nature of the Internet means that this network is not suitable for the
real-time data transmission, this results in the quality of the audio sent to
through the Internet has a generally poor quality. This thesis specifically addresses the analysis and
solution to these problems to enable an audio conferencing or phone application on the Internet,
to adapt one's behavior to maintain an acceptable auditory quality even in cases where the
the network is quite congested. These solutions, in the form of control mechanisms, have been
implemented and tested on the Free Phone Internet audio conferencing and phone software that we
have developed. A study on the behavior that these mechanisms would have in an Internet that
evolved to incorporate the Fair Queueing service discipline showed that these mechanisms, which would be
still necessary, would even have better performance in this type of network.
RTP (Real-time Transfer Protocol)
The purpose of RTP is to provide a uniform means of transmitting data subjected to IP.
real-time constraints (audio, video, ... ). The main role of RTP is to implement
IP packet sequence numbers to reconstruct voice or video information even if the network
Underlying changes the order of the packets.
More generally, RTP allows:
to identify the type of information being transported,
to add time markers and sequence numbers to the information carried
to control the arrival at destination of the packages.
In addition, RTP can be carried by multicast packets to route conversations to
multiple recipients.
RTCP (Real-time Transfer Control Protocol)
The RTCP protocol is based on periodic transmissions of control packets by all the
participants in the session.
C'est un protocole de contrôle des flux RTP, permettant de véhiculer des informations basiques sur les
participants in a session, and on the quality of service
Intended use of RTP and RTCP
RTP allows for the management of multimedia streams (voice, video) over IP. RTP operates on UDP. The RTP header
contains synchronization and numbering information. The data encoding will depend on the
type of compression. The RFCxxxx specifies RTP, on the other hand, the adaptation of a compression method to
RTP will be described in a specific RFC, for example H261 on RTP is described in RFCxxxx. A
RTP channel is used by stream type: one for audio, one for video. The xxx field is used for
synchronization. RTP offers an end-to-end service. It adds a header that provides the information of
timing necessary for the synchronization of real-time streams such as audio and video. RTP (Realtime Transport
Protocol) and its companion RTCP (Realtime Transport Control Protocol) allow respectively to
transport and control data streams that have real-time properties. RTP and RTCP are
protocols that are at the application level and use the underlying TCP transport protocols
or UDP. But the use of RTP/RTCP is generally done over UDP. RTP and RTCP can
use both Unicast mode (point to point) and Multicast mode (multipoint). Each of them uses a
a separate port of a pair of ports. RTP uses the even port and RTCP uses the immediately higher odd port.
Format of headings and their contents
The RTP header will contain the following information:
32 bits
V=2 P X CC M Sequence number
Timestamp
Synchronization Source Identifier (SSRC)
Contributor Source Identifiers (CSRC)
Here is the meaning of the different fields of the header:
The Version field Vde of length 2 bits indicates the protocol version (V=2)
The padding field P: 1 bit, if P equals 1, the packet contains additional padding bytes.
(padding) to finish the last packet.
The extension field X: 1 bit, if X=1 the header is followed by an extension packet
The CSRC count field CC: 4 bits, contains the number of CSRC that follow the header.
The marker field M: 1 bit, its interpretation is defined by an application profile.
The payload type field PT: 7 bits, this field identifies the type of payload (audio, video, image, text,
html, etc.)
The sequence number field: 16 bits, its initial value is random and it increments by 1 each time.
package sent, it can be used to detect lost packets
The timestamp field: 32 bits, reflects the moment when the first byte of the RTP packet was sampled.
This moment must be derived from a clock that increases monotonically and linearly over time.
to enable synchronization and the calculation of jitter at the destination
The SSRC field: 32 bits, uniquely identifies the source, its value is chosen in such a way
random by the application. The SSRC field identifies the synchronization source (or simply says 'the
source). This identifier is chosen randomly with the interest of being unique among all the
sources of the same session The list of CSRC identifies the sources (SSRC) that contributed to
the acquisition of the data contained in the packet that contains these identifiers. The number
Identifiers are provided in the CC field.
The CSRC field: 32 bits, identifies the contributing sources.
The RTCP header
The objective of RTCP is to provide different types of information and feedback regarding the quality of reception.
The RTCP header will contain the following information:
The field version (2 bits)
The padding field (1 bit) indicates that there is padding whose size is indicated in the last byte.
The reception report count field (5 bits): number of reports in the packet
The packet type field (8 bits) 200 for SR
The field length (16 bits) length of the packet in 32-bit words
The SSRC field (32 bits): identification of the source specific to the sender
The NTP timestamp field (64 bits)
The RTP timestamp field (32 bits)
The field sender's packet count (32 bits)
The sender's octet count field (32 bits) statistics
The SSRC-n field (32 bits) is the source number of the analyzed stream.
The fraction lost field (8 bits)
The field cumulative number of packets lost (24 bits)
The extended highest sequence number received field (32 bits)
The interarrival jitter field (32 bits). It is an estimate of the time interval of a packet
RTP data that is measured with the timestamp and is in the form of an integer. It is actually the time.
transit relative between two data packets.
The formula to calculate it is: J=J+(|D(i-1,i)|-J)/16 The interarrival jitter is calculated with each packet of
data received by source SSRC_n i --> First packet i-1 --> previous packet D --> difference J -->
Second package
The field last SR timestamp (32 bits)
The field delay since last SR (32 bits)
How is RTCP used in relation to RTP?
RTCP is a control protocol associated with RTP, it measures performance, but it does not offer any
guarantee. For this, it is necessary to use a reservation protocol of the RSVP type or to ensure that the
The communication links used are properly sized in relation to their usage.
Above which protocols do RTP and RTCP operate
RTP/RTCP is above the UDP/TCP transport, but practically above UDP.
RTP is a session protocol, but it is placed in the application. It is up to the developer to integrate it.
How is the type of flow conveyed?
RTP has nothing to do with the type of stream, it is above UDP which is above IP. The type of stream
is theoretically used in IP.
RTP provides a sequence number, a timestamp, and a unique source identifier (SSRC).
Article written by Nico VanHaute, Julien Barascud, and Jean-Roland Conca.
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