RTP – Real-time Transport Protocol

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Presentation transcript:

RTP – Real-time Transport Protocol Presentation by: Richard Bourgault and Steven Campbell

RTP - Introduction “RTP represents a new style of protocol following the principles of application level framing and integrated layer processing proposed by Clark and Tennenhouse.” Read: RTP is intended to be continuously reconfigurable often being integrated into the application itself. RTP is intended to be tailored through modifications or additions to the headers as needed.

Goals of RTP To provide end-to-end delivery services for data with real-time characteristics. Primarily designed to satisfy the needs of multi- participant multimedia conferences (using multicast distribution if provided by the underlying network ). Services include payload type identification, sequence numbering, timestamping and delivery monitoring.

RTP - Defined RTP consists of two closely-linked parts: RTP- transport protocol that has real-time properties. RTCP – control protocol to monitor the quality of service (QoS) and to convey session information.

Defined - Continued Does not guarantee timely service, quality of service, or sequenced delivery RTP is carried on top of IP and UDP. What are the advantages of this?

Answer To make use of its multiplexing and checksum services

Question? Since RTP is a real-time protocol that can lie on top of IP or UDP, what are some applications that can make use of these underlying protocols?

Answer Streaming audio, video, video-conferencing, and storage of continuous simulation data. Also active badge and control and measurement applications. Any application relying on continuous data without requiring sequential packeting.

Example of Usage – Multicast Audio Conference Host obtains a multicast group address and pair of ports: - One port for audio - One for control (RTCP) packets This address and port information is distributed to the intended participants Each participant sends data in small packets, where each piece of data is preceded by an RTP header RTP header and data are encapsulated in a UDP packet.

Each header indicates what type of audio encoding (such as PCM, ADPCM or LPC) is present in the data. To cope with loss/reorder delays, the RTP header contains timing information and a sequence number that allows the receivers to reconstruct the timing produced by the source. Why is it beneficial to include the type of audio encoding in each packet header?

Answer In order to allow for different users to use different types of encoding during the same conference. Allows for both low and high bandwidth encoding

A/V Conferencing Similar to simple audio with the exception that audio and video are treated as different RTP sessions How does this work?

Simultaneous RTP Sessions A participant may be involved in multiple RTP sessions at the same time Distinguishes between RTP sessions by using different pairs of destination transport addresses One pair of transport addresses is comprised of a network address and a pair of ports for RTP and RTCP

Simultaneous - Continued Participants included in an RTP session consist of those able to receive a global SSRC identifier This identifier can be transmitted by any of the participants either in RTP as the SSRC or in RTCP SSRC is carried in the RTP header

RTP Header 32 Bits Long 2 1 1 4 1 7 16

RTP Header Version (V) – 2 bits. Defines RTP Version. Padding (P) – 1 bit. Defines number of padding octets that are not part of the payload. Extension (X) – 1 bit. If set, the header must be followed by exactly one header extension. CSRC Count (CC) – 4 bits. Defines the number of CSRC identifiers following the header. Marker (M) – 1 bit. Specifies significant events such as frame boundaries.

Payload Type (PT) – 7 bits. Sequence Number – 16 bits. Timestamp – 32 bits. Reflects the sampling instant of the first RTP packet. SSRC (Synchronization Source) – 32 bits. Randomly specified identifier. CSRC (Contributing Source) – 32 bits. Allows up to 16 different sources.

Question? What might happen if more than 16 sources attempt to become part of the RTP session? (i.e. how can each source be uniquely identified?) Compared to the various TCP flavors what are the advantages/disadvantages of RTP?