![ The control protocol RTCP is used to specify quality of service (QoS) feedback
and synchronization between the media streams. The bandwidth of RTCP traffic compared to
RTP is small, typically around 5%.
An optional signaling protocol such as H.323, Session Initiation Protocol (SIP), or Jingle (XMPP)
An optional media description protocol such as Session Description Protocol
Sessions
An RTP Session is established for each multimedia stream. A session consists of an IP address with
a pair of ports for RTP and RTCP. For example, audio and video streams will have separate RTP
sessions, enabling a receiver to deselect a particular stream. The ports which form a session are
negotiated using other protocols such as RTSP (using SDP in the setup method) and SIP. According
to the specification, an RTP port should be even and the RTCP port is the next higher odd port
number.RTP and RTCP typically use unprivileged UDP ports (1024 to 65535) but may use other
transport protocols (most notably, SCTP and DCCP) as well, as the protocol design is transport
independent.
Packet header
RTP packet header
Bit offset[b]
0-1 2 3 4-7 8 9-15 16-31
0 Version P X CC M PT Sequence Number
32 Timestamp
64 SSRC identifier
96
CSRC identifiers
...
96+32×CC Profile-specific extension header ID Extension header length
128+32×CC
Extension header
...
The RTP header has a minimum size of 12 bytes. After the header, optional header extensions may
be present. This is followed by the RTP payload, the format of which is determined by the particular
class of application.[19]
The fields in the header are as follows:](https://image.slidesharecdn.com/assignmentondatanetwork-150322235353-conversion-gate01/85/Assignment-on-data-network-5-320.jpg)
Assignment on data network
- 1. Assignment on SCTP, RTP and RTCP Protocol Submitted by Abhishek Kesharwani College: United Institute of Technology Course: M.Tech Branch: (CSE)
- 2. Stream Control Transmission Protocol In computer networking, the Stream Control Transmission Protocol (SCTP) is a transport layer protocol (protocol number 132 ), serving in a similar role to the popular protocols Transmission Control Protocol (TCP) and User Datagram Protocol(UDP). It provides some of the same service features of both: it is message-oriented like UDP and ensures reliable, in-sequence transport of messages with congestion control like TCP. The protocol was defined by the IETF Signaling Transport (SIGTRAN) working group in 2000, and is maintained by the IETF Transport Area (TSVWG) working group. RFC 4960 defines the protocol. Message-based multi-streaming SCTP applications submit their data to be transmitted in messages (groups of bytes) to the SCTP transport layer. SCTP places messages and control information into separate chunks (data chunks and control chunks), each identified by a chunk header. The protocol can fragment a message into a number of data chunks, but each data chunk contains data from only one user message. SCTP bundles the chunks into SCTP packets. The SCTP packet, which is submitted to the Internet Protocol, consists of a packet header, SCTP control chunks (when necessary), followed by SCTP data chunks (when available). SCTP may be characterized as message-oriented, meaning it transports a sequence of messages (each being a group of bytes), rather than transporting an unbroken stream of bytes as does TCP. As in UDP, in SCTP a sender sends a message in one operation, and that exact message is passed to the receiving application process in one operation. In contrast, TCP is a stream-oriented protocol, transporting streams of bytes reliably and in order. However TCP does not allow the receiver to know how many times the sender application called on the TCP transport passing it groups of bytes to be sent out. At the sender, TCP simply appends more bytes to a queue of bytes waiting to go out over the network, rather than having to keep a queue of individual separate outbound messages which must be preserved as such. The term multi-streaming refers to the capability of SCTP to transmit several independent streams of chunks in parallel, for example transmitting web page images together with the web page text. In essence, it involves bundling several connections into a single SCTP association, operating on messages (or chunks) rather than bytes. TCP preserves byte order in the stream by assigning a sequence number to each packet. SCTP, on the other hand, assigns a sequence number to each message sent in a stream. This allows independent ordering of messages in different streams. However, message ordering is optional in SCTP; a receiving application may choose to process messages in the order of receipt instead of the order they were sent. Features Features of SCTP include: Multihoming support in which one or both endpoints of a connection can consist of more than one IP address, enabling transparent fail-over between redundant network paths. Delivery of chunks within independent streams eliminates unnecessary head-of-line blocking, as opposed to TCP byte-stream delivery. Path selection and monitoring to select a primary data transmission path and test the connectivity of the transmission path. Validation and acknowledgment mechanisms protect against flooding attacks and provide notification of duplicated or missing data chunks.
- 3. Improved error detection suitable for Ethernet jumbo frames. Association TCP has provided the primary means to transfer data reliably across the Internet. However, TCP has imposed limitations on several applications. From RFC 4960: TCP provides both reliable data transfer and strict order-of-transmission delivery of data. Some applications need reliable transfer without sequence maintenance, while others would be satisfied with partial ordering of the data. In both of these cases, the head-of-line blocking offered by TCP causes unnecessary delay. The stream-oriented nature of TCP is often an inconvenience. Applications must add their own record marking to delineate their messages, and must make explicit use of the push facility to ensure that a complete message is transferred in a reasonable time. The limited scope of TCP sockets complicates the task of providing highly-available data transfer capability using multi-homed hosts. TCP is relatively vulnerable to denial-of-service attacks, such as SYN attacks. Packet structure Bits 0–7 8–15 16–23 24–31 +0 Source port Destination port 32 Verification tag 64 Checksum 96 Chunk 1 type Chunk 1 flags Chunk 1 length 128 Chunk 1 data … … … Chunk N type Chunk N flags Chunk N length … Chunk N data SCTP packets have a simpler basic structure than TCP packets. Each consists of two basic sections: 1. The common header, which occupies the first 12 bytes and is highlighted in blue, and 2. The data chunks, which occupy the remaining portion of the packet. The first chunk is highlighted in green, and the last of Nchunks (Chunk N) is highlighted in red. Each chunk has a type identifier that is one byte long yielding, at most, 255 different chunk types. RFC 4960 defines a list of chunk types and there are currently 15 types defined. The remainder of the chunk is a two byte length (maximum size of 65,535 bytes) and the data. If the chunk does not form a multiple of 4 bytes (i.e., the length is not a multiple of 4) then it is implicitly padded with zeros which are not included in the chunk length. Security Although encryption was not part of the original SCTP design, SCTP was designed with features for improved security, such as 4-way handshake (compared to TCP 3-way handshake) to protect against SYN flooding attacks, and large "cookies" for association verification and authenticity.
- 4. Reliability was also a key part of the security design of SCTP. Multihoming enables an association to stay open even when some routes and interfaces are down. This is of particular importance for SIGTRAN as it carries SS7 over an IP network using SCTP, and requires strong resilience during link outages to maintain telecommunication service even when enduring network anomalies. SCTP is sometimes a good fingerprinting candidate. Some operating systems ship with SCTP support enabled, and, as it is not as well known as TCP or UDP, it is sometimes overlooked in firewall and intrusion detection configurations, thus often permitting probing traffic. Real-time Transport Protocol The Real-time Transport Protocol (RTP, sometimes referred to as RTTP) defines a standardized packet format for delivering audio and video over IP networks. RTP is used extensively in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications, television services and web-based push-to-talk features. RTP is used in conjunction with the RTP Control Protocol (RTCP). While RTP carries the media streams (e.g., audio and video), RTCP is used to monitor transmission statistics and quality of service (QoS) and aids synchronization of multiple streams. RTP is one of the technical foundations of Voice over IP and in this context is often used in conjunction with a signaling protocol such as the Session Initiation Protocol (SIP) which assists in setting up connections across the network. RTP was developed by the Audio-Video Transport Working Group of the Internet Engineering Task Force (IETF) and first published in 1996 as RFC 1889, superseded by RFC 3550 in 2003. Overview RTP is designed for end-to-end, real-time, transfer of stream data. The protocol provides facilities for jitter compensation and detection of out of sequence arrival in data, which are common during transmissions on an IP network. RTP allows data transfer to multiple destinations through IP multicast. RTP is regarded as the primary standard for audio/video transport in IP networks and is used with an associated profile and payload format. Real-time multimedia streaming applications require timely delivery of information and can tolerate some packet loss to achieve this goal. For example, loss of a packet in audio application may result in loss of a fraction of a second of audio data, which can be made unnoticeable with suitable error concealment algorithms The Transmission Control Protocol (TCP), although standardized for RTP use is not normally used in RTP applications because TCP favors reliability over timeliness. Instead the majority of the RTP implementations are built on the User Datagram Protocol (UDP). Other transport protocols specifically designed for multimedia sessions are SCTP and DCCP, although, as of 2010, they are not in widespread use. RTP was developed by the Audio/Video Transport working group of the IETF standards organization. RTP is used in conjunction with other protocols such as H.323 and RTSP The RTP standard defines a pair of protocols, RTP and RTCP. RTP is used for transfer of multimedia data, and the RTCP is used to periodically send control information and QoS parameters. Protocol components The RTP specification describes two sub-protocols: The data transfer protocol, RTP, which deals with the transfer of real-time data. Information provided by this protocol includes timestamps (for synchronization), sequence numbers (for packet loss and reordering detection) and the payload format which indicates the encoded format of the data.
- 5. The control protocol RTCP is used to specify quality of service (QoS) feedback and synchronization between the media streams. The bandwidth of RTCP traffic compared to RTP is small, typically around 5%. An optional signaling protocol such as H.323, Session Initiation Protocol (SIP), or Jingle (XMPP) An optional media description protocol such as Session Description Protocol Sessions An RTP Session is established for each multimedia stream. A session consists of an IP address with a pair of ports for RTP and RTCP. For example, audio and video streams will have separate RTP sessions, enabling a receiver to deselect a particular stream. The ports which form a session are negotiated using other protocols such as RTSP (using SDP in the setup method) and SIP. According to the specification, an RTP port should be even and the RTCP port is the next higher odd port number.RTP and RTCP typically use unprivileged UDP ports (1024 to 65535) but may use other transport protocols (most notably, SCTP and DCCP) as well, as the protocol design is transport independent. Packet header RTP packet header Bit offset[b] 0-1 2 3 4-7 8 9-15 16-31 0 Version P X CC M PT Sequence Number 32 Timestamp 64 SSRC identifier 96 CSRC identifiers ... 96+32×CC Profile-specific extension header ID Extension header length 128+32×CC Extension header ... The RTP header has a minimum size of 12 bytes. After the header, optional header extensions may be present. This is followed by the RTP payload, the format of which is determined by the particular class of application.[19] The fields in the header are as follows:
- 6. Version: (2 bits) Indicates the version of the protocol. Current version is 2. P (Padding): (1 bit) Used to indicate if there are extra padding bytes at the end of the RTP packet. A padding might be used to fill up a block of certain size, for example as required by an encryption algorithm. The last byte of the padding contains the number of padding bytes that were added (including itself). X (Extension): (1 bit) Indicates presence of an Extension header between standard header and payload data. This is application or profile specific. CC (CSRC Count): (4 bits) Contains the number of CSRC identifiers (defined below) that follow the fixed header. M (Marker): (1 bit) Used at the application level and defined by a profile. If it is set, it means that the current data has some special relevance for the application. PT (Payload Type): (7 bits) Indicates the format of the payload and determines its interpretation by the application. This is specified by an RTP profile. For example, see RTP Profile for audio and video conferences with minimal control (RFC 3551). Sequence Number: (16 bits) The sequence number is incremented by one for each RTP data packet sent and is to be used by the receiver to detect packet loss and to restore packet sequence. The RTP does not specify any action on packet loss; it is left to the application to take appropriate action. For example, video applications may play the last known frame in place of the missing frame. According to RFC 3550, the initial value of the sequence number should be random to make known-plaintext attacks on encryption more difficult. RTP provides no guarantee of delivery, but the presence of sequence numbers makes it possible to detect missing packets. Timestamp: (32 bits) Used to enable the receiver to play back the received samples at appropriate intervals. When several media streams are present, the timestamps are independent in each stream, and may not be relied upon for media synchronization. The granularity of the timing is application specific. For example, an audio application that samples data once every 125 µs (8 kHz, a common sample rate in digital telephony) would use that value as its clock resolution. The clock granularity is one of the details that is specified in the RTP profile for an application. SSRC: (32 bits) Synchronization source identifier uniquely identifies the source of a stream. The synchronization sources within the same RTP session will be unique. CSRC: (32 bits each) Contributing source IDs enumerate contributing sources to a stream which has been generated from multiple sources. Extension header: (optional) the first 32-bit word contains a profile-specific identifier (16 bits) and a length specifies (16 bits) that indicates the length of the extension (EHL=extension header length) in 32-bit units, excluding the 32 bits of the extension header. RTP-based systems A complete network based system includes other protocols and standards in conjunction with RTP. Protocols such as SIP, Jingle, RTSP, H.225 and H.245 are used for session initiation, control and termination. Other standards, such as H.264, MPEG and H.263, are used to encode the payload data as specified via RTP Profile. An RTP sender captures the multimedia data, then encodes frames and transmits it as RTP packets with appropriate timestamps and increasing sequence numbers. Depending on the RTP Profile in use, the sender may set the Payload Type field. The RTP receiver captures the RTP packets, detects missing packets, and may reorder packets. It decodes the frames according to the payload format and presents the stream to its user.
- 7. (RTCP)RTP Control Protocol The RTP Control Protocol (RTCP) is a sister protocol of the Real-time Transport Protocol (RTP). Its basic functionality and packet structure is defined in RFC 3550. RTCP provides out-of-band statistics and control information for an RTP session. It partners with RTP in the delivery and packaging of multimedia data, but does not transport any media streams itself. The primary function of RTCP is to provide feedback on the quality of service (QoS) in media distribution by periodically sending statistics information to participants in a streaming multimedia session. RTCP transports statistics for a media connection and information such as transmitted octet and packet counts, lost packet counts, jitter, and round-trip delay time. An application may use this information to control quality of service parameters, perhaps by limiting flow, or using a different codec. Protocol functions Typically RTP will be sent on an even-numbered UDP port, with RTCP messages being sent over the next higher odd-numbered port. RTCP itself does not provide any flow encryption or authentication methods. Such mechanisms may be implemented, for example, with the Secure Real-time Transport Protocol (SRTP). RTCP provides three basic functions expected to be implemented in all RTP sessions: The primary function of RTCP is to gather statistics on quality aspects of the media distribution during a session and transmit this data to the session media source and other session participants. Such information may be used by the source for adaptive media encoding (codec) and detection of transmission faults. If the session is carried over a multicast network, this permits non-intrusive session quality monitoring. RTCP provides canonical end-point identifiers (CNAME) to all session participants. Although a source identifier (SSRC) of an RTP stream is expected to be unique, the instantaneous binding of source identifiers to end-points may change during a session. The CNAME establishes unique identification of end-points across an application instance (multiple use of media tools) and for third-party monitoring. RTCP reports are expected to be sent by all participants, even in a multicast session which may involve thousands of recipients. Such traffic will increase proportionally with the number of participants. Thus, to avoid network congestion, the protocol must include session bandwidth management. This is achieved by dynamically controlling the frequency of report transmissions. RTCP bandwidth usage should generally not exceed 5% of total session bandwidth. Furthermore, 25% of the RTCP bandwidth should be reserved to media sources at all times, so
- 8. that in large conferences new participants can receive the CNAME identifiers of the senders without excessive delay. A fourth, optional feature, is the provisioning of session control functions, because RTCP is a convenient means to reach all session participants, whereas RTP itself is not. RTP is only transmitted by a media source. Message types RTCP distinguishes several types of packets: sender report, receiver report, source, description, and bye. In addition, the protocol is extensible and allows application-specific RTCP packets. A standards-based extension of RTCP is the Extended Report packet type introduced by RFC 3611 Sender report (SR) The sender report is sent periodically by the active senders in a conference to report transmission and reception statistics for all RTP packets sent during the interval. The sender report includes an absolute timestamp, which is the number of seconds elapsed since midnight on January 1, 1900. The absolute timestamp allows the receiver to synchronize RTP messages. It is particularly important when both audio and video are transmitted simultaneously, because audio and video streams use independent relative timestamps. Receiver report (RR) The receiver report is for passive participants, those that do not send RTP packets. The report informs the sender and other receivers about the quality of service. Source description (SDES) The Source Description message is used to send the CNAME item to session participants. It may also be used to provide additional information such as the name, e-mail address, telephone number, and address of the owner or controller of the source. End of participation (BYE) A source sends a BYE message to shut down a stream. It allows an end-point to announce that it is leaving the conference. Although other sources can detect the absence of a source, this message is a direct announcement. It is also useful to a media mixer. Application-specific message (APP) The application-specific message provides a mechanism to design application-specific extensions to the RTCP protocol.