RFCs in HTML Format


RFC 1889

          RTP: A Transport Protocol for Real-Time Applications


Table of Contents

   1.         Introduction ........................................    3
   2.         RTP Use Scenarios ...................................    5
   2.1        Simple Multicast Audio Conference ...................    5
   2.2        Audio and Video Conference ..........................    6
   2.3        Mixers and Translators ..............................    6
   3.         Definitions .........................................    7
   4.         Byte Order, Alignment, and Time Format ..............    9
   5.         RTP Data Transfer Protocol ..........................   10
   5.1        RTP Fixed Header Fields .............................   10
   5.2        Multiplexing RTP Sessions ...........................   13



Schulzrinne, et al          Standards Track                     [Page 1]

RFC 1889 RTP January 1996 5.3 Profile-Specific Modifications to the RTP Header..... 14 5.3.1 RTP Header Extension ................................ 14 6. RTP Control Protocol -- RTCP ........................ 15 6.1 RTCP Packet Format .................................. 17 6.2 RTCP Transmission Interval .......................... 19 6.2.1 Maintaining the number of session members ........... 21 6.2.2 Allocation of source description bandwidth .......... 21 6.3 Sender and Receiver Reports ......................... 22 6.3.1 SR: Sender report RTCP packet ....................... 23 6.3.2 RR: Receiver report RTCP packet ..................... 28 6.3.3 Extending the sender and receiver reports ........... 29 6.3.4 Analyzing sender and receiver reports ............... 29 6.4 SDES: Source description RTCP packet ................ 31 6.4.1 CNAME: Canonical end-point identifier SDES item ..... 32 6.4.2 NAME: User name SDES item ........................... 34 6.4.3 EMAIL: Electronic mail address SDES item ............ 34 6.4.4 PHONE: Phone number SDES item ....................... 34 6.4.5 LOC: Geographic user location SDES item ............. 35 6.4.6 TOOL: Application or tool name SDES item ............ 35 6.4.7 NOTE: Notice/status SDES item ....................... 35 6.4.8 PRIV: Private extensions SDES item .................. 36 6.5 BYE: Goodbye RTCP packet ............................ 37 6.6 APP: Application-defined RTCP packet ................ 38 7. RTP Translators and Mixers .......................... 39 7.1 General Description ................................. 39 7.2 RTCP Processing in Translators ...................... 41 7.3 RTCP Processing in Mixers ........................... 43 7.4 Cascaded Mixers ..................................... 44 8. SSRC Identifier Allocation and Use .................. 44 8.1 Probability of Collision ............................ 44 8.2 Collision Resolution and Loop Detection ............. 45 9. Security ............................................ 49 9.1 Confidentiality ..................................... 49 9.2 Authentication and Message Integrity ................ 50 10. RTP over Network and Transport Protocols ............ 51 11. Summary of Protocol Constants ....................... 51 11.1 RTCP packet types ................................... 52 11.2 SDES types .......................................... 52 12. RTP Profiles and Payload Format Specifications ...... 53 A. Algorithms .......................................... 56 A.1 RTP Data Header Validity Checks ..................... 59 A.2 RTCP Header Validity Checks ......................... 63 A.3 Determining the Number of RTP Packets Expected and Lost ................................................ 63 A.4 Generating SDES RTCP Packets ........................ 64 A.5 Parsing RTCP SDES Packets ........................... 65 A.6 Generating a Random 32-bit Identifier ............... 66 A.7 Computing the RTCP Transmission Interval ............ 68 Schulzrinne, et al Standards Track [Page 2]
RFC 1889 RTP January 1996 A.8 Estimating the Interarrival Jitter .................. 71 B. Security Considerations ............................. 72 C. Addresses of Authors ................................ 72 D. Bibliography ........................................ 73 1. Introduction This memorandum specifies the real-time transport protocol (RTP), which provides end-to-end delivery services for data with real-time characteristics, such as interactive audio and video. Those services include payload type identification, sequence numbering, timestamping and delivery monitoring. Applications typically run RTP on top of UDP to make use of its multiplexing and checksum services; both protocols contribute parts of the transport protocol functionality. However, RTP may be used with other suitable underlying network or transport protocols (see Section 10). RTP supports data transfer to multiple destinations using multicast distribution if provided by the underlying network. Note that RTP itself does not provide any mechanism to ensure timely delivery or provide other quality-of-service guarantees, but relies on lower-layer services to do so. It does not guarantee delivery or prevent out-of-order delivery, nor does it assume that the underlying network is reliable and delivers packets in sequence. The sequence numbers included in RTP allow the receiver to reconstruct the sender's packet sequence, but sequence numbers might also be used to determine the proper location of a packet, for example in video decoding, without necessarily decoding packets in sequence. While RTP is primarily designed to satisfy the needs of multi- participant multimedia conferences, it is not limited to that particular application. Storage of continuous data, interactive distributed simulation, active badge, and control and measurement applications may also find RTP applicable. This document defines RTP, consisting of two closely-linked parts: o the real-time transport protocol (RTP), to carry data that has real-time properties. o the RTP control protocol (RTCP), to monitor the quality of service and to convey information about the participants in an on-going session. The latter aspect of RTCP may be sufficient for "loosely controlled" sessions, i.e., where there is no explicit membership control and set-up, but it is not necessarily intended to support all of an application's control communication requirements. This functionality may be fully or partially subsumed by a separate session control protocol, Schulzrinne, et al Standards Track [Page 3]
RFC 1889 RTP January 1996 which is beyond the scope of this document. RTP represents a new style of protocol following the principles of application level framing and integrated layer processing proposed by Clark and Tennenhouse [1]. That is, RTP is intended to be malleable to provide the information required by a particular application and will often be integrated into the application processing rather than being implemented as a separate layer. RTP is a protocol framework that is deliberately not complete. This document specifies those functions expected to be common across all the applications for which RTP would be appropriate. Unlike conventional protocols in which additional functions might be accommodated by making the protocol more general or by adding an option mechanism that would require parsing, RTP is intended to be tailored through modifications and/or additions to the headers as needed. Examples are given in Sections 5.3 and 6.3.3. Therefore, in addition to this document, a complete specification of RTP for a particular application will require one or more companion documents (see Section 12): o a profile specification document, which defines a set of payload type codes and their mapping to payload formats (e.g., media encodings). A profile may also define extensions or modifications to RTP that are specific to a particular class of applications. Typically an application will operate under only one profile. A profile for audio and video data may be found in the companion RFC TBD. o payload format specification documents, which define how a particular payload, such as an audio or video encoding, is to be carried in RTP. A discussion of real-time services and algorithms for their implementation as well as background discussion on some of the RTP design decisions can be found in [2]. Several RTP applications, both experimental and commercial, have already been implemented from draft specifications. These applications include audio and video tools along with diagnostic tools such as traffic monitors. Users of these tools number in the thousands. However, the current Internet cannot yet support the full potential demand for real-time services. High-bandwidth services using RTP, such as video, can potentially seriously degrade the quality of service of other network services. Thus, implementors should take appropriate precautions to limit accidental bandwidth usage. Application documentation should clearly outline the limitations and possible operational impact of high-bandwidth real- Schulzrinne, et al Standards Track [Page 4]
RFC 1889 RTP January 1996 time services on the Internet and other network services. 2. RTP Use Scenarios The following sections describe some aspects of the use of RTP. The examples were chosen to illustrate the basic operation of applications using RTP, not to limit what RTP may be used for. In these examples, RTP is carried on top of IP and UDP, and follows the conventions established by the profile for audio and video specified in the companion Internet-Draft draft-ietf-avt-profile 2.1 Simple Multicast Audio Conference A working group of the IETF meets to discuss the latest protocol draft, using the IP multicast services of the Internet for voice communications. Through some allocation mechanism the working group chair obtains a multicast group address and pair of ports. One port is used for audio data, and the other is used for control (RTCP) packets. This address and port information is distributed to the intended participants. If privacy is desired, the data and control packets may be encrypted as specified in Section 9.1, in which case an encryption key must also be generated and distributed. The exact details of these allocation and distribution mechanisms are beyond the scope of RTP. The audio conferencing application used by each conference participant sends audio data in small chunks of, say, 20 ms duration. Each chunk of audio data is preceded by an RTP header; RTP header and data are in turn contained in a UDP packet. The RTP header indicates what type of audio encoding (such as PCM, ADPCM or LPC) is contained in each packet so that senders can change the encoding during a conference, for example, to accommodate a new participant that is connected through a low-bandwidth link or react to indications of network congestion. The Internet, like other packet networks, occasionally loses and reorders packets and delays them by variable amounts of time. To cope with these impairments, the RTP header contains timing information and a sequence number that allow the receivers to reconstruct the timing produced by the source, so that in this example, chunks of audio are contiguously played out the speaker every 20 ms. This timing reconstruction is performed separately for each source of RTP packets in the conference. The sequence number can also be used by the receiver to estimate how many packets are being lost. Since members of the working group join and leave during the conference, it is useful to know who is participating at any moment and how well they are receiving the audio data. For that purpose, Schulzrinne, et al Standards Track [Page 5]
RFC 1889 RTP January 1996 each instance of the audio application in the conference periodically multicasts a reception report plus the name of its user on the RTCP (control) port. The reception report indicates how well the current speaker is being received and may be used to control adaptive encodings. In addition to the user name, other identifying information may also be included subject to control bandwidth limits. A site sends the RTCP BYE packet (Section 6.5) when it leaves the conference. 2.2 Audio and Video Conference If both audio and video media are used in a conference, they are transmitted as separate RTP sessions RTCP packets are transmitted for each medium using two different UDP port pairs and/or multicast addresses. There is no direct coupling at the RTP level between the audio and video sessions, except that a user participating in both sessions should use the same distinguished (canonical) name in the RTCP packets for both so that the sessions can be associated. One motivation for this separation is to allow some participants in the conference to receive only one medium if they choose. Further explanation is given in Section 5.2. Despite the separation, synchronized playback of a source's audio and video can be achieved using timing information carried in the RTCP packets for both sessions. 2.3 Mixers and Translators So far, we have assumed that all sites want to receive media data in the same format. However, this may not always be appropriate. Consider the case where participants in one area are connected through a low-speed link to the majority of the conference participants who enjoy high-speed network access. Instead of forcing everyone to use a lower-bandwidth, reduced-quality audio encoding, an RTP-level relay called a mixer may be placed near the low-bandwidth area. This mixer resynchronizes incoming audio packets to reconstruct the constant 20 ms spacing generated by the sender, mixes these reconstructed audio streams into a single stream, translates the audio encoding to a lower-bandwidth one and forwards the lower- bandwidth packet stream across the low-speed link. These packets might be unicast to a single recipient or multicast on a different address to multiple recipients. The RTP header includes a means for mixers to identify the sources that contributed to a mixed packet so that correct talker indication can be provided at the receivers. Some of the intended participants in the audio conference may be connected with high bandwidth links but might not be directly reachable via IP multicast. For example, they might be behind an Schulzrinne, et al Standards Track [Page 6]
RFC 1889 RTP January 1996 application-level firewall that will not let any IP packets pass. For these sites, mixing may not be necessary, in which case another type of RTP-level relay called a translator may be used. Two translators are installed, one on either side of the firewall, with the outside one funneling all multicast packets received through a secure connection to the translator inside the firewall. The translator inside the firewall sends them again as multicast packets to a multicast group restricted to the site's internal network. Mixers and translators may be designed for a variety of purposes. An example is a video mixer that scales the images of individual people in separate video streams and composites them into one video stream to simulate a group scene. Other examples of translation include the connection of a group of hosts speaking only IP/UDP to a group of hosts that understand only ST-II, or the packet-by-packet encoding translation of video streams from individual sources without resynchronization or mixing. Details of the operation of mixers and translators are given in Section 7. 3. Definitions RTP payload: The data transported by RTP in a packet, for example audio samples or compressed video data. The payload format and interpretation are beyond the scope of this document. RTP packet: A data packet consisting of the fixed RTP header, a possibly empty list of contributing sources (see below), and the payload data. Some underlying protocols may require an encapsulation of the RTP packet to be defined. Typically one packet of the underlying protocol contains a single RTP packet, but several RTP packets may be contained if permitted by the encapsulation method (see Section 10). RTCP packet: A control packet consisting of a fixed header part similar to that of RTP data packets, followed by structured elements that vary depending upon the RTCP packet type. The formats are defined in Section 6. Typically, multiple RTCP packets are sent together as a compound RTCP packet in a single packet of the underlying protocol; this is enabled by the length field in the fixed header of each RTCP packet. Port: The "abstraction that transport protocols use to distinguish among multiple destinations within a given host computer. TCP/IP protocols identify ports using small positive integers." [3] The transport selectors (TSEL) used by the OSI transport layer are equivalent to ports. RTP depends upon the lower-layer protocol to provide some mechanism such as ports to multiplex the RTP and RTCP packets of a session. Schulzrinne, et al Standards Track [Page 7]
RFC 1889 RTP January 1996 Transport address: The combination of a network address and port that identifies a transport-level endpoint, for example an IP address and a UDP port. Packets are transmitted from a source transport address to a destination transport address. RTP session: The association among a set of participants communicating with RTP. For each participant, the session is defined by a particular pair of destination transport addresses (one network address plus a port pair for RTP and RTCP). The destination transport address pair may be common for all participants, as in the case of IP multicast, or may be different for each, as in the case of individual unicast network addresses plus a common port pair. In a multimedia session, each medium is carried in a separate RTP session with its own RTCP packets. The multiple RTP sessions are distinguished by different port number pairs and/or different multicast addresses. Synchronization source (SSRC): The source of a stream of RTP packets, identified by a 32-bit numeric SSRC identifier carried in the RTP header so as not to be dependent upon the network address. All packets from a synchronization source form part of the same timing and sequence number space, so a receiver groups packets by synchronization source for playback. Examples of synchronization sources include the sender of a stream of packets derived from a signal source such as a microphone or a camera, or an RTP mixer (see below). A synchronization source may change its data format, e.g., audio encoding, over time. The SSRC identifier is a randomly chosen value meant to be globally unique within a particular RTP session (see Section 8). A participant need not use the same SSRC identifier for all the RTP sessions in a multimedia session; the binding of the SSRC identifiers is provided through RTCP (see Section 6.4.1). If a participant generates multiple streams in one RTP session, for example from separate video cameras, each must be identified as a different SSRC. Contributing source (CSRC): A source of a stream of RTP packets that has contributed to the combined stream produced by an RTP mixer (see below). The mixer inserts a list of the SSRC identifiers of the sources that contributed to the generation of a particular packet into the RTP header of that packet. This list is called the CSRC list. An example application is audio conferencing where a mixer indicates all the talkers whose speech was combined to produce the outgoing packet, allowing the receiver to indicate the current talker, even though all the audio packets contain the same SSRC identifier (that of the mixer). Schulzrinne, et al Standards Track [Page 8]
RFC 1889 RTP January 1996 End system: An application that generates the content to be sent in RTP packets and/or consumes the content of received RTP packets. An end system can act as one or more synchronization sources in a particular RTP session, but typically only one. Mixer: An intermediate system that receives RTP packets from one or more sources, possibly changes the data format, combines the packets in some manner and then forwards a new RTP packet. Since the timing among multiple input sources will not generally be synchronized, the mixer will make timing adjustments among the streams and generate its own timing for the combined stream. Thus, all data packets originating from a mixer will be identified as having the mixer as their synchronization source. Translator: An intermediate system that forwards RTP packets with their synchronization source identifier intact. Examples of translators include devices that convert encodings without mixing, replicators from multicast to unicast, and application- level filters in firewalls. Monitor: An application that receives RTCP packets sent by participants in an RTP session, in particular the reception reports, and estimates the current quality of service for distribution monitoring, fault diagnosis and long-term statistics. The monitor function is likely to be built into the application(s) participating in the session, but may also be a separate application that does not otherwise participate and does not send or receive the RTP data packets. These are called third party monitors. Non-RTP means: Protocols and mechanisms that may be needed in addition to RTP to provide a usable service. In particular, for multimedia conferences, a conference control application may distribute multicast addresses and keys for encryption, negotiate the encryption algorithm to be used, and define dynamic mappings between RTP payload type values and the payload formats they represent for formats that do not have a predefined payload type value. For simple applications, electronic mail or a conference database may also be used. The specification of such protocols and mechanisms is outside the scope of this document. 4. Byte Order, Alignment, and Time Format All integer fields are carried in network byte order, that is, most significant byte (octet) first. This byte order is commonly known as big-endian. The transmission order is described in detail in [4]. Unless otherwise noted, numeric constants are in decimal (base 10). Schulzrinne, et al Standards Track [Page 9]
RFC 1889 RTP January 1996 All header data is aligned to its natural length, i.e., 16-bit fields are aligned on even offsets, 32-bit fields are aligned at offsets divisible by four, etc. Octets designated as padding have the value zero. Wallclock time (absolute time) is represented using the timestamp format of the Network Time Protocol (NTP), which is in seconds relative to 0h UTC on 1 January 1900 [5]. The full resolution NTP timestamp is a 64-bit unsigned fixed-point number with the integer part in the first 32 bits and the fractional part in the last 32 bits. In some fields where a more compact representation is appropriate, only the middle 32 bits are used; that is, the low 16 bits of the integer part and the high 16 bits of the fractional part. The high 16 bits of the integer part must be determined independently. 5. RTP Data Transfer Protocol 5.1 RTP Fixed Header Fields The RTP header has the following format: 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |V=2|P|X| CC |M| PT | sequence number | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | timestamp | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | synchronization source (SSRC) identifier | +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ | contributing source (CSRC) identifiers | | .... | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ The first twelve octets are present in every RTP packet, while the list of CSRC identifiers is present only when inserted by a mixer. The fields have the following meaning: version (V): 2 bits This field identifies the version of RTP. The version defined by this specification is two (2). (The value 1 is used by the first draft version of RTP and the value 0 is used by the protocol initially implemented in the "vat" audio tool.) padding (P): 1 bit If the padding bit is set, the packet contains one or more additional padding octets at the end which are not part of the Schulzrinne, et al Standards Track [Page 10]
RFC 1889 RTP January 1996 payload. The last octet of the padding contains a count of how many padding octets should be ignored. Padding may be needed by some encryption algorithms with fixed block sizes or for carrying several RTP packets in a lower-layer protocol data unit. extension (X): 1 bit If the extension bit is set, the fixed header is followed by exactly one header extension, with a format defined in Section 5.3.1. CSRC count (CC): 4 bits The CSRC count contains the number of CSRC identifiers that follow the fixed header. marker (M): 1 bit The interpretation of the marker is defined by a profile. It is intended to allow significant events such as frame boundaries to be marked in the packet stream. A profile may define additional marker bits or specify that there is no marker bit by changing the number of bits in the payload type field (see Section 5.3). payload type (PT): 7 bits This field identifies the format of the RTP payload and determines its interpretation by the application. A profile specifies a default static mapping of payload type codes to payload formats. Additional payload type codes may be defined dynamically through non-RTP means (see Section 3). An initial set of default mappings for audio and video is specified in the companion profile Internet-Draft draft-ietf-avt-profile, and may be extended in future editions of the Assigned Numbers RFC [6]. An RTP sender emits a single RTP payload type at any given time; this field is not intended for multiplexing separate media streams (see Section 5.2). sequence number: 16 bits The sequence number increments by one for each RTP data packet sent, and may be used by the receiver to detect packet loss and to restore packet sequence. The initial value of the sequence number is random (unpredictable) to make known-plaintext attacks on encryption more difficult, even if the source itself does not encrypt, because the packets may flow through a translator that does. Techniques for choosing unpredictable numbers are discussed in [7]. timestamp: 32 bits The timestamp reflects the sampling instant of the first octet in the RTP data packet. The sampling instant must be derived Schulzrinne, et al Standards Track [Page 11]
RFC 1889 RTP January 1996 from a clock that increments monotonically and linearly in time to allow synchronization and jitter calculations (see Section 6.3.1). The resolution of the clock must be sufficient for the desired synchronization accuracy and for measuring packet arrival jitter (one tick per video frame is typically not sufficient). The clock frequency is dependent on the format of data carried as payload and is specified statically in the profile or payload format specification that defines the format, or may be specified dynamically for payload formats defined through non-RTP means. If RTP packets are generated periodically, the nominal sampling instant as determined from the sampling clock is to be used, not a reading of the system clock. As an example, for fixed-rate audio the timestamp clock would likely increment by one for each sampling period. If an audio application reads blocks covering 160 sampling periods from the input device, the timestamp would be increased by 160 for each such block, regardless of whether the block is transmitted in a packet or dropped as silent. The initial value of the timestamp is random, as for the sequence number. Several consecutive RTP packets may have equal timestamps if they are (logically) generated at once, e.g., belong to the same video frame. Consecutive RTP packets may contain timestamps that are not monotonic if the data is not transmitted in the order it was sampled, as in the case of MPEG interpolated video frames. (The sequence numbers of the packets as transmitted will still be monotonic.) SSRC: 32 bits The SSRC field identifies the synchronization source. This identifier is chosen randomly, with the intent that no two synchronization sources within the same RTP session will have the same SSRC identifier. An example algorithm for generating a random identifier is presented in Appendix A.6. Although the probability of multiple sources choosing the same identifier is low, all RTP implementations must be prepared to detect and resolve collisions. Section 8 describes the probability of collision along with a mechanism for resolving collisions and detecting RTP-level forwarding loops based on the uniqueness of the SSRC identifier. If a source changes its source transport address, it must also choose a new SSRC identifier to avoid being interpreted as a looped source. CSRC list: 0 to 15 items, 32 bits each The CSRC list identifies the contributing sources for the payload contained in this packet. The number of identifiers is given by the CC field. If there are more than 15 contributing sources, only 15 may be identified. CSRC identifiers are Schulzrinne, et al Standards Track [Page 12]
RFC 1889 RTP January 1996 inserted by mixers, using the SSRC identifiers of contributing sources. For example, for audio packets the SSRC identifiers of all sources that were mixed together to create a packet are listed, allowing correct talker indication at the receiver. 5.2 Multiplexing RTP Sessions For efficient protocol processing, the number of multiplexing points should be minimized, as described in the integrated layer processing design principle [1]. In RTP, multiplexing is provided by the destination transport address (network address and port number) which define an RTP session. For example, in a teleconference composed of audio and video media encoded separately, each medium should be carried in a separate RTP session with its own destination transport address. It is not intended that the audio and video be carried in a single RTP session and demultiplexed based on the payload type or SSRC fields. Interleaving packets with different payload types but using the same SSRC would introduce several problems: 1. If one payload type were switched during a session, there would be no general means to identify which of the old values the new one replaced. 2. An SSRC is defined to identify a single timing and sequence number space. Interleaving multiple payload types would require different timing spaces if the media clock rates differ and would require different sequence number spaces to tell which payload type suffered packet loss. 3. The RTCP sender and receiver reports (see Section 6.3) can only describe one timing and sequence number space per SSRC and do not carry a payload type field. 4. An RTP mixer would not be able to combine interleaved streams of incompatible media into one stream. 5. Carrying multiple media in one RTP session precludes: the use of different network paths or network resource allocations if appropriate; reception of a subset of the media if desired, for example just audio if video would exceed the available bandwidth; and receiver implementations that use separate processes for the different media, whereas using separate RTP sessions permits either single- or multiple-process implementations. Using a different SSRC for each medium but sending them in the same RTP session would avoid the first three problems but not the last two. Schulzrinne, et al Standards Track [Page 13]
RFC 1889 RTP January 1996 5.3 Profile-Specific Modifications to the RTP Header The existing RTP data packet header is believed to be complete for the set of functions required in common across all the application classes that RTP might support. However, in keeping with the ALF design principle, the header may be tailored through modifications or additions defined in a profile specification while still allowing profile-independent monitoring and recording tools to function. o The marker bit and payload type field carry profile-specific information, but they are allocated in the fixed header since many applications are expected to need them and might otherwise have to add another 32-bit word just to hold them. The octet containing these fields may be redefined by a profile to suit different requirements, for example with a more or fewer marker bits. If there are any marker bits, one should be located in the most significant bit of the octet since profile-independent monitors may be able to observe a correlation between packet loss patterns and the marker bit. o Additional information that is required for a particular payload format, such as a video encoding, should be carried in the payload section of the packet. This might be in a header that is always present at the start of the payload section, or might be indicated by a reserved value in the data pattern. o If a particular class of applications needs additional functionality independent of payload format, the profile under which those applications operate should define additional fixed fields to follow immediately after the SSRC field of the existing fixed header. Those applications will be able to quickly and directly access the additional fields while profile-independent monitors or recorders can still process the RTP packets by interpreting only the first twelve octets. If it turns out that additional functionality is needed in common across all profiles, then a new version of RTP should be defined to make a permanent change to the fixed header. 5.3.1 RTP Header Extension An extension mechanism is provided to allow individual implementations to experiment with new payload-format-independent functions that require additional information to be carried in the RTP data packet header. This mechanism is designed so that the header extension may be ignored by other interoperating implementations that have not been extended. Schulzrinne, et al Standards Track [Page 14]
RFC 1889 RTP January 1996 Note that this header extension is intended only for limited use. Most potential uses of this mechanism would be better done another way, using the methods described in the previous section. For example, a profile-specific extension to the fixed header is less expensive to process because it is not conditional nor in a variable location. Additional information required for a particular payload format should not use this header extension, but should be carried in the payload section of the packet. 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | defined by profile | length | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | header extension | | .... | If the X bit in the RTP header is one, a variable-length header extension is appended to the RTP header, following the CSRC list if present. The header extension contains a 16-bit length field that counts the number of 32-bit words in the extension, excluding the four-octet extension header (therefore zero is a valid length). Only a single extension may be appended to the RTP data header. To allow multiple interoperating implementations to each experiment independently with different header extensions, or to allow a particular implementation to experiment with more than one type of header extension, the first 16 bits of the header extension are left open for distinguishing identifiers or parameters. The format of these 16 bits is to be defined by the profile specification under which the implementations are operating. This RTP specification does not define any header extensions itself. 6. RTP Control Protocol -- RTCP The RTP control protocol (RTCP) is based on the periodic transmission of control packets to all participants in the session, using the same distribution mechanism as the data packets. The underlying protocol must provide multiplexing of the data and control packets, for example using separate port numbers with UDP. RTCP performs four functions: 1. The primary function is to provide feedback on the quality of the data distribution. This is an integral part of the RTP's role as a transport protocol and is related to the flow and congestion control functions of other transport protocols. The feedback may be directly useful for control of adaptive encodings [8,9], but experiments with IP Schulzrinne, et al Standards Track [Page 15]
RFC 1889 RTP January 1996 multicasting have shown that it is also critical to get feedback from the receivers to diagnose faults in the distribution. Sending reception feedback reports to all participants allows one who is observing problems to evaluate whether those problems are local or global. With a distribution mechanism like IP multicast, it is also possible for an entity such as a network service provider who is not otherwise involved in the session to receive the feedback information and act as a third-party monitor to diagnose network problems. This feedback function is performed by the RTCP sender and receiver reports, described below in Section 6.3. 2. RTCP carries a persistent transport-level identifier for an RTP source called the canonical name or CNAME, Section 6.4.1. Since the SSRC identifier may change if a conflict is discovered or a program is restarted, receivers require the CNAME to keep track of each participant. Receivers also require the CNAME to associate multiple data streams from a given participant in a set of related RTP sessions, for example to synchronize audio and video. 3. The first two functions require that all participants send RTCP packets, therefore the rate must be controlled in order for RTP to scale up to a large number of participants. By having each participant send its control packets to all the others, each can independently observe the number of participants. This number is used to calculate the rate at which the packets are sent, as explained in Section 6.2. 4. A fourth, optional function is to convey minimal session control information, for example participant identification to be displayed in the user interface. This is most likely to be useful in "loosely controlled" sessions where participants enter and leave without membership control or parameter negotiation. RTCP serves as a convenient channel to reach all the participants, but it is not necessarily expected to support all the control communication requirements of an application. A higher-level session control protocol, which is beyond the scope of this document, may be needed. Functions 1-3 are mandatory when RTP is used in the IP multicast environment, and are recommended for all environments. RTP application designers are advised to avoid mechanisms that can only work in unicast mode and will not scale to larger numbers. Schulzrinne, et al Standards Track [Page 16]
RFC 1889 RTP January 1996
RFC 1889 RTP January 1996 RTP data header extensions: The contents of the first 16 bits of the RTP data header extension structure must be defined if use of that mechanism is to be allowed under the profile for implementation-specific extensions (Section 5.3.1). RTCP packet types: New application-class-specific RTCP packet types may be defined and registered with IANA. RTCP report interval: A profile should specify that the values suggested in Section 6.2 for the constants employed in the calculation of the RTCP report interval will be used. Those are the RTCP fraction of session bandwidth, the minimum report interval, and the bandwidth split between senders and receivers. A profile may specify alternate values if they have been demonstrated to work in a scalable manner. SR/RR extension: An extension section may be defined for the RTCP SR and RR packets if there is additional information that should be reported regularly about the sender or receivers (Section 6.3.3). SDES use: The profile may specify the relative priorities for RTCP SDES items to be transmitted or excluded entirely (Section 6.2.2); an alternate syntax or semantics for the CNAME item (Section 6.4.1); the format of the LOC item (Section 6.4.5); the semantics and use of the NOTE item (Section 6.4.7); or new SDES item types to be registered with IANA. Security: A profile may specify which security services and algorithms should be offered by applications, and may provide guidance as to their appropriate use (Section 9). String-to-key mapping: A profile may specify how a user-provided password or pass phrase is mapped into an encryption key. Underlying protocol: Use of a particular underlying network or transport layer protocol to carry RTP packets may be required. Transport mapping: A mapping of RTP and RTCP to transport-level addresses, e.g., UDP ports, other than the standard mapping defined in Section 10 may be specified. Encapsulation: An encapsulation of RTP packets may be defined to allow multiple RTP data packets to be carried in one lower-layer packet or to provide framing over underlying protocols that do not already do so (Section 10). Schulzrinne, et al Standards Track [Page 54]
RFC 1889 RTP January 1996 It is not expected that a new profile will be required for every application. Within one application class, it would be better to extend an existing profile rather than make a new one in order to facilitate interoperation among the applications since each will typically run under only one profile. Simple extensions such as the definition of additional payload type values or RTCP packet types may be accomplished by registering them through the Internet Assigned Numbers Authority and publishing their descriptions in an addendum to the profile or in a payload format specification. Schulzrinne, et al Standards Track [Page 55]
RFC 1889 RTP January 1996 A. Algorithms We provide examples of C code for aspects of RTP sender and receiver algorithms. There may be other implementation methods that are faster in particular operating environments or have other advantages. These implementation notes are for informational purposes only and are meant to clarify the RTP specification. The following definitions are used for all examples; for clarity and brevity, the structure definitions are only valid for 32-bit big- endian (most significant octet first) architectures. Bit fields are assumed to be packed tightly in big-endian bit order, with no additional padding. Modifications would be required to construct a portable implementation. /* * rtp.h -- RTP header file (RFC XXXX) */ #include <sys/types.h> /* * The type definitions below are valid for 32-bit architectures and * may have to be adjusted for 16- or 64-bit architectures. */ typedef unsigned char u_int8; typedef unsigned short u_int16; typedef unsigned int u_int32; typedef short int16; /* * Current protocol version. */ #define RTP_VERSION 2 #define RTP_SEQ_MOD (1<<16) #define RTP_MAX_SDES 255 /* maximum text length for SDES */ typedef enum { RTCP_SR = 200, RTCP_RR = 201, RTCP_SDES = 202, RTCP_BYE = 203, RTCP_APP = 204 } rtcp_type_t; typedef enum { RTCP_SDES_END = 0, RTCP_SDES_CNAME = 1, Schulzrinne, et al Standards Track [Page 56]
RFC 1889 RTP January 1996 RTCP_SDES_NAME = 2, RTCP_SDES_EMAIL = 3, RTCP_SDES_PHONE = 4, RTCP_SDES_LOC = 5, RTCP_SDES_TOOL = 6, RTCP_SDES_NOTE = 7, RTCP_SDES_PRIV = 8 } rtcp_sdes_type_t; /* * RTP data header */ typedef struct { unsigned int version:2; /* protocol version */ unsigned int p:1; /* padding flag */ unsigned int x:1; /* header extension flag */ unsigned int cc:4; /* CSRC count */ unsigned int m:1; /* marker bit */ unsigned int pt:7; /* payload type */ u_int16 seq; /* sequence number */ u_int32 ts; /* timestamp */ u_int32 ssrc; /* synchronization source */ u_int32 csrc[1]; /* optional CSRC list */ } rtp_hdr_t; /* * RTCP common header word */ typedef struct { unsigned int version:2; /* protocol version */ unsigned int p:1; /* padding flag */ unsigned int count:5; /* varies by packet type */ unsigned int pt:8; /* RTCP packet type */ u_int16 length; /* pkt len in words, w/o this word */ } rtcp_common_t; /* * Big-endian mask for version, padding bit and packet type pair */ #define RTCP_VALID_MASK (0xc000 | 0x2000 | 0xfe) #define RTCP_VALID_VALUE ((RTP_VERSION << 14) | RTCP_SR) /* * Reception report block */ typedef struct { u_int32 ssrc; /* data source being reported */ unsigned int fraction:8; /* fraction lost since last SR/RR */ Schulzrinne, et al Standards Track [Page 57]
RFC 1889 RTP January 1996 int lost:24; /* cumul. no. pkts lost (signed!) */ u_int32 last_seq; /* extended last seq. no. received */ u_int32 jitter; /* interarrival jitter */ u_int32 lsr; /* last SR packet from this source */ u_int32 dlsr; /* delay since last SR packet */ } rtcp_rr_t; /* * SDES item */ typedef struct { u_int8 type; /* type of item (rtcp_sdes_type_t) */ u_int8 length; /* length of item (in octets) */ char data[1]; /* text, not null-terminated */ } rtcp_sdes_item_t; /* * One RTCP packet */ typedef struct { rtcp_common_t common; /* common header */ union { /* sender report (SR) */ struct { u_int32 ssrc; /* sender generating this report */ u_int32 ntp_sec; /* NTP timestamp */ u_int32 ntp_frac; u_int32 rtp_ts; /* RTP timestamp */ u_int32 psent; /* packets sent */ u_int32 osent; /* octets sent */ rtcp_rr_t rr[1]; /* variable-length list */ } sr; /* reception report (RR) */ struct { u_int32 ssrc; /* receiver generating this report */ rtcp_rr_t rr[1]; /* variable-length list */ } rr; /* source description (SDES) */ struct rtcp_sdes { u_int32 src; /* first SSRC/CSRC */ rtcp_sdes_item_t item[1]; /* list of SDES items */ } sdes; /* BYE */ struct { u_int32 src[1]; /* list of sources */ Schulzrinne, et al Standards Track [Page 58]
RFC 1889 RTP January 1996 /* can't express trailing text for reason */ } bye; } r; } rtcp_t; typedef struct rtcp_sdes rtcp_sdes_t; /* * Per-source state information */ typedef struct { u_int16 max_seq; /* highest seq. number seen */ u_int32 cycles; /* shifted count of seq. number cycles */ u_int32 base_seq; /* base seq number */ u_int32 bad_seq; /* last 'bad' seq number + 1 */ u_int32 probation; /* sequ. packets till source is valid */ u_int32 received; /* packets received */ u_int32 expected_prior; /* packet expected at last interval */ u_int32 received_prior; /* packet received at last interval */ u_int32 transit; /* relative trans time for prev pkt */ u_int32 jitter; /* estimated jitter */ /* ... */ } source; A.1 RTP Data Header Validity Checks An RTP receiver should check the validity of the RTP header on incoming packets since they might be encrypted or might be from a different application that happens to be misaddressed. Similarly, if encryption is enabled, the header validity check is needed to verify that incoming packets have been correctly decrypted, although a failure of the header validity check (e.g., unknown payload type) may not necessarily indicate decryption failure. Only weak validity checks are possible on an RTP data packet from a source that has not been heard before: o RTP version field must equal 2. o The payload type must be known, in particular it must not be equal to SR or RR. o If the P bit is set, then the last octet of the packet must contain a valid octet count, in particular, less than the total packet length minus the header size. o The X bit must be zero if the profile does not specify that the header extension mechanism may be used. Otherwise, the Schulzrinne, et al Standards Track [Page 59]
RFC 1889 RTP January 1996 extension length field must be less than the total packet size minus the fixed header length and padding. o The length of the packet must be consistent with CC and payload type (if payloads have a known length). The last three checks are somewhat complex and not always possible, leaving only the first two which total just a few bits. If the SSRC identifier in the packet is one that has been received before, then the packet is probably valid and checking if the sequence number is in the expected range provides further validation. If the SSRC identifier has not been seen before, then data packets carrying that identifier may be considered invalid until a small number of them arrive with consecutive sequence numbers. The routine update_seq shown below ensures that a source is declared valid only after MIN_SEQUENTIAL packets have been received in sequence. It also validates the sequence number seq of a newly received packet and updates the sequence state for the packet's source in the structure to which s points. When a new source is heard for the first time, that is, its SSRC identifier is not in the table (see Section 8.2), and the per-source state is allocated for it, s->probation should be set to the number of sequential packets required before declaring a source valid (parameter MIN_SEQUENTIAL ) and s->max_seq initialized to seq-1 s- >probation marks the source as not yet valid so the state may be discarded after a short timeout rather than a long one, as discussed in Section 6.2.1. After a source is considered valid, the sequence number is considered valid if it is no more than MAX_DROPOUT ahead of s->max_seq nor more than MAX_MISORDER behind. If the new sequence number is ahead of max_seq modulo the RTP sequence number range (16 bits), but is smaller than max_seq , it has wrapped around and the (shifted) count of sequence number cycles is incremented. A value of one is returned to indicate a valid sequence number. Otherwise, the value zero is returned to indicate that the validation failed, and the bad sequence number is stored. If the next packet received carries the next higher sequence number, it is considered the valid start of a new packet sequence presumably caused by an extended dropout or a source restart. Since multiple complete sequence number cycles may have been missed, the packet loss statistics are reset. Typical values for the parameters are shown, based on a maximum misordering time of 2 seconds at 50 packets/second and a maximum Schulzrinne, et al Standards Track [Page 60]
RFC 1889 RTP January 1996 dropout of 1 minute. The dropout parameter MAX_DROPOUT should be a small fraction of the 16-bit sequence number space to give a reasonable probability that new sequence numbers after a restart will not fall in the acceptable range for sequence numbers from before the restart. void init_seq(source *s, u_int16 seq) { s->base_seq = seq - 1; s->max_seq = seq; s->bad_seq = RTP_SEQ_MOD + 1; s->cycles = 0; s->received = 0; s->received_prior = 0; s->expected_prior = 0; /* other initialization */ } int update_seq(source *s, u_int16 seq) { u_int16 udelta = seq - s->max_seq; const int MAX_DROPOUT = 3000; const int MAX_MISORDER = 100; const int MIN_SEQUENTIAL = 2; /* * Source is not valid until MIN_SEQUENTIAL packets with * sequential sequence numbers have been received. */ if (s->probation) { /* packet is in sequence */ if (seq == s->max_seq + 1) { s->probation--; s->max_seq = seq; if (s->probation == 0) { init_seq(s, seq); s->received++; return 1; } } else { s->probation = MIN_SEQUENTIAL - 1; s->max_seq = seq; } return 0; } else if (udelta < MAX_DROPOUT) { /* in order, with permissible gap */ if (seq < s->max_seq) { /* Schulzrinne, et al Standards Track [Page 61]
RFC 1889 RTP January 1996 * Sequence number wrapped - count another 64K cycle. */ s->cycles += RTP_SEQ_MOD; } s->max_seq = seq; } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) { /* the sequence number made a very large jump */ if (seq == s->bad_seq) { /* * Two sequential packets -- assume that the other side * restarted without telling us so just re-sync * (i.e., pretend this was the first packet). */ init_seq(s, seq); } else { s->bad_seq = (seq + 1) & (RTP_SEQ_MOD-1); return 0; } } else { /* duplicate or reordered packet */ } s->received++; return 1; } The validity check can be made stronger requiring more than two packets in sequence. The disadvantages are that a larger number of initial packets will be discarded and that high packet loss rates could prevent validation. However, because the RTCP header validation is relatively strong, if an RTCP packet is received from a source before the data packets, the count could be adjusted so that only two packets are required in sequence. If initial data loss for a few seconds can be tolerated, an application could choose to discard all data packets from a source until a valid RTCP packet has been received from that source. Depending on the application and encoding, algorithms may exploit additional knowledge about the payload format for further validation. For payload types where the timestamp increment is the same for all packets, the timestamp values can be predicted from the previous packet received from the same source using the sequence number difference (assuming no change in payload type). A strong "fast-path" check is possible since with high probability the first four octets in the header of a newly received RTP data packet will be just the same as that of the previous packet from the same SSRC except that the sequence number will have increased by one. Schulzrinne, et al Standards Track [Page 62]
RFC 1889 RTP January 1996 Similarly, a single-entry cache may be used for faster SSRC lookups in applications where data is typically received from one source at a time. A.2 RTCP Header Validity Checks The following checks can be applied to RTCP packets. o RTP version field must equal 2. o The payload type field of the first RTCP packet in a compound packet must be equal to SR or RR. o The padding bit (P) should be zero for the first packet of a compound RTCP packet because only the last should possibly need padding. o The length fields of the individual RTCP packets must total to the overall length of the compound RTCP packet as received. This is a fairly strong check. The code fragment below performs all of these checks. The packet type is not checked for subsequent packets since unknown packet types may be present and should be ignored. u_int32 len; /* length of compound RTCP packet in words */ rtcp_t *r; /* RTCP header */ rtcp_t *end; /* end of compound RTCP packet */ if ((*(u_int16 *)r & RTCP_VALID_MASK) != RTCP_VALID_VALUE) { /* something wrong with packet format */ } end = (rtcp_t *)((u_int32 *)r + len); do r = (rtcp_t *)((u_int32 *)r + r->common.length + 1); while (r < end && r->common.version == 2); if (r != end) { /* something wrong with packet format */ } A.3 Determining the Number of RTP Packets Expected and Lost In order to compute packet loss rates, the number of packets expected and actually received from each source needs to be known, using per- source state information defined in struct source referenced via pointer s in the code below. The number of packets received is simply the count of packets as they arrive, including any late or duplicate Schulzrinne, et al Standards Track [Page 63]
RFC 1889 RTP January 1996 packets. The number of packets expected can be computed by the receiver as the difference between the highest sequence number received ( s->max_seq ) and the first sequence number received ( s- >base_seq ). Since the sequence number is only 16 bits and will wrap around, it is necessary to extend the highest sequence number with the (shifted) count of sequence number wraparounds ( s->cycles ). Both the received packet count and the count of cycles are maintained the RTP header validity check routine in Appendix A.1. extended_max = s->cycles + s->max_seq; expected = extended_max - s->base_seq + 1; The number of packets lost is defined to be the number of packets expected less the number of packets actually received: lost = expected - s->received; Since this number is carried in 24 bits, it should be clamped at 0xffffff rather than wrap around to zero. The fraction of packets lost during the last reporting interval (since the previous SR or RR packet was sent) is calculated from differences in the expected and received packet counts across the interval, where expected_prior and received_prior are the values saved when the previous reception report was generated: expected_interval = expected - s->expected_prior; s->expected_prior = expected; received_interval = s->received - s->received_prior; s->received_prior = s->received; lost_interval = expected_interval - received_interval; if (expected_interval == 0 || lost_interval <= 0) fraction = 0; else fraction = (lost_interval << 8) / expected_interval; The resulting fraction is an 8-bit fixed point number with the binary point at the left edge. A.4 Generating SDES RTCP Packets This function builds one SDES chunk into buffer b composed of argc items supplied in arrays type , value and length b char *rtp_write_sdes(char *b, u_int32 src, int argc, rtcp_sdes_type_t type[], char *value[], int length[]) { rtcp_sdes_t *s = (rtcp_sdes_t *)b; rtcp_sdes_item_t *rsp; Schulzrinne, et al Standards Track [Page 64]
RFC 1889 RTP January 1996 int i; int len; int pad; /* SSRC header */ s->src = src; rsp = &s->item[0]; /* SDES items */ for (i = 0; i < argc; i++) { rsp->type = type[i]; len = length[i]; if (len > RTP_MAX_SDES) { /* invalid length, may want to take other action */ len = RTP_MAX_SDES; } rsp->length = len; memcpy(rsp->data, value[i], len); rsp = (rtcp_sdes_item_t *)&rsp->data[len]; } /* terminate with end marker and pad to next 4-octet boundary */ len = ((char *) rsp) - b; pad = 4 - (len & 0x3); b = (char *) rsp; while (pad--) *b++ = RTCP_SDES_END; return b; } A.5 Parsing RTCP SDES Packets This function parses an SDES packet, calling functions find_member() to find a pointer to the information for a session member given the SSRC identifier and member_sdes() to store the new SDES information for that member. This function expects a pointer to the header of the RTCP packet. void rtp_read_sdes(rtcp_t *r) { int count = r->common.count; rtcp_sdes_t *sd = &r->r.sdes; rtcp_sdes_item_t *rsp, *rspn; rtcp_sdes_item_t *end = (rtcp_sdes_item_t *) ((u_int32 *)r + r->common.length + 1); source *s; while (--count >= 0) { Schulzrinne, et al Standards Track [Page 65]
RFC 1889 RTP January 1996 rsp = &sd->item[0]; if (rsp >= end) break; s = find_member(sd->src); for (; rsp->type; rsp = rspn ) { rspn = (rtcp_sdes_item_t *)((char*)rsp+rsp->length+2); if (rspn >= end) { rsp = rspn; break; } member_sdes(s, rsp->type, rsp->data, rsp->length); } sd = (rtcp_sdes_t *) ((u_int32 *)sd + (((char *)rsp - (char *)sd) >> 2)+1); } if (count >= 0) { /* invalid packet format */ } } A.6 Generating a Random 32-bit Identifier The following subroutine generates a random 32-bit identifier using the MD5 routines published in RFC 1321 [23]. The system routines may not be present on all operating systems, but they should serve as hints as to what kinds of information may be used. Other system calls that may be appropriate include o getdomainname() , o getwd() , or o getrusage() "Live" video or audio samples are also a good source of random numbers, but care must be taken to avoid using a turned-off microphone or blinded camera as a source [7]. Use of this or similar routine is suggested to generate the initial seed for the random number generator producing the RTCP period (as shown in Appendix A.7), to generate the initial values for the sequence number and timestamp, and to generate SSRC values. Since this routine is likely to be CPU-intensive, its direct use to generate RTCP periods is inappropriate because predictability is not an issue. Note that this routine produces the same result on repeated calls until the value of the system clock changes unless different values are supplied for the type argument. Schulzrinne, et al Standards Track [Page 66]
RFC 1889 RTP January 1996 /* * Generate a random 32-bit quantity. */ #include <sys/types.h> /* u_long */ #include <sys/time.h> /* gettimeofday() */ #include <unistd.h> /* get..() */ #include <stdio.h> /* printf() */ #include <time.h> /* clock() */ #include <sys/utsname.h> /* uname() */ #include "global.h" /* from RFC 1321 */ #include "md5.h" /* from RFC 1321 */ #define MD_CTX MD5_CTX #define MDInit MD5Init #define MDUpdate MD5Update #define MDFinal MD5Final static u_long md_32(char *string, int length) { MD_CTX context; union { char c[16]; u_long x[4]; } digest; u_long r; int i; MDInit (&context); MDUpdate (&context, string, length); MDFinal ((unsigned char *)&digest, &context); r = 0; for (i = 0; i < 3; i++) { r ^= digest.x[i]; } return r; } /* md_32 */ /* * Return random unsigned 32-bit quantity. Use 'type' argument if you * need to generate several different values in close succession. */ u_int32 random32(int type) { struct { int type; struct timeval tv; clock_t cpu; Schulzrinne, et al Standards Track [Page 67]
RFC 1889 RTP January 1996 pid_t pid; u_long hid; uid_t uid; gid_t gid; struct utsname name; } s; gettimeofday(&s.tv, 0); uname(&s.name); s.type = type; s.cpu = clock(); s.pid = getpid(); s.hid = gethostid(); s.uid = getuid(); s.gid = getgid(); return md_32((char *)&s, sizeof(s)); } /* random32 */ A.7 Computing the RTCP Transmission Interval The following function returns the time between transmissions of RTCP packets, measured in seconds. It should be called after sending one compound RTCP packet to calculate the delay until the next should be sent. This function should also be called to calculate the delay before sending the first RTCP packet upon startup rather than send the packet immediately. This avoids any burst of RTCP packets if an application is started at many sites simultaneously, for example as a result of a session announcement. The parameters have the following meaning: rtcp_bw: The target RTCP bandwidth, i.e., the total bandwidth that will be used for RTCP packets by all members of this session, in octets per second. This should be 5% of the "session bandwidth" parameter supplied to the application at startup. senders: Number of active senders since sending last report, known from construction of receiver reports for this RTCP packet. Includes ourselves, if we also sent during this interval. members: The estimated number of session members, including ourselves. Incremented as we discover new session members from the receipt of RTP or RTCP packets, and decremented as session members leave (via RTCP BYE) or their state is timed out (30 minutes is recommended). On the first call, this parameter should have the value 1. Schulzrinne, et al Standards Track [Page 68]
RFC 1889 RTP January 1996 we_sent: Flag that is true if we have sent data during the last two RTCP intervals. If the flag is true, the compound RTCP packet just sent contained an SR packet. packet_size: The size of the compound RTCP packet just sent, in octets, including the network encapsulation (e.g., 28 octets for UDP over IP). avg_rtcp_size: Pointer to estimator for compound RTCP packet size; initialized and updated by this function for the packet just sent, and also updated by an identical line of code in the RTCP receive routine for every RTCP packet received from other participants in the session. initial: Flag that is true for the first call upon startup to calculate the time until the first report should be sent. #include <math.h> double rtcp_interval(int members, int senders, double rtcp_bw, int we_sent, int packet_size, int *avg_rtcp_size, int initial) { /* * Minimum time between RTCP packets from this site (in seconds). * This time prevents the reports from `clumping' when sessions * are small and the law of large numbers isn't helping to smooth * out the traffic. It also keeps the report interval from * becoming ridiculously small during transient outages like a * network partition. */ double const RTCP_MIN_TIME = 5.; /* * Fraction of the RTCP bandwidth to be shared among active * senders. (This fraction was chosen so that in a typical * session with one or two active senders, the computed report * time would be roughly equal to the minimum report time so that * we don't unnecessarily slow down receiver reports.) The * receiver fraction must be 1 - the sender fraction. */ double const RTCP_SENDER_BW_FRACTION = 0.25; double const RTCP_RCVR_BW_FRACTION = (1-RTCP_SENDER_BW_FRACTION); /* * Gain (smoothing constant) for the low-pass filter that Schulzrinne, et al Standards Track [Page 69]
RFC 1889 RTP January 1996 * estimates the average RTCP packet size (see Cadzow reference). */ double const RTCP_SIZE_GAIN = (1./16.); double t; /* interval */ double rtcp_min_time = RTCP_MIN_TIME; int n; /* no. of members for computation */ /* * Very first call at application start-up uses half the min * delay for quicker notification while still allowing some time * before reporting for randomization and to learn about other * sources so the report interval will converge to the correct * interval more quickly. The average RTCP size is initialized * to 128 octets which is conservative (it assumes everyone else * is generating SRs instead of RRs: 20 IP + 8 UDP + 52 SR + 48 * SDES CNAME). */ if (initial) { rtcp_min_time /= 2; *avg_rtcp_size = 128; } /* * If there were active senders, give them at least a minimum * share of the RTCP bandwidth. Otherwise all participants share * the RTCP bandwidth equally. */ n = members; if (senders > 0 && senders < members * RTCP_SENDER_BW_FRACTION) { if (we_sent) { rtcp_bw *= RTCP_SENDER_BW_FRACTION; n = senders; } else { rtcp_bw *= RTCP_RCVR_BW_FRACTION; n -= senders; } } /* * Update the average size estimate by the size of the report * packet we just sent. */ *avg_rtcp_size += (packet_size - *avg_rtcp_size)*RTCP_SIZE_GAIN; /* * The effective number of sites times the average packet size is * the total number of octets sent when each site sends a report. Schulzrinne, et al Standards Track [Page 70]
RFC 1889 RTP January 1996 * Dividing this by the effective bandwidth gives the time * interval over which those packets must be sent in order to * meet the bandwidth target, with a minimum enforced. In that * time interval we send one report so this time is also our * average time between reports. */ t = (*avg_rtcp_size) * n / rtcp_bw; if (t < rtcp_min_time) t = rtcp_min_time; /* * To avoid traffic bursts from unintended synchronization with * other sites, we then pick our actual next report interval as a * random number uniformly distributed between 0.5*t and 1.5*t. */ return t * (drand48() + 0.5); } A.8 Estimating the Interarrival Jitter The code fragments below implement the algorithm given in Section 6.3.1 for calculating an estimate of the statistical variance of the RTP data interarrival time to be inserted in the interarrival jitter field of reception reports. The inputs are r->ts , the timestamp from the incoming packet, and arrival , the current time in the same units. Here s points to state for the source; s->transit holds the relative transit time for the previous packet, and s->jitter holds the estimated jitter. The jitter field of the reception report is measured in timestamp units and expressed as an unsigned integer, but the jitter estimate is kept in a floating point. As each data packet arrives, the jitter estimate is updated: int transit = arrival - r->ts; int d = transit - s->transit; s->transit = transit; if (d < 0) d = -d; s->jitter += (1./16.) * ((double)d - s->jitter); When a reception report block (to which rr points) is generated for this member, the current jitter estimate is returned: rr->jitter = (u_int32) s->jitter; Alternatively, the jitter estimate can be kept as an integer, but scaled to reduce round-off error. The calculation is the same except for the last line: s->jitter += d - ((s->jitter + 8) >> 4); Schulzrinne, et al Standards Track [Page 71]
RFC 1889 RTP January 1996 In this case, the estimate is sampled for the reception report as: rr->jitter = s->jitter >> 4; B. Security Considerations RTP suffers from the same security liabilities as the underlying protocols. For example, an impostor can fake source or destination network addresses, or change the header or payload. Within RTCP, the CNAME and NAME information may be used to impersonate another participant. In addition, RTP may be sent via IP multicast, which provides no direct means for a sender to know all the receivers of the data sent and therefore no measure of privacy. Rightly or not, users may be more sensitive to privacy concerns with audio and video communication than they have been with more traditional forms of network communication [24]. Therefore, the use of security mechanisms with RTP is important. These mechanisms are discussed in Section 9. RTP-level translators or mixers may be used to allow RTP traffic to reach hosts behind firewalls. Appropriate firewall security principles and practices, which are beyond the scope of this document, should be followed in the design and installation of these devices and in the admission of RTP applications for use behind the firewall. C. Authors' Addresses Henning Schulzrinne GMD Fokus Hardenbergplatz 2 D-10623 Berlin Germany EMail: schulzrinne@fokus.gmd.de Stephen L. Casner Precept Software, Inc. 21580 Stevens Creek Boulevard, Suite 207 Cupertino, CA 95014 United States EMail: casner@precept.com Schulzrinne, et al Standards Track [Page 72]
RFC 1889 RTP January 1996 Ron Frederick Xerox Palo Alto Research Center 3333 Coyote Hill Road Palo Alto, CA 94304 United States EMail: frederic@parc.xerox.com Van Jacobson MS 46a-1121 Lawrence Berkeley National Laboratory Berkeley, CA 94720 United States EMail: van@ee.lbl.gov Acknowledgments This memorandum is based on discussions within the IETF Audio/Video Transport working group chaired by Stephen Casner. The current protocol has its origins in the Network Voice Protocol and the Packet Video Protocol (Danny Cohen and Randy Cole) and the protocol implemented by the vat application (Van Jacobson and Steve McCanne). Christian Huitema provided ideas for the random identifier generator. D. Bibliography [1] D. D. Clark and D. L. Tennenhouse, "Architectural considerations for a new generation of protocols," in SIGCOMM Symposium on Communications Architectures and Protocols , (Philadelphia, Pennsylvania), pp. 200--208, IEEE, Sept. 1990. Computer Communications Review, Vol. 20(4), Sept. 1990. [2] H. Schulzrinne, "Issues in designing a transport protocol for audio and video conferences and other multiparticipant real-time applications", Work in Progress. [3] D. E. Comer, Internetworking with TCP/IP , vol. 1. Englewood Cliffs, New Jersey: Prentice Hall, 1991. [4] Postel, J., "Internet Protocol", STD 5, RFC 791, USC/Information Sciences Institute, September 1981. [5] Mills, D., "Network Time Protocol Version 3", RFC 1305, UDEL, March 1992. Schulzrinne, et al Standards Track [Page 73]
RFC 1889 RTP January 1996 [6] Reynolds, J., and J. Postel, "Assigned Numbers", STD 2, RFC 1700, USC/Information Sciences Institute, October 1994. [7] Eastlake, D., Crocker, S., and J. Schiller, "Randomness Recommendations for Security", RFC 1750, DEC, Cybercash, MIT, December 1994. [8] J.-C. Bolot, T. Turletti, and I. Wakeman, "Scalable feedback control for multicast video distribution in the internet," in SIGCOMM Symposium on Communications Architectures and Protocols , (London, England), pp. 58--67, ACM, Aug. 1994. [9] I. Busse, B. Deffner, and H. Schulzrinne, "Dynamic QoS control of multimedia applications based on RTP," Computer Communications , Jan. 1996. [10] S. Floyd and V. Jacobson, "The synchronization of periodic routing messages," in SIGCOMM Symposium on Communications Architectures and Protocols (D. P. Sidhu, ed.), (San Francisco, California), pp. 33--44, ACM, Sept. 1993. also in [25]. [11] J. A. Cadzow, Foundations of digital signal processing and data analysis New York, New York: Macmillan, 1987. [12] International Standards Organization, "ISO/IEC DIS 10646-1:1993 information technology -- universal multiple-octet coded character set (UCS) -- part I: Architecture and basic multilingual plane," 1993. [13] The Unicode Consortium, The Unicode Standard New York, New York: Addison-Wesley, 1991. [14] Mockapetris, P., "Domain Names - Concepts and Facilities", STD 13, RFC 1034, USC/Information Sciences Institute, November 1987. [15] Mockapetris, P., "Domain Names - Implementation and Specification", STD 13, RFC 1035, USC/Information Sciences Institute, November 1987. [16] Braden, R., "Requirements for Internet Hosts - Application and Support", STD 3, RFC 1123, Internet Engineering Task Force, October 1989. [17] Rekhter, Y., Moskowitz, R., Karrenberg, D., and G. de Groot, "Address Allocation for Private Internets", RFC 1597, T.J. Watson Research Center, IBM Corp., Chrysler Corp., RIPE NCC, March 1994. Schulzrinne, et al Standards Track [Page 74]
RFC 1889 RTP January 1996 [18] Lear, E., Fair, E., Crocker, D., and T. Kessler, "Network 10 Considered Harmful (Some Practices Shouldn't be Codified)", RFC 1627, Silicon Graphics, Inc., Apple Computer, Inc., Silicon Graphics, Inc., July 1994. [19] Crocker, D., "Standard for the Format of ARPA Internet Text Messages", STD 11, RFC 822, UDEL, August 1982. [20] W. Feller, An Introduction to Probability Theory and its Applications, Volume 1 , vol. 1. New York, New York: John Wiley and Sons, third ed., 1968. [21] Balenson, D., "Privacy Enhancement for Internet Electronic Mail: Part III: Algorithms, Modes, and Identifiers", RFC 1423, TIS, IAB IRTF PSRG, IETF PEM WG, February 1993. [22] V. L. Voydock and S. T. Kent, "Security mechanisms in high-level network protocols," ACM Computing Surveys , vol. 15, pp. 135-- 171, June 1983.



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