苏辉的父亲:RTP: A Transport Protocol for Real-Time Applications

Network Working Group                Audio-Video Transport Working GroupRequest for Comments: 1889                                H. SchulzrinneCategory: Standards Track                                      GMD Fokus                                                               S. Casner                                                  Precept Software, Inc.                                                            R. Frederick                                         Xerox Palo Alto Research Center                                                             V. Jacobson                                   Lawrence Berkeley National Laboratory                                                            January 1996          RTP: A Transport Protocol for Real-Time ApplicationsStatus of this Memo   This document specifies an Internet standards track protocol for the   Internet community, and requests discussion and suggestions for   improvements.  Please refer to the current edition of the "Internet   Official Protocol Standards" (STD 1) for the standardization state   and status of this protocol.  Distribution of this memo is unlimited.Abstract   This memorandum describes RTP, the real-time transport protocol. RTP   provides end-to-end network transport functions suitable for   applications transmitting real-time data, such as audio, video or   simulation data, over multicast or unicast network services. RTP does   not address resource reservation and does not guarantee quality-of-   service for real-time services. The data transport is augmented by a   control protocol (RTCP) to allow monitoring of the data delivery in a   manner scalable to large multicast networks, and to provide minimal   control and identification functionality. RTP and RTCP are designed   to be independent of the underlying transport and network layers. The   protocol supports the use of RTP-level translators and mixers.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 ...........................   13Schulzrinne, 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 ............   68Schulzrinne, 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 ........................................   731.  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-profile2.1 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 anSchulzrinne, 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 Protocol5.1 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 theSchulzrinne, 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 derivedSchulzrinne, 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 areSchulzrinne, 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 19965.3 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 IPSchulzrinne, 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 19966.1 RTCP Packet Format   This specification defines several RTCP packet types to carry a   variety of control information:   SR: Sender report, for transmission and reception statistics from        participants that are active senders   RR: Receiver report, for reception statistics from participants that        are not active senders   SDES: Source description items, including CNAME   BYE: Indicates end of participation   APP: Application specific functions   Each RTCP packet begins with a fixed part similar to that of RTP data   packets, followed by structured elements that may be of variable   length according to the packet type but always end on a 32-bit   boundary. The alignment requirement and a length field in the fixed   part are included to make RTCP packets "stackable". Multiple RTCP   packets may be concatenated without any intervening separators to   form a compound RTCP packet that is sent in a single packet of the   lower layer protocol, for example UDP. There is no explicit count of   individual RTCP packets in the compound packet since the lower layer   protocols are expected to provide an overall length to determine the   end of the compound packet.   Each individual RTCP packet in the compound packet may be processed   independently with no requirements upon the order or combination of   packets. However, in order to perform the functions of the protocol,   the following constraints are imposed:        o Reception statistics (in SR or RR) should be sent as often as         bandwidth constraints will allow to maximize the resolution of         the statistics, therefore each periodically transmitted         compound RTCP packet should include a report packet.        o New receivers need to receive the CNAME for a source as soon         as possible to identify the source and to begin associating         media for purposes such as lip-sync, so each compound RTCP         packet should also include the SDES CNAME.        o The number of packet types that may appear first in the         compound packet should be limited to increase the number of         constant bits in the first word and the probability of         successfully validating RTCP packets against misaddressed RTPSchulzrinne, et al          Standards Track                    [Page 17]RFC 1889                          RTP                       January 1996         data packets or other unrelated packets.   Thus, all RTCP packets must be sent in a compound packet of at least   two individual packets, with the following format recommended:   Encryption prefix:  If and only if the compound packet is to be        encrypted, it is prefixed by a random 32-bit quantity redrawn        for every compound packet transmitted.   SR or RR:  The first RTCP packet in the compound packet must always        be a report packet to facilitate header validation as described        in Appendix A.2. This is true even if no data has been sent nor        received, in which case an empty RR is sent, and even if the        only other RTCP packet in the compound packet is a BYE.   Additional RRs:  If the number of sources for which reception        statistics are being reported exceeds 31, the number that will        fit into one SR or RR packet, then additional RR packets should        follow the initial report packet.   SDES:  An SDES packet containing a CNAME item must be included in        each compound RTCP packet. Other source description items may        optionally be included if required by a particular application,        subject to bandwidth constraints (see Section 6.2.2).   BYE or APP:  Other RTCP packet types, including those yet to be        defined, may follow in any order, except that BYE should be the        last packet sent with a given SSRC/CSRC. Packet types may appear        more than once.   It is advisable for translators and mixers to combine individual RTCP   packets from the multiple sources they are forwarding into one   compound packet whenever feasible in order to amortize the packet   overhead (see Section 7). An example RTCP compound packet as might be   produced by a mixer is shown in Fig. 1.  If the overall length of a   compound packet would exceed the maximum transmission unit (MTU) of   the network path, it may be segmented into multiple shorter compound   packets to be transmitted in separate packets of the underlying   protocol. Note that each of the compound packets must begin with an   SR or RR packet.   An implementation may ignore incoming RTCP packets with types unknown   to it. Additional RTCP packet types may be registered with the   Internet Assigned Numbers Authority (IANA).Schulzrinne, et al          Standards Track                    [Page 18]RFC 1889                          RTP                       January 19966.2 RTCP Transmission Interval   if encrypted: random 32-bit integer    |    |[------- packet -------][----------- packet -----------][-packet-]    |    |             receiver reports          chunk        chunk    V                                    item  item     item  item   --------------------------------------------------------------------   |R[SR|# sender #site#site][SDES|# CNAME PHONE |#CNAME LOC][BYE##why]   |R[  |# report #  1 #  2 ][    |#             |#         ][   ##   ]   |R[  |#        #    #    ][    |#             |#         ][   ##   ]   |R[  |#        #    #    ][    |#             |#         ][   ##   ]   --------------------------------------------------------------------   |<------------------  UDP packet (compound packet) --------------->|   #: SSRC/CSRC              Figure 1: Example of an RTCP compound packet   RTP is designed to allow an application to scale automatically over   session sizes ranging from a few participants to thousands. For   example, in an audio conference the data traffic is inherently self-   limiting because only one or two people will speak at a time, so with   multicast distribution the data rate on any given link remains   relatively constant independent of the number of participants.   However, the control traffic is not self-limiting. If the reception   reports from each participant were sent at a constant rate, the   control traffic would grow linearly with the number of participants.   Therefore, the rate must be scaled down.   For each session, it is assumed that the data traffic is subject to   an aggregate limit called the "session bandwidth" to be divided among   the participants. This bandwidth might be reserved and the limit   enforced by the network, or it might just be a reasonable share. The   session bandwidth may be chosen based or some cost or a priori   knowledge of the available network bandwidth for the session. It is   somewhat independent of the media encoding, but the encoding choice   may be limited by the session bandwidth. The session bandwidth   parameter is expected to be supplied by a session management   application when it invokes a media application, but media   applications may also set a default based on the single-sender data   bandwidth for the encoding selected for the session. The application   may also enforce bandwidth limits based on multicast scope rules or   other criteria.Schulzrinne, et al          Standards Track                    [Page 19]RFC 1889                          RTP                       January 1996   Bandwidth calculations for control and data traffic include lower-   layer transport and network protocols (e.g., UDP and IP) since that   is what the resource reservation system would need to know. The   application can also be expected to know which of these protocols are   in use. Link level headers are not included in the calculation since   the packet will be encapsulated with different link level headers as   it travels.   The control traffic should be limited to a small and known fraction   of the session bandwidth: small so that the primary function of the   transport protocol to carry data is not impaired; known so that the   control traffic can be included in the bandwidth specification given   to a resource reservation protocol, and so that each participant can   independently calculate its share. It is suggested that the fraction   of the session bandwidth allocated to RTCP be fixed at 5%. While the   value of this and other constants in the interval calculation is not   critical, all participants in the session must use the same values so   the same interval will be calculated. Therefore, these constants   should be fixed for a particular profile.   The algorithm described in Appendix A.7 was designed to meet the   goals outlined above. It calculates the interval between sending   compound RTCP packets to divide the allowed control traffic bandwidth   among the participants. This allows an application to provide fast   response for small sessions where, for example, identification of all   participants is important, yet automatically adapt to large sessions.   The algorithm incorporates the following characteristics:        o Senders are collectively allocated at least 1/4 of the control         traffic bandwidth so that in sessions with a large number of         receivers but a small number of senders, newly joining         participants will more quickly receive the CNAME for the         sending sites.        o The calculated interval between RTCP packets is required to be         greater than a minimum of 5 seconds to avoid having bursts of         RTCP packets exceed the allowed bandwidth when the number of         participants is small and the traffic isn't smoothed according         to the law of large numbers.        o The interval between RTCP packets is varied randomly over the         range [0.5,1.5] times the calculated interval to avoid         unintended synchronization of all participants [10].  The first         RTCP packet sent after joining a session is also delayed by a         random variation of half the minimum RTCP interval in case the         application is started at multiple sites simultaneously, for         example as initiated by a session announcement.Schulzrinne, et al          Standards Track                    [Page 20]RFC 1889                          RTP                       January 1996        o A dynamic estimate of the average compound RTCP packet size is         calculated, including all those received and sent, to         automatically adapt to changes in the amount of control         information carried.   This algorithm may be used for sessions in which all participants are   allowed to send. In that case, the session bandwidth parameter is the   product of the individual sender's bandwidth times the number of   participants, and the RTCP bandwidth is 5% of that.6.2.1 Maintaining the number of session members   Calculation of the RTCP packet interval depends upon an estimate of   the number of sites participating in the session. New sites are added   to the count when they are heard, and an entry for each is created in   a table indexed by the SSRC or CSRC identifier (see Section 8.2) to   keep track of them. New entries may not be considered valid until   multiple packets carrying the new SSRC have been received (see   Appendix A.1). Entries may be deleted from the table when an RTCP BYE   packet with the corresponding SSRC identifier is received.   A participant may mark another site inactive, or delete it if not yet   valid, if no RTP or RTCP packet has been received for a small number   of RTCP report intervals (5 is suggested). This provides some   robustness against packet loss. All sites must calculate roughly the   same value for the RTCP report interval in order for this timeout to   work properly.   Once a site has been validated, then if it is later marked inactive   the state for that site should still be retained and the site should   continue to be counted in the total number of sites sharing RTCP   bandwidth for a period long enough to span typical network   partitions.  This is to avoid excessive traffic, when the partition   heals, due to an RTCP report interval that is too small. A timeout of   30 minutes is suggested. Note that this is still larger than 5 times   the largest value to which the RTCP report interval is expected to   usefully scale, about 2 to 5 minutes.6.2.2 Allocation of source description bandwidth   This specification defines several source description (SDES) items in   addition to the mandatory CNAME item, such as NAME (personal name)   and EMAIL (email address). It also provides a means to define new   application-specific RTCP packet types. Applications should exercise   caution in allocating control bandwidth to this additional   information because it will slow down the rate at which reception   reports and CNAME are sent, thus impairing the performance of the   protocol. It is recommended that no more than 20% of the RTCPSchulzrinne, et al          Standards Track                    [Page 21]RFC 1889                          RTP                       January 1996   bandwidth allocated to a single participant be used to carry the   additional information.  Furthermore, it is not intended that all   SDES items should be included in every application. Those that are   included should be assigned a fraction of the bandwidth according to   their utility.  Rather than estimate these fractions dynamically, it   is recommended that the percentages be translated statically into   report interval counts based on the typical length of an item.   For example, an application may be designed to send only CNAME, NAME   and EMAIL and not any others. NAME might be given much higher   priority than EMAIL because the NAME would be displayed continuously   in the application's user interface, whereas EMAIL would be displayed   only when requested. At every RTCP interval, an RR packet and an SDES   packet with the CNAME item would be sent. For a small session   operating at the minimum interval, that would be every 5 seconds on   the average. Every third interval (15 seconds), one extra item would   be included in the SDES packet. Seven out of eight times this would   be the NAME item, and every eighth time (2 minutes) it would be the   EMAIL item.   When multiple applications operate in concert using cross-application   binding through a common CNAME for each participant, for example in a   multimedia conference composed of an RTP session for each medium, the   additional SDES information might be sent in only one RTP session.   The other sessions would carry only the CNAME item.6.3 Sender and Receiver Reports   RTP receivers provide reception quality feedback using RTCP report   packets which may take one of two forms depending upon whether or not   the receiver is also a sender. The only difference between the sender   report (SR) and receiver report (RR) forms, besides the packet type   code, is that the sender report includes a 20-byte sender information   section for use by active senders. The SR is issued if a site has   sent any data packets during the interval since issuing the last   report or the previous one, otherwise the RR is issued.   Both the SR and RR forms include zero or more reception report   blocks, one for each of the synchronization sources from which this   receiver has received RTP data packets since the last report. Reports   are not issued for contributing sources listed in the CSRC list. Each   reception report block provides statistics about the data received   from the particular source indicated in that block. Since a maximum   of 31 reception report blocks will fit in an SR or RR packet,   additional RR packets may be stacked after the initial SR or RR   packet as needed to contain the reception reports for all sources   heard during the interval since the last report.Schulzrinne, et al          Standards Track                    [Page 22]RFC 1889                          RTP                       January 1996   The next sections define the formats of the two reports, how they may   be extended in a profile-specific manner if an application requires   additional feedback information, and how the reports may be used.   Details of reception reporting by translators and mixers is given in   Section 7.6.3.1 SR: Sender report RTCP 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+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+|V=2|P|    RC   |   PT=SR=200   |             length            | header+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+|                         SSRC of sender                        |+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+|              NTP timestamp, most significant word             | sender+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ info|             NTP timestamp, least significant word             |+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+|                         RTP timestamp                         |+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+|                     sender's packet count                     |+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+|                      sender's octet count                     |+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+|                 SSRC_1 (SSRC of first source)                 | report+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ block| fraction lost |       cumulative number of packets lost       |   1-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+|           extended highest sequence number received           |+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+|                      interarrival jitter                      |+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+|                         last SR (LSR)                         |+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+|                   delay since last SR (DLSR)                  |+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+|                 SSRC_2 (SSRC of second source)                | report+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ block:                               ...                             :   2+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+|                  profile-specific extensions                  |+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+   The sender report packet consists of three sections, possibly   followed by a fourth profile-specific extension section if defined.   The first section, the header, is 8 octets long. The fields have the   following meaning:Schulzrinne, et al          Standards Track                    [Page 23]RFC 1889                          RTP                       January 1996   version (V): 2 bits        Identifies the version of RTP, which is the same in RTCP packets        as in RTP data packets. The version defined by this        specification is two (2).   padding (P): 1 bit        If the padding bit is set, this RTCP packet contains some        additional padding octets at the end which are not part of the        control information. The last octet of the padding is a count of        how many padding octets should be ignored. Padding may be needed        by some encryption algorithms with fixed block sizes. In a        compound RTCP packet, padding should only be required on the        last individual packet because the compound packet is encrypted        as a whole.   reception report count (RC): 5 bits        The number of reception report blocks contained in this packet.        A value of zero is valid.   packet type (PT): 8 bits        Contains the constant 200 to identify this as an RTCP SR packet.   length: 16 bits        The length of this RTCP packet in 32-bit words minus one,        including the header and any padding. (The offset of one makes        zero a valid length and avoids a possible infinite loop in        scanning a compound RTCP packet, while counting 32-bit words        avoids a validity check for a multiple of 4.)   SSRC: 32 bits        The synchronization source identifier for the originator of this        SR packet.   The second section, the sender information, is 20 octets long and is   present in every sender report packet. It summarizes the data   transmissions from this sender. The fields have the following   meaning:   NTP timestamp: 64 bits        Indicates the wallclock time when this report was sent so that        it may be used in combination with timestamps returned in        reception reports from other receivers to measure round-trip        propagation to those receivers. Receivers should expect that the        measurement accuracy of the timestamp may be limited to far less        than the resolution of the NTP timestamp. The measurement        uncertainty of the timestamp is not indicated as it may not be        known. A sender that can keep track of elapsed time but has no        notion of wallclock time may use the elapsed time since joiningSchulzrinne, et al          Standards Track                    [Page 24]RFC 1889                          RTP                       January 1996        the session instead. This is assumed to be less than 68 years,        so the high bit will be zero. It is permissible to use the        sampling clock to estimate elapsed wallclock time. A sender that        has no notion of wallclock or elapsed time may set the NTP        timestamp to zero.   RTP timestamp: 32 bits        Corresponds to the same time as the NTP timestamp (above), but        in the same units and with the same random offset as the RTP        timestamps in data packets. This correspondence may be used for        intra- and inter-media synchronization for sources whose NTP        timestamps are synchronized, and may be used by media-        independent receivers to estimate the nominal RTP clock        frequency. Note that in most cases this timestamp will not be        equal to the RTP timestamp in any adjacent data packet. Rather,        it is calculated from the corresponding NTP timestamp using the        relationship between the RTP timestamp counter and real time as        maintained by periodically checking the wallclock time at a        sampling instant.   sender's packet count: 32 bits        The total number of RTP data packets transmitted by the sender        since starting transmission up until the time this SR packet was        generated.  The count is reset if the sender changes its SSRC        identifier.   sender's octet count: 32 bits        The total number of payload octets (i.e., not including header        or padding) transmitted in RTP data packets by the sender since        starting transmission up until the time this SR packet was        generated. The count is reset if the sender changes its SSRC        identifier. This field can be used to estimate the average        payload data rate.   The third section contains zero or more reception report blocks   depending on the number of other sources heard by this sender since   the last report. Each reception report block conveys statistics on   the reception of RTP packets from a single synchronization source.   Receivers do not carry over statistics when a source changes its SSRC   identifier due to a collision. These statistics are:   SSRC_n (source identifier): 32 bits        The SSRC identifier of the source to which the information in        this reception report block pertains.   fraction lost: 8 bits        The fraction of RTP data packets from source SSRC_n lost since        the previous SR or RR packet was sent, expressed as a fixedSchulzrinne, et al          Standards Track                    [Page 25]RFC 1889                          RTP                       January 1996        point number with the binary point at the left edge of the        field. (That is equivalent to taking the integer part after        multiplying the loss fraction by 256.) This fraction is defined        to be the number of packets lost divided by the number of        packets expected,  as defined in the next paragraph.  An        implementation is shown in Appendix A.3. If the loss is negative        due to duplicates, the fraction lost is set to zero. Note that a        receiver cannot tell whether any packets were lost after the        last one received, and that there will be no reception report        block issued for a source if all packets from that source sent        during the last reporting interval have been lost.   cumulative number of packets lost: 24 bits        The total number of RTP data packets from source SSRC_n that        have been lost since the beginning of reception. This number is        defined to be the number of packets expected less the number of        packets actually received, where the number of packets received        includes any which are late or duplicates. Thus packets that        arrive late are not counted as lost, and the loss may be        negative if there are duplicates.  The number of packets        expected is defined to be the extended last sequence number        received, as defined next, less the initial sequence number        received. This may be calculated as shown in Appendix A.3.   extended highest sequence number received: 32 bits        The low 16 bits contain the highest sequence number received in        an RTP data packet from source SSRC_n, and the most significant        16 bits extend that sequence number with the corresponding count        of sequence number cycles, which may be maintained according to        the algorithm in Appendix A.1. Note that different receivers        within the same session will generate different extensions to        the sequence number if their start times differ significantly.   interarrival jitter: 32 bits        An estimate of the statistical variance of the RTP data packet        interarrival time, measured in timestamp units and expressed as        an unsigned integer. The interarrival jitter J is defined to be        the mean deviation (smoothed absolute value) of the difference D        in packet spacing at the receiver compared to the sender for a        pair of packets. As shown in the equation below, this is        equivalent to the difference in the "relative transit time" for        the two packets; the relative transit time is the difference        between a packet's RTP timestamp and the receiver's clock at the        time of arrival, measured in the same units.Schulzrinne, et al          Standards Track                    [Page 26]RFC 1889                          RTP                       January 1996   If Si is the RTP timestamp from packet i, and Ri is the time of   arrival in RTP timestamp units for packet i, then for two packets i   and j, D may be expressed as                 D(i,j)=(Rj-Ri)-(Sj-Si)=(Rj-Sj)-(Ri-Si)   The interarrival jitter is calculated continuously as each data   packet i is received from source SSRC_n, using this difference D for   that packet and the previous packet i-1 in order of arrival (not   necessarily in sequence), according to the formula                    J=J+(|D(i-1,i)|-J)/16   Whenever a reception report is issued, the current value of J is   sampled.   The jitter calculation is prescribed here to allow profile-   independent monitors to make valid interpretations of reports coming   from different implementations. This algorithm is the optimal first-   order estimator and the gain parameter 1/16 gives a good noise   reduction ratio while maintaining a reasonable rate of convergence   [11].  A sample implementation is shown in Appendix A.8.   last SR timestamp (LSR): 32 bits        The middle 32 bits out of 64 in the NTP timestamp (as explained        in Section 4) received as part of the most recent RTCP sender        report (SR) packet from source SSRC_n.  If no SR has been        received yet, the field is set to zero.   delay since last SR (DLSR): 32 bits        The delay, expressed in units of 1/65536 seconds, between        receiving the last SR packet from source SSRC_n and sending this        reception report block.  If no SR packet has been received yet        from SSRC_n, the DLSR field is set to zero.   Let SSRC_r denote the receiver issuing this receiver report. Source   SSRC_n can compute the round propagation delay to SSRC_r by recording   the time A when this reception report block is received.  It   calculates the total round-trip time A-LSR using the last SR   timestamp (LSR) field, and then subtracting this field to leave the   round-trip propagation delay as (A- LSR - DLSR).  This is illustrated   in Fig. 2.   This may be used as an approximate measure of distance to cluster   receivers, although some links have very asymmetric delays.Schulzrinne, et al          Standards Track                    [Page 27]RFC 1889                          RTP                       January 19966.3.2 RR: Receiver report RTCP packet   [10 Nov 1995 11:33:25.125]           [10 Nov 1995 11:33:36.5]   n                 SR(n)              A=b710:8000 (46864.500 s)   ---------------------------------------------------------------->                      v                 ^   ntp_sec =0xb44db705 v               ^ dlsr=0x0005.4000 (    5.250s)   ntp_frac=0x20000000  v             ^  lsr =0xb705:2000 (46853.125s)     (3024992016.125 s)  v           ^   r                      v         ^ RR(n)   ---------------------------------------------------------------->                          |<-DLSR->|                           (5.250 s)   A     0xb710:8000 (46864.500 s)   DLSR -0x0005:4000 (    5.250 s)   LSR  -0xb705:2000 (46853.125 s)   -------------------------------   delay 0x   6:2000 (    6.125 s)           Figure 2: Example for round-trip time computation 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|    RC   |   PT=RR=201   |             length            | header+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+|                     SSRC of packet sender                     |+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+|                 SSRC_1 (SSRC of first source)                 | report+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ block| fraction lost |       cumulative number of packets lost       |   1+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+|           extended highest sequence number received           |+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+|                      interarrival jitter                      |+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+|                         last SR (LSR)                         |+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+|                   delay since last SR (DLSR)                  |+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+|                 SSRC_2 (SSRC of second source)                | report+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ block:                               ...                             :   2+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+|                  profile-specific extensions                  |+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+Schulzrinne, et al          Standards Track                    [Page 28]RFC 1889                          RTP                       January 1996   The format of the receiver report (RR) packet is the same as that of   the SR packet except that the packet type field contains the constant   201 and the five words of sender information are omitted (these are   the NTP and RTP timestamps and sender's packet and octet counts). The   remaining fields have the same meaning as for the SR packet.   An empty RR packet (RC = 0) is put at the head of a compound RTCP   packet when there is no data transmission or reception to report.6.3.3 Extending the sender and receiver reports   A profile should define profile- or application-specific extensions   to the sender report and receiver if there is additional information   that should be reported regularly about the sender or receivers. This   method should be used in preference to defining another RTCP packet   type because it requires less overhead:        o fewer octets in the packet (no RTCP header or SSRC field);        o simpler and faster parsing because applications running under         that profile would be programmed to always expect the extension         fields in the directly accessible location after the reception         reports.   If additional sender information is required, it should be included   first in the extension for sender reports, but would not be present   in receiver reports. If information about receivers is to be   included, that data may be structured as an array of blocks parallel   to the existing array of reception report blocks; that is, the number   of blocks would be indicated by the RC field.6.3.4 Analyzing sender and receiver reports   It is expected that reception quality feedback will be useful not   only for the sender but also for other receivers and third-party   monitors.  The sender may modify its transmissions based on the   feedback; receivers can determine whether problems are local,   regional or global; network managers may use profile-independent   monitors that receive only the RTCP packets and not the corresponding   RTP data packets to evaluate the performance of their networks for   multicast distribution.   Cumulative counts are used in both the sender information and   receiver report blocks so that differences may be calculated between   any two reports to make measurements over both short and long time   periods, and to provide resilience against the loss of a report. The   difference between the last two reports received can be used to   estimate the recent quality of the distribution. The NTP timestamp isSchulzrinne, et al          Standards Track                    [Page 29]RFC 1889                          RTP                       January 1996   included so that rates may be calculated from these differences over   the interval between two reports. Since that timestamp is independent   of the clock rate for the data encoding, it is possible to implement   encoding- and profile-independent quality monitors.   An example calculation is the packet loss rate over the interval   between two reception reports. The difference in the cumulative   number of packets lost gives the number lost during that interval.   The difference in the extended last sequence numbers received gives   the number of packets expected during the interval. The ratio of   these two is the packet loss fraction over the interval. This ratio   should equal the fraction lost field if the two reports are   consecutive, but otherwise not. The loss rate per second can be   obtained by dividing the loss fraction by the difference in NTP   timestamps, expressed in seconds. The number of packets received is   the number of packets expected minus the number lost. The number of   packets expected may also be used to judge the statistical validity   of any loss estimates.  For example, 1 out of 5 packets lost has a   lower significance than 200 out of 1000.   From the sender information, a third-party monitor can calculate the   average payload data rate and the average packet rate over an   interval without receiving the data. Taking the ratio of the two   gives the average payload size. If it can be assumed that packet loss   is independent of packet size, then the number of packets received by   a particular receiver times the average payload size (or the   corresponding packet size) gives the apparent throughput available to   that receiver.   In addition to the cumulative counts which allow long-term packet   loss measurements using differences between reports, the fraction   lost field provides a short-term measurement from a single report.   This becomes more important as the size of a session scales up enough   that reception state information might not be kept for all receivers   or the interval between reports becomes long enough that only one   report might have been received from a particular receiver.   The interarrival jitter field provides a second short-term measure of   network congestion. Packet loss tracks persistent congestion while   the jitter measure tracks transient congestion. The jitter measure   may indicate congestion before it leads to packet loss. Since the   interarrival jitter field is only a snapshot of the jitter at the   time of a report, it may be necessary to analyze a number of reports   from one receiver over time or from multiple receivers, e.g., within   a single network.Schulzrinne, et al          Standards Track                    [Page 30]RFC 1889                          RTP                       January 19966.4 SDES: Source description RTCP 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+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+|V=2|P|    SC   |  PT=SDES=202  |             length            | header+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+|                          SSRC/CSRC_1                          | chunk+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+   1|                           SDES items                          ||                              ...                              |+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+|                          SSRC/CSRC_2                          | chunk+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+   2|                           SDES items                          ||                              ...                              |+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+   The SDES packet is a three-level structure composed of a header and   zero or more chunks, each of of which is composed of items describing   the source identified in that chunk. The items are described   individually in subsequent sections.   version (V), padding (P), length:        As described for the SR packet (see Section 6.3.1).   packet type (PT): 8 bits        Contains the constant 202 to identify this as an RTCP SDES        packet.   source count (SC): 5 bits        The number of SSRC/CSRC chunks contained in this SDES packet. A        value of zero is valid but useless.   Each chunk consists of an SSRC/CSRC identifier followed by a list of   zero or more items, which carry information about the SSRC/CSRC. Each   chunk starts on a 32-bit boundary. Each item consists of an 8-bit   type field, an 8-bit octet count describing the length of the text   (thus, not including this two-octet header), and the text itself.   Note that the text can be no longer than 255 octets, but this is   consistent with the need to limit RTCP bandwidth consumption.   The text is encoded according to the UTF-2 encoding specified in   Annex F of ISO standard 10646 [12,13]. This encoding is also known as   UTF-8 or UTF-FSS. It is described in "File System Safe UCS   Transformation Format (FSS_UTF)", X/Open Preliminary Specification,   Document Number P316 and Unicode Technical Report #4. US-ASCII is a   subset of this encoding and requires no additional encoding. TheSchulzrinne, et al          Standards Track                    [Page 31]RFC 1889                          RTP                       January 1996   presence of multi-octet encodings is indicated by setting the most   significant bit of a character to a value of one.   Items are contiguous, i.e., items are not individually padded to a   32-bit boundary. Text is not null terminated because some multi-octet   encodings include null octets. The list of items in each chunk is   terminated by one or more null octets, the first of which is   interpreted as an item type of zero to denote the end of the list,   and the remainder as needed to pad until the next 32-bit boundary. A   chunk with zero items (four null octets) is valid but useless.   End systems send one SDES packet containing their own source   identifier (the same as the SSRC in the fixed RTP header). A mixer   sends one SDES packet containing a chunk for each contributing source   from which it is receiving SDES information, or multiple complete   SDES packets in the format above if there are more than 31 such   sources (see Section 7).   The SDES items currently defined are described in the next sections.   Only the CNAME item is mandatory. Some items shown here may be useful   only for particular profiles, but the item types are all assigned   from one common space to promote shared use and to simplify profile-   independent applications. Additional items may be defined in a   profile by registering the type numbers with IANA.6.4.1 CNAME: Canonical end-point identifier SDES item    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   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+   |    CNAME=1    |     length    | user and domain name         ...   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+   The CNAME identifier has the following properties:        o Because the randomly allocated SSRC identifier may change if a         conflict is discovered or if a program is restarted, the CNAME         item is required to provide the binding from the SSRC         identifier to an identifier for the source that remains         constant.        o Like the SSRC identifier, the CNAME identifier should also be         unique among all participants within one RTP session.        o To provide a binding across multiple media tools used by one         participant in a set of related RTP sessions, the CNAME should         be fixed for that participant.Schulzrinne, et al          Standards Track                    [Page 32]RFC 1889                          RTP                       January 1996        o To facilitate third-party monitoring, the CNAME should be         suitable for either a program or a person to locate the source.   Therefore, the CNAME should be derived algorithmically and not   entered manually, when possible. To meet these requirements, the   following format should be used unless a profile specifies an   alternate syntax or semantics. The CNAME item should have the format   "user@host", or "host" if a user name is not available as on single-   user systems.  For both formats, "host" is either the fully qualified   domain name of the host from which the real-time data originates,   formatted according to the rules specified in RFC 1034 [14], RFC 1035   [15] and Section 2.1 of RFC 1123 [16]; or the standard ASCII   representation of the host's numeric address on the interface used   for the RTP communication. For example, the standard ASCII   representation of an IP Version 4 address is "dotted decimal", also   known as dotted quad. Other address types are expected to have ASCII   representations that are mutually unique.  The fully qualified domain   name is more convenient for a human observer and may avoid the need   to send a NAME item in addition, but it may be difficult or   impossible to obtain reliably in some operating environments.   Applications that may be run in such environments should use the   ASCII representation of the address instead.   Examples are "doe@sleepy.megacorp.com" or "doe@192.0.2.89" for a   multi-user system. On a system with no user name, examples would be   "sleepy.megacorp.com" or "192.0.2.89".   The user name should be in a form that a program such as "finger" or   "talk" could use, i.e., it typically is the login name rather than   the personal name. The host name is not necessarily identical to the   one in the participant's electronic mail address.   This syntax will not provide unique identifiers for each source if an   application permits a user to generate multiple sources from one   host.  Such an application would have to rely on the SSRC to further   identify the source, or the profile for that application would have   to specify additional syntax for the CNAME identifier.   If each application creates its CNAME independently, the resulting   CNAMEs may not be identical as would be required to provide a binding   across multiple media tools belonging to one participant in a set of   related RTP sessions. If cross-media binding is required, it may be   necessary for the CNAME of each tool to be externally configured with   the same value by a coordination tool.   Application writers should be aware that private network address   assignments such as the Net-10 assignment proposed in RFC 1597 [17]   may create network addresses that are not globally unique. This wouldSchulzrinne, et al          Standards Track                    [Page 33]RFC 1889                          RTP                       January 1996   lead to non-unique CNAMEs if hosts with private addresses and no   direct IP connectivity to the public Internet have their RTP packets   forwarded to the public Internet through an RTP-level translator.   (See also RFC 1627 [18].) To handle this case, applications may   provide a means to configure a unique CNAME, but the burden is on the   translator to translate CNAMEs from private addresses to public   addresses if necessary to keep private addresses from being exposed.6.4.2 NAME: User name SDES item    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   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+   |     NAME=2    |     length    | common name of source        ...   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+   This is the real name used to describe the source, e.g., "John Doe,   Bit Recycler, Megacorp". It may be in any form desired by the user.   For applications such as conferencing, this form of name may be the   most desirable for display in participant lists, and therefore might   be sent most frequently of those items other than CNAME. Profiles may   establish such priorities.  The NAME value is expected to remain   constant at least for the duration of a session. It should not be   relied upon to be unique among all participants in the session.6.4.3 EMAIL: Electronic mail address SDES item    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   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+   |    EMAIL=3    |     length    | email address of source      ...   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+   The email address is formatted according to RFC 822 [19], for   example, "John.Doe@megacorp.com". The EMAIL value is expected to   remain constant for the duration of a session.6.4.4 PHONE: Phone number SDES item    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   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+   |    PHONE=4    |     length    | phone number of source       ...   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+   The phone number should be formatted with the plus sign replacing the   international access code.  For example, "+1 908 555 1212" for a   number in the United States.Schulzrinne, et al          Standards Track                    [Page 34]RFC 1889                          RTP                       January 19966.4.5 LOC: Geographic user location SDES item    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   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+   |     LOC=5     |     length    | geographic location of site  ...   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+   Depending on the application, different degrees of detail are   appropriate for this item. For conference applications, a string like   "Murray Hill, New Jersey" may be sufficient, while, for an active   badge system, strings like "Room 2A244, AT&T BL MH" might be   appropriate. The degree of detail is left to the implementation   and/or user, but format and content may be prescribed by a profile.   The LOC value is expected to remain constant for the duration of a   session, except for mobile hosts.6.4.6 TOOL: Application or tool name SDES item    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   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+   |     TOOL=6    |     length    | name/version of source appl. ...   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+   A string giving the name and possibly version of the application   generating the stream, e.g., "videotool 1.2". This information may be   useful for debugging purposes and is similar to the Mailer or Mail-   System-Version SMTP headers. The TOOL value is expected to remain   constant for the duration of the session.6.4.7 NOTE: Notice/status SDES item    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   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+   |     NOTE=7    |     length    | note about the source        ...   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+   The following semantics are suggested for this item, but these or   other semantics may be explicitly defined by a profile. The NOTE item   is intended for transient messages describing the current state of   the source, e.g., "on the phone, can't talk". Or, during a seminar,   this item might be used to convey the title of the talk. It should be   used only to carry exceptional information and should not be included   routinely by all participants because this would slow down the rate   at which reception reports and CNAME are sent, thus impairing the   performance of the protocol. In particular, it should not be includedSchulzrinne, et al          Standards Track                    [Page 35]RFC 1889                          RTP                       January 1996   as an item in a user's configuration file nor automatically generated   as in a quote-of-the-day.   Since the NOTE item may be important to display while it is active,   the rate at which other non-CNAME items such as NAME are transmitted   might be reduced so that the NOTE item can take that part of the RTCP   bandwidth. When the transient message becomes inactive, the NOTE item   should continue to be transmitted a few times at the same repetition   rate but with a string of length zero to signal the receivers.   However, receivers should also consider the NOTE item inactive if it   is not received for a small multiple of the repetition rate, or   perhaps 20-30 RTCP intervals.6.4.8 PRIV: Private extensions SDES item      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     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+     |     PRIV=8    |     length    | prefix length | prefix string...     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+    ...              |                  value string                ...     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+   This item is used to define experimental or application-specific SDES   extensions. The item contains a prefix consisting of a length-string   pair, followed by the value string filling the remainder of the item   and carrying the desired information. The prefix length field is 8   bits long. The prefix string is a name chosen by the person defining   the PRIV item to be unique with respect to other PRIV items this   application might receive. The application creator might choose to   use the application name plus an additional subtype identification if   needed.  Alternatively, it is recommended that others choose a name   based on the entity they represent, then coordinate the use of the   name within that entity.   Note that the prefix consumes some space within the item's total   length of 255 octets, so the prefix should be kept as short as   possible. This facility and the constrained RTCP bandwidth should not   be overloaded; it is not intended to satisfy all the control   communication requirements of all applications.   SDES PRIV prefixes will not be registered by IANA. If some form of   the PRIV item proves to be of general utility, it should instead be   assigned a regular SDES item type registered with IANA so that no   prefix is required. This simplifies use and increases transmission   efficiency.Schulzrinne, et al          Standards Track                    [Page 36]RFC 1889                          RTP                       January 19966.5 BYE: Goodbye RTCP 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   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+   |V=2|P|    SC   |   PT=BYE=203  |             length            |   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+   |                           SSRC/CSRC                           |   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+   :                              ...                              :   +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+   |     length    |               reason for leaving             ... (opt)   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+   The BYE packet indicates that one or more sources are no longer   active.   version (V), padding (P), length:        As described for the SR packet (see Section 6.3.1).   packet type (PT): 8 bits        Contains the constant 203 to identify this as an RTCP BYE        packet.   source count (SC): 5 bits        The number of SSRC/CSRC identifiers included in this BYE packet.        A count value of zero is valid, but useless.   If a BYE packet is received by a mixer, the mixer forwards the BYE   packet with the SSRC/CSRC identifier(s) unchanged. If a mixer shuts   down, it should send a BYE packet listing all contributing sources it   handles, as well as its own SSRC identifier. Optionally, the BYE   packet may include an 8-bit octet count followed by that many octets   of text indicating the reason for leaving, e.g., "camera malfunction"   or "RTP loop detected". The string has the same encoding as that   described for SDES. If the string fills the packet to the next 32-bit   boundary, the string is not null terminated. If not, the BYE packet   is padded with null octets.Schulzrinne, et al          Standards Track                    [Page 37]RFC 1889                          RTP                       January 19966.6 APP: Application-defined RTCP 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   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+   |V=2|P| subtype |   PT=APP=204  |             length            |   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+   |                           SSRC/CSRC                           |   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+   |                          name (ASCII)                         |   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+   |                   application-dependent data                 ...   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+   The APP packet is intended for experimental use as new applications   and new features are developed, without requiring packet type value   registration. APP packets with unrecognized names should be ignored.   After testing and if wider use is justified, it is recommended that   each APP packet be redefined without the subtype and name fields and   registered with the Internet Assigned Numbers Authority using an RTCP   packet type.   version (V), padding (P), length:        As described for the SR packet (see Section 6.3.1).   subtype: 5 bits        May be used as a subtype to allow a set of APP packets to be        defined under one unique name, or for any application-dependent        data.   packet type (PT): 8 bits        Contains the constant 204 to identify this as an RTCP APP        packet.   name: 4 octets        A name chosen by the person defining the set of APP packets to        be unique with respect to other APP packets this application        might receive. The application creator might choose to use the        application name, and then coordinate the allocation of subtype        values to others who want to define new packet types for the        application.  Alternatively, it is recommended that others        choose a name based on the entity they represent, then        coordinate the use of the name within that entity. The name is        interpreted as a sequence of four ASCII characters, with        uppercase and lowercase characters treated as distinct.Schulzrinne, et al          Standards Track                    [Page 38]RFC 1889                          RTP                       January 1996   application-dependent data: variable length        Application-dependent data may or may not appear in an APP        packet. It is interpreted by the application and not RTP itself.        It must be a multiple of 32 bits long.7.  RTP Translators and Mixers   In addition to end systems, RTP supports the notion of "translators"   and "mixers", which could be considered as "intermediate systems" at   the RTP level. Although this support adds some complexity to the   protocol, the need for these functions has been clearly established   by experiments with multicast audio and video applications in the   Internet. Example uses of translators and mixers given in Section 2.3   stem from the presence of firewalls and low bandwidth connections,   both of which are likely to remain.7.1 General Description   An RTP translator/mixer connects two or more transport-level   "clouds".  Typically, each cloud is defined by a common network and   transport protocol (e.g., IP/UDP), multicast address or pair of   unicast addresses, and transport level destination port.  (Network-   level protocol translators, such as IP version 4 to IP version 6, may   be present within a cloud invisibly to RTP.) One system may serve as   a translator or mixer for a number of RTP sessions, but each is   considered a logically separate entity.   In order to avoid creating a loop when a translator or mixer is   installed, the following rules must be observed:        o Each of the clouds connected by translators and mixers         participating in one RTP session either must be distinct from         all the others in at least one of these parameters (protocol,         address, port), or must be isolated at the network level from         the others.        o A derivative of the first rule is that there must not be         multiple translators or mixers connected in parallel unless by         some arrangement they partition the set of sources to be         forwarded.   Similarly, all RTP end systems that can communicate through one or   more RTP translators or mixers share the same SSRC space, that is,   the SSRC identifiers must be unique among all these end systems.   Section 8.2 describes the collision resolution algorithm by which   SSRC identifiers are kept unique and loops are detected.Schulzrinne, et al          Standards Track                    [Page 39]RFC 1889                          RTP                       January 1996   There may be many varieties of translators and mixers designed for   different purposes and applications. Some examples are to add or   remove encryption, change the encoding of the data or the underlying   protocols, or replicate between a multicast address and one or more   unicast addresses. The distinction between translators and mixers is   that a translator passes through the data streams from different   sources separately, whereas a mixer combines them to form one new   stream:   Translator: Forwards RTP packets with their SSRC identifier intact;        this makes it possible for receivers to identify individual        sources even though packets from all the sources pass through        the same translator and carry the translator's network source        address. Some kinds of translators will pass through the data        untouched, but others may change the encoding of the data and        thus the RTP data payload type and timestamp. If multiple data        packets are re-encoded into one, or vice versa, a translator        must assign new sequence numbers to the outgoing packets. Losses        in the incoming packet stream may induce corresponding gaps in        the outgoing sequence numbers. Receivers cannot detect the        presence of a translator unless they know by some other means        what payload type or transport address was used by the original        source.   Mixer: Receives streams of RTP data packets from one or more sources,        possibly changes the data format, combines the streams in some        manner and then forwards the combined stream. 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, so it is the        synchronization source. Thus, all data packets forwarded by a        mixer will be marked with the mixer's own SSRC identifier. In        order to preserve the identity of the original sources        contributing to the mixed packet, the mixer should insert their        SSRC identifiers into the CSRC identifier list following the        fixed RTP header of the packet. A mixer that is also itself a        contributing source for some packet should explicitly include        its own SSRC identifier in the CSRC list for that packet.   For some applications, it may be acceptable for a mixer not to   identify sources in the CSRC list. However, this introduces the   danger that loops involving those sources could not be detected.   The advantage of a mixer over a translator for applications like   audio is that the output bandwidth is limited to that of one source   even when multiple sources are active on the input side. This may be   important for low-bandwidth links. The disadvantage is that receivers   on the output side don't have any control over which sources areSchulzrinne, et al          Standards Track                    [Page 40]RFC 1889                          RTP                       January 1996   passed through or muted, unless some mechanism is implemented for   remote control of the mixer. The regeneration of synchronization   information by mixers also means that receivers can't do inter-media   synchronization of the original streams. A multi-media mixer could do   it.         [E1]                                    [E6]          |                                       |    E1:17 |                                 E6:15 |          |                                       |   E6:15          V  M1:48 (1,17)         M1:48 (1,17)    V   M1:48 (1,17)         (M1)------------->----------------->-------------->[E7]          ^                 ^     E4:47           ^   E4:47     E2:1 |           E4:47 |                     |   M3:89 (64,45)          |                 |                     |         [E2]              [E4]     M3:89 (64,45) |                                                  |        legend:   [E3] --------->(M2)----------->(M3)------------|        [End system]          E3:64        M2:12 (64)  ^                       (Mixer)                                   | E5:45                                                    |                                  [E5]          source: SSRC (CSRCs)                                                -------------------> Figure 3: Sample RTP network with end systems, mixers and translators   A collection of mixers and translators is shown in Figure 3 to   illustrate their effect on SSRC and CSRC identifiers. In the figure,   end systems are shown as rectangles (named E), translators as   triangles (named T) and mixers as ovals (named M). The notation "M1:   48(1,17)" designates a packet originating a mixer M1, identified with   M1's (random) SSRC value of 48 and two CSRC identifiers, 1 and 17,   copied from the SSRC identifiers of packets from E1 and E2.7.2 RTCP Processing in Translators   In addition to forwarding data packets, perhaps modified, translators   and mixers must also process RTCP packets. In many cases, they will   take apart the compound RTCP packets received from end systems to   aggregate SDES information and to modify the SR or RR packets.   Retransmission of this information may be triggered by the packet   arrival or by the RTCP interval timer of the translator or mixer   itself.   A translator that does not modify the data packets, for example one   that just replicates between a multicast address and a unicast   address, may simply forward RTCP packets unmodified as well. ASchulzrinne, et al          Standards Track                    [Page 41]RFC 1889                          RTP                       January 1996   translator that transforms the payload in some way must make   corresponding transformations in the SR and RR information so that it   still reflects the characteristics of the data and the reception   quality. These translators must not simply forward RTCP packets. In   general, a translator should not aggregate SR and RR packets from   different sources into one packet since that would reduce the   accuracy of the propagation delay measurements based on the LSR and   DLSR fields.   SR sender information:  A translator does not generate its own sender        information, but forwards the SR packets received from one cloud        to the others. The SSRC is left intact but the sender        information must be modified if required by the translation. If        a translator changes the data encoding, it must change the        "sender's byte count" field. If it also combines several data        packets into one output packet, it must change the "sender's        packet count" field. If it changes the timestamp frequency, it        must change the "RTP timestamp" field in the SR packet.   SR/RR reception report blocks:  A translator forwards reception        reports received from one cloud to the others. Note that these        flow in the direction opposite to the data.  The SSRC is left        intact. If a translator combines several data packets into one        output packet, and therefore changes the sequence numbers, it        must make the inverse manipulation for the packet loss fields        and the "extended last sequence number" field. This may be        complex. In the extreme case, there may be no meaningful way to        translate the reception reports, so the translator may pass on        no reception report at all or a synthetic report based on its        own reception. The general rule is to do what makes sense for a        particular translation.   A translator does not require an SSRC identifier of its own, but may   choose to allocate one for the purpose of sending reports about what   it has received. These would be sent to all the connected clouds,   each corresponding to the translation of the data stream as sent to   that cloud, since reception reports are normally multicast to all   participants.   SDES:  Translators typically forward without change the SDES        information they receive from one cloud to the others, but may,        for example, decide to filter non-CNAME SDES information if        bandwidth is limited. The CNAMEs must be forwarded to allow SSRC        identifier collision detection to work. A translator that        generates its own RR packets must send SDES CNAME information        about itself to the same clouds that it sends those RR packets.Schulzrinne, et al          Standards Track                    [Page 42]RFC 1889                          RTP                       January 1996   BYE:  Translators forward BYE packets unchanged. Translators with        their own SSRC should generate BYE packets with that SSRC        identifier if they are about to cease forwarding packets.   APP:  Translators forward APP packets unchanged.7.3 RTCP Processing in Mixers   Since a mixer generates a new data stream of its own, it does not   pass through SR or RR packets at all and instead generates new   information for both sides.   SR sender information:  A mixer does not pass through sender        information from the sources it mixes because the        characteristics of the source streams are lost in the mix. As a        synchronization source, the mixer generates its own SR packets        with sender information about the mixed data stream and sends        them in the same direction as the mixed stream.   SR/RR reception report blocks:  A mixer generates its own reception        reports for sources in each cloud and sends them out only to the        same cloud. It does not send these reception reports to the        other clouds and does not forward reception reports from one        cloud to the others because the sources would not be SSRCs there        (only CSRCs).   SDES:  Mixers typically forward without change the SDES information        they receive from one cloud to the others, but may, for example,        decide to filter non-CNAME SDES information if bandwidth is        limited. The CNAMEs must be forwarded to allow SSRC identifier        collision detection to work. (An identifier in a CSRC list        generated by a mixer might collide with an SSRC identifier        generated by an end system.) A mixer must send SDES CNAME        information about itself to the same clouds that it sends SR or        RR packets.   Since mixers do not forward SR or RR packets, they will typically be   extracting SDES packets from a compound RTCP packet. To minimize   overhead, chunks from the SDES packets may be aggregated into a   single SDES packet which is then stacked on an SR or RR packet   originating from the mixer. The RTCP packet rate may be different on   each side of the mixer.   A mixer that does not insert CSRC identifiers may also refrain from   forwarding SDES CNAMEs. In this case, the SSRC identifier spaces in   the two clouds are independent. As mentioned earlier, this mode of   operation creates a danger that loops can't be detected.Schulzrinne, et al          Standards Track                    [Page 43]RFC 1889                          RTP                       January 1996   BYE:  Mixers need to forward BYE packets. They should generate BYE        packets with their own SSRC identifiers if they are about to        cease forwarding packets.   APP:  The treatment of APP packets by mixers is application-specific.7.4 Cascaded Mixers   An RTP session may involve a collection of mixers and translators as   shown in Figure 3. If two mixers are cascaded, such as M2 and M3 in   the figure, packets received by a mixer may already have been mixed   and may include a CSRC list with multiple identifiers. The second   mixer should build the CSRC list for the outgoing packet using the   CSRC identifiers from already-mixed input packets and the SSRC   identifiers from unmixed input packets. This is shown in the output   arc from mixer M3 labeled M3:89(64,45) in the figure. As in the case   of mixers that are not cascaded, if the resulting CSRC list has more   than 15 identifiers, the remainder cannot be included.8.  SSRC Identifier Allocation and Use   The SSRC identifier carried in the RTP header and in various fields   of RTCP packets is a random 32-bit number that is required to be   globally unique within an RTP session. It is crucial that the number   be chosen with care in order that participants on the same network or   starting at the same time are not likely to choose the same number.   It is not sufficient to use the local network address (such as an   IPv4 address) for the identifier because the address may not be   unique. Since RTP translators and mixers enable interoperation among   multiple networks with different address spaces, the allocation   patterns for addresses within two spaces might result in a much   higher rate of collision than would occur with random allocation.   Multiple sources running on one host would also conflict.   It is also not sufficient to obtain an SSRC identifier simply by   calling random() without carefully initializing the state. An example   of how to generate a random identifier is presented in Appendix A.6.8.1 Probability of Collision   Since the identifiers are chosen randomly, it is possible that two or   more sources will choose the same number. Collision occurs with the   highest probability when all sources are started simultaneously, for   example when triggered automatically by some session management   event. If N is the number of sources and L the length of the   identifier (here, 32 bits), the probability that two sourcesSchulzrinne, et al          Standards Track                    [Page 44]RFC 1889                          RTP                       January 1996   independently pick the same value can be approximated for large N   [20] as 1 - exp(-N**2 / 2**(L+1)). For N=1000, the probability is   roughly 10**-4.   The typical collision probability is much lower than the worst-case   above. When one new source joins an RTP session in which all the   other sources already have unique identifiers, the probability of   collision is just the fraction of numbers used out of the space.   Again, if N is the number of sources and L the length of the   identifier, the probability of collision is N / 2**L. For N=1000, the   probability is roughly 2*10**-7.   The probability of collision is further reduced by the opportunity   for a new source to receive packets from other participants before   sending its first packet (either data or control). If the new source   keeps track of the other participants (by SSRC identifier), then   before transmitting its first packet the new source can verify that   its identifier does not conflict with any that have been received, or   else choose again.8.2 Collision Resolution and Loop Detection   Although the probability of SSRC identifier collision is low, all RTP   implementations must be prepared to detect collisions and take the   appropriate actions to resolve them. If a source discovers at any   time that another source is using the same SSRC identifier as its   own, it must send an RTCP BYE packet for the old identifier and   choose another random one. If a receiver discovers that two other   sources are colliding, it may keep the packets from one and discard   the packets from the other when this can be detected by different   source transport addresses or CNAMEs. The two sources are expected to   resolve the collision so that the situation doesn't last.   Because the random identifiers are kept globally unique for each RTP   session, they can also be used to detect loops that may be introduced   by mixers or translators. A loop causes duplication of data and   control information, either unmodified or possibly mixed, as in the   following examples:        o A translator may incorrectly forward a packet to the same         multicast group from which it has received the packet, either         directly or through a chain of translators. In that case, the         same packet appears several times, originating from different         network sources.        o Two translators incorrectly set up in parallel, i.e., with the         same multicast groups on both sides, would both forward packets         from one multicast group to the other. UnidirectionalSchulzrinne, et al          Standards Track                    [Page 45]RFC 1889                          RTP                       January 1996         translators would produce two copies; bidirectional translators         would form a loop.        o A mixer can close a loop by sending to the same transport         destination upon which it receives packets, either directly or         through another mixer or translator. In this case a source         might show up both as an SSRC on a data packet and a CSRC in a         mixed data packet.   A source may discover that its own packets are being looped, or that   packets from another source are being looped (a third-party loop).   Both loops and collisions in the random selection of a source   identifier result in packets arriving with the same SSRC identifier   but a different source transport address, which may be that of the   end system originating the packet or an intermediate system.   Consequently, if a source changes its source transport address, it   must also choose a new SSRC identifier to avoid being interpreted as   a looped source. Loops or collisions occurring on the far side of a   translator or mixer cannot be detected using the source transport   address if all copies of the packets go through the translator or   mixer, however collisions may still be detected when chunks from two   RTCP SDES packets contain the same SSRC identifier but different   CNAMEs.   To detect and resolve these conflicts, an RTP implementation must   include an algorithm similar to the one described below. It ignores   packets from a new source or loop that collide with an established   source. It resolves collisions with the participant's own SSRC   identifier by sending an RTCP BYE for the old identifier and choosing   a new one. However, when the collision was induced by a loop of the   participant's own packets, the algorithm will choose a new identifier   only once and thereafter ignore packets from the looping source   transport address. This is required to avoid a flood of BYE packets.   This algorithm depends upon the source transport address being the   same for both RTP and RTCP packets from a source. The algorithm would   require modifications to support applications that don't meet this   constraint.   This algorithm requires keeping a table indexed by source identifiers   and containing the source transport address from which the identifier   was (first) received, along with other state for that source. Each   SSRC or CSRC identifier received in a data or control packet is   looked up in this table in order to process that data or control   information.  For control packets, each element with its own SSRC,   for example an SDES chunk, requires a separate lookup. (The SSRC in a   reception report block is an exception.) If the SSRC or CSRC is notSchulzrinne, et al          Standards Track                    [Page 46]RFC 1889                          RTP                       January 1996   found, a new entry is created. These table entries are removed when   an RTCP BYE packet is received with the corresponding SSRC, or after   no packets have arrived for a relatively long time (see Section   6.2.1).   In order to track loops of the participant's own data packets, it is   also necessary to keep a separate list of source transport addresses   (not identifiers) that have been found to be conflicting. Note that   this should be a short list, usually empty. Each element in this list   stores the source address plus the time when the most recent   conflicting packet was received. An element may be removed from the   list when no conflicting packet has arrived from that source for a   time on the order of 10 RTCP report intervals (see Section 6.2).   For the algorithm as shown, it is assumed that the participant's own   source identifier and state are included in the source identifier   table. The algorithm could be restructured to first make a separate   comparison against the participant's own source identifier.       IF the SSRC or CSRC identifier is not found in the source          identifier table:       THEN create a new entry storing the source transport address            and the SSRC or CSRC along with other state.            CONTINUE with normal processing.       (identifier is found in the table)       IF the source transport address from the packet matches          the one saved in the table entry for this identifier:       THEN CONTINUE with normal processing.       (an identifier collision or a loop is indicated)       IF the source identifier is not the participant's own:       THEN IF the source identifier is from an RTCP SDES chunk               containing a CNAME item that differs from the CNAME               in the table entry:            THEN (optionally) count a third-party collision.            ELSE (optionally) count a third-party loop.            ABORT processing of data packet or control element.       (a collision or loop of the participant's own data)       IF the source transport address is found in the list of         conflicting addresses:       THEN IF the source identifier is not from an RTCP SDES chunk               containing a CNAME item OR if that CNAME is the               participant's own:Schulzrinne, et al          Standards Track                    [Page 47]RFC 1889                          RTP                       January 1996            THEN (optionally) count occurrence of own traffic looped.                 mark current time in conflicting address list entry.                 ABORT processing of data packet or control element.       log occurrence of a collision.       create a new entry in the conflicting address list and       mark current time.       send an RTCP BYE packet with the old SSRC identifier.       choose a new identifier.       create a new entry in the source identifier table with the         old SSRC plus the source transport address from the packet         being processed.       CONTINUE with normal processing.   In this algorithm, packets from a newly conflicting source address   will be ignored and packets from the original source will be kept.   (If the original source was through a mixer and later the same source   is received directly, the receiver may be well advised to switch   unless other sources in the mix would be lost.) If no packets arrive   from the original source for an extended period, the table entry will   be timed out and the new source will be able to take over. This might   occur if the original source detects the collision and moves to a new   source identifier, but in the usual case an RTCP BYE packet will be   received from the original source to delete the state without having   to wait for a timeout.   When a new SSRC identifier is chosen due to a collision, the   candidate identifier should first be looked up in the source   identifier table to see if it was already in use by some other   source. If so, another candidate should be generated and the process   repeated.   A loop of data packets to a multicast destination can cause severe   network flooding. All mixers and translators are required to   implement a loop detection algorithm like the one here so that they   can break loops. This should limit the excess traffic to no more than   one duplicate copy of the original traffic, which may allow the   session to continue so that the cause of the loop can be found and   fixed. However, in extreme cases where a mixer or translator does not   properly break the loop and high traffic levels result, it may be   necessary for end systems to cease transmitting data or control   packets entirely. This decision may depend upon the application. An   error condition should be indicated as appropriate. Transmission   might be attempted again periodically after a long, random time (on   the order of minutes).Schulzrinne, et al          Standards Track                    [Page 48]RFC 1889                          RTP                       January 19969.  Security   Lower layer protocols may eventually provide all the security   services that may be desired for applications of RTP, including   authentication, integrity, and confidentiality. These services  have   recently been specified for IP. Since the need for a confidentiality   service is well established in the initial audio and video   applications that are expected to use RTP, a confidentiality service   is defined in the next section for use with RTP and RTCP until lower   layer services are available. The overhead on the protocol for this   service is low, so the penalty will be minimal if this service is   obsoleted by lower layer services in the future.   Alternatively, other services, other implementations of services and   other algorithms may be defined for RTP in the future if warranted.   The selection presented here is meant to simplify implementation of   interoperable, secure applications and provide guidance to   implementors. No claim is made that the methods presented here are   appropriate for a particular security need. A profile may specify   which services and algorithms should be offered by applications, and   may provide guidance as to their appropriate use.   Key distribution and certificates are outside the scope of this   document.9.1 Confidentiality   Confidentiality means that only the intended receiver(s) can decode   the received packets; for others, the packet contains no useful   information. Confidentiality of the content is achieved by   encryption.   When encryption of RTP or RTCP is desired, all the octets that will   be encapsulated for transmission in a single lower-layer packet are   encrypted as a unit. For RTCP, a 32-bit random number is prepended to   the unit before encryption to deter known plaintext attacks. For RTP,   no prefix is required because the sequence number and timestamp   fields are initialized with random offsets.   For RTCP, it is allowed to split a compound RTCP packet into two   lower-layer packets, one to be encrypted and one to be sent in the   clear. For example, SDES information might be encrypted while   reception reports were sent in the clear to accommodate third-party   monitors that are not privy to the encryption key. In this example,   depicted in Fig. 4, the SDES information must be appended to an RR   packet with no reports (and the encrypted) to satisfy the requirement   that all compound RTCP packets begin with an SR or RR packet.Schulzrinne, et al          Standards Track                    [Page 49]RFC 1889                          RTP                       January 1996                 UDP packet                        UDP packet   -------------------------------------  -------------------------   [32-bit ][       ][     #           ]  [    # sender # receiver]   [random ][  RR   ][SDES # CNAME, ...]  [ SR # report # report  ]   [integer][(empty)][     #           ]  [    #        #         ]   -------------------------------------  -------------------------                 encrypted                       not encrypted   #: SSRC           Figure 4: Encrypted and non-encrypted RTCP packets   The presence of encryption and the use of the correct key are   confirmed by the receiver through header or payload validity checks.   Examples of such validity checks for RTP and RTCP headers are given   in Appendices A.1 and A.2.   The default encryption algorithm is the Data Encryption Standard   (DES) algorithm in cipher block chaining (CBC) mode, as described in   Section 1.1 of RFC 1423 [21], except that padding to a multiple of 8   octets is indicated as described for the P bit in Section 5.1. The   initialization vector is zero because random values are supplied in   the RTP header or by the random prefix for compound RTCP packets. For   details on the use of CBC initialization vectors, see [22].   Implementations that support encryption should always support the DES   algorithm in CBC mode as the default to maximize interoperability.   This method is chosen because it has been demonstrated to be easy and   practical to use in experimental audio and video tools in operation   on the Internet. Other encryption algorithms may be specified   dynamically for a session by non-RTP means.   As an alternative to encryption at the RTP level as described above,   profiles may define additional payload types for encrypted encodings.   Those encodings must specify how padding and other aspects of the   encryption should be handled. This method allows encrypting only the   data while leaving the headers in the clear for applications where   that is desired. It may be particularly useful for hardware devices   that will handle both decryption and decoding.9.2 Authentication and Message Integrity   Authentication and message integrity are not defined in the current   specification of RTP since these services would not be directly   feasible without a key management infrastructure. It is expected that   authentication and integrity services will be provided by lower layer   protocols in the future.Schulzrinne, et al          Standards Track                    [Page 50]RFC 1889                          RTP                       January 199610.  RTP over Network and Transport Protocols   This section describes issues specific to carrying RTP packets within   particular network and transport protocols. The following rules apply   unless superseded by protocol-specific definitions outside this   specification.   RTP relies on the underlying protocol(s) to provide demultiplexing of   RTP data and RTCP control streams. For UDP and similar protocols, RTP   uses an even port number and the corresponding RTCP stream uses the   next higher (odd) port number. If an application is supplied with an   odd number for use as the RTP port, it should replace this number   with the next lower (even) number.   RTP data packets contain no length field or other delineation,   therefore RTP relies on the underlying protocol(s) to provide a   length indication. The maximum length of RTP packets is limited only   by the underlying protocols.   If RTP packets are to be carried in an underlying protocol that   provides the abstraction of a continuous octet stream rather than   messages (packets), an encapsulation of the RTP packets must be   defined to provide a framing mechanism. Framing is also needed if the   underlying protocol may contain padding so that the extent of the RTP   payload cannot be determined. The framing mechanism is not defined   here.   A profile may specify a framing method to be used even when RTP is   carried in protocols that do provide framing in order to allow   carrying several RTP packets in one lower-layer protocol data unit,   such as a UDP packet. Carrying several RTP packets in one network or   transport packet reduces header overhead and may simplify   synchronization between different streams.11.  Summary of Protocol Constants   This section contains a summary listing of the constants defined in   this specification.   The RTP payload type (PT) constants are defined in profiles rather   than this document. However, the octet of the RTP header which   contains the marker bit(s) and payload type must avoid the reserved   values 200 and 201 (decimal) to distinguish RTP packets from the RTCP   SR and RR packet types for the header validation procedure described   in Appendix A.1. For the standard definition of one marker bit and a   7-bit payload type field as shown in this specification, this   restriction means that payload types 72 and 73 are reserved.Schulzrinne, et al          Standards Track                    [Page 51]RFC 1889                          RTP                       January 199611.1 RTCP packet types   abbrev.    name                   value   SR         sender report            200   RR         receiver report          201   SDES       source description       202   BYE        goodbye                  203   APP        application-defined      204   These type values were chosen in the range 200-204 for improved   header validity checking of RTCP packets compared to RTP packets or   other unrelated packets. When the RTCP packet type field is compared   to the corresponding octet of the RTP header, this range corresponds   to the marker bit being 1 (which it usually is not in data packets)   and to the high bit of the standard payload type field being 1 (since   the static payload types are typically defined in the low half). This   range was also chosen to be some distance numerically from 0 and 255   since all-zeros and all-ones are common data patterns.   Since all compound RTCP packets must begin with SR or RR, these codes   were chosen as an even/odd pair to allow the RTCP validity check to   test the maximum number of bits with mask and value.   Other constants are assigned by IANA. Experimenters are encouraged to   register the numbers they need for experiments, and then unregister   those which prove to be unneeded.11.2 SDES types   abbrev.    name                              value   END        end of SDES list                      0   CNAME      canonical name                        1   NAME       user name                             2   EMAIL      user's electronic mail address        3   PHONE      user's phone number                   4   LOC        geographic user location              5   TOOL       name of application or tool           6   NOTE       notice about the source               7   PRIV       private extensions                    8   Other constants are assigned by IANA. Experimenters are encouraged to   register the numbers they need for experiments, and then unregister   those which prove to be unneeded.Schulzrinne, et al          Standards Track                    [Page 52]RFC 1889                          RTP                       January 199612.  RTP Profiles and Payload Format Specifications   A complete specification of RTP for a particular application will   require one or more companion documents of two types described here:   profiles, and payload format specifications.   RTP may be used for a variety of applications with somewhat differing   requirements. The flexibility to adapt to those requirements is   provided by allowing multiple choices in the main protocol   specification, then selecting the appropriate choices or defining   extensions for a particular environment and class of applications in   a separate profile document. Typically an application will operate   under only one profile so there is no explicit indication of which   profile is in use. A profile for audio and video applications may be   found in the companion Internet-Draft draft-ietf-avt-profile for   The second type of companion document is a payload format   specification, which defines how a particular kind of payload data,   such as H.261 encoded video, should be carried in RTP. These   documents are typically titled "RTP Payload Format for XYZ   Audio/Video Encoding". Payload formats may be useful under multiple   profiles and may therefore be defined independently of any particular   profile. The profile documents are then responsible for assigning a   default mapping of that format to a payload type value if needed.   Within this specification, the following items have been identified   for possible definition within a profile, but this list is not meant   to be exhaustive:   RTP data header: The octet in the RTP data header that contains the        marker bit and payload type field may be redefined by a profile        to suit different requirements, for example with more or fewer        marker bits (Section 5.3).   Payload types: Assuming that a payload type field is included, the        profile will usually define a set of payload formats (e.g.,        media encodings) and a default static mapping of those formats        to payload type values. Some of the payload formats may be        defined by reference to separate payload format specifications.        For each payload type defined, the profile must specify the RTP        timestamp clock rate to be used (Section 5.1).   RTP data header additions: Additional fields may be appended to the        fixed RTP data header if some additional functionality is        required across the profile's class of applications independent        of payload type (Section 5.3).Schulzrinne, et al          Standards Track                    [Page 53]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 1996A.  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    /*    * 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, theSchulzrinne, 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 maximumSchulzrinne, 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 duplicateSchulzrinne, 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    /* u_long */   #include     /* gettimeofday() */   #include       /* get..() */   #include        /* printf() */   #include         /* clock() */   #include  /* 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    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 thatSchulzrinne, 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.comSchulzrinne, 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.govAcknowledgments   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.  [23] Rivest, R., "The MD5 Message-Digest Algorithm", RFC 1321, MIT       Laboratory for Computer Science and RSA Data Security, Inc.,       April 1992.  [24] S. Stubblebine, "Security services for multimedia conferencing,"       in 16th National Computer Security Conference , (Baltimore,       Maryland), pp. 391--395, Sept. 1993.  [25] S. Floyd and V. Jacobson, "The synchronization of periodic       routing messages," IEEE/ACM Transactions on Networking , vol. 2,       pp.  122-136, April 1994.Schulzrinne, et al          Standards Track                    [Page 75]