Internet Engineering Task Force Audio-Video Transport WG INTERNET-DRAFT H. Schulzrinne/S. Casner draft-ietf-avt-rtp-04.txt AT&T/ISI October 20, 1993 Expires: 12/31/93 RTP: A Transport Protocol for Real-Time Applications Status of this Memo This document is an Internet Draft. Internet Drafts are working documents of the Internet Engineering Task Force (IETF), its Areas, and its Working Groups. Note that other groups may also distribute working documents as Internet Drafts. Internet Drafts are draft documents valid for a maximum of six months. Internet Drafts may be updated, replaced, or obsoleted by other documents at any time. It is not appropriate to use Internet Drafts as reference material or to cite them other than as a ``working draft'' or ``work in progress.'' Please check the I-D abstract listing contained in each Internet Draft directory to learn the current status of this or any other Internet Draft. Distribution of this document is unlimited. Abstract This memorandum describes the real-time transport protocol, RTP. 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) designed to provide minimal control and identification functionality particularly in multicast networks. Within multicast associations, sites can also direct control messages to individual sites. 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 bridges. This specification is a product of the Audio/Video Transport working group within the Internet Engineering Task Force. Comments are solicited and INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 should be addressed to the working group's mailing list at rem-conf@es.net and/or the authors. Contents 1 Introduction 3 2 RTP Use Scenarios 5 2.1 Simple Multicast Audio Conference. . . . . . . . . . . . . . . . . 5 2.2 Bridges. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6 2.3 Translators. . . . . . . . . . . . . . . . . . . . . . . . . . . . 6 2.4 Security . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7 3 Definitions 8 4 Byte Order, Alignment, and Reserved Values 10 5 RTP Data Transfer Protocol 11 5.1 RTP Fixed Header Fields. . . . . . . . . . . . . . . . . . . . . . 11 5.2 The RTP Options. . . . . . . . . . . . . . . . . . . . . . . . . . 13 5.2.1CSRC: Content source identifiers. . . . . . . . . . . . . . . . 13 5.2.2SSRC: Synchronization source identifier . . . . . . . . . . . . 14 5.2.3BOS: Beginning of synchronization unit. . . . . . . . . . . . . 15 5.3 APP: Application-specific option . . . . . . . . . . . . . . . . . 15 5.4 Reverse-Path Option. . . . . . . . . . . . . . . . . . . . . . . . 16 5.4.1SDST: Synchronization destination identifier. . . . . . . . . . 17 5.5 Security Options . . . . . . . . . . . . . . . . . . . . . . . . . 18 5.5.1ENC: Encryption . . . . . . . . . . . . . . . . . . . . . . . . 21 5.5.2MIC: Messsage integrity check . . . . . . . . . . . . . . . . . 22 5.5.3MICA: Message integrity check, asymmetric encryption. . . . . . 22 H. Schulzrinne/S. Casner Expires 12/31/93 [Page 2]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 5.5.4MICK: Message integrity check, keyed. . . . . . . . . . . . . . 23 5.5.5MICS: Message integrity check, symmetric-key encrypted. . . . . 24 6 RTP Control Protocol --- RTCP 25 6.1 FMT: Format description. . . . . . . . . . . . . . . . . . . . . . 25 6.2 SDES: Source descriptor. . . . . . . . . . . . . . . . . . . . . . 26 6.3 BYE: Goodbye . . . . . . . . . . . . . . . . . . . . . . . . . . . 30 6.4 QOS: Quality of service measurement. . . . . . . . . . . . . . . . 30 7 Security Considerations 32 8 RTP over Network and Transport Protocols 33 8.1 Defaults . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33 8.2 ST-II. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33 9 RTP Profiles 34 A Implementation Notes 35 A.1 Timestamp Recovery . . . . . . . . . . . . . . . . . . . . . . . . 36 A.2 Detecting the Beginning of a Synchronization Unit. . . . . . . . . 37 A.3 Demultiplexing and Locating the Synchronization Source . . . . . . 38 A.4 Parsing RTP Options. . . . . . . . . . . . . . . . . . . . . . . . 38 A.5 Determining the Expected Number of RTP Packets . . . . . . . . . . 39 B Addresses of Authors 40 1 Introduction This memorandum specifies the real-time transport protocol (RTP), which provides end-to-end delivery services for data with real-time characteristics, for example, interactive audio and video. 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. H. Schulzrinne/S. Casner Expires 12/31/93 [Page 3]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 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 end system 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. RTP is designed to run on top of a variety of network and transport protocols, for example, IP, ST-II or UDP.(1) RTP transfers data in a single direction, possibly to multiple destinations if supported by the underlying network. A mechanism for sending control data in the opposite direction, reversing the path traversed by regular data, is provided. 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. Profiles are used to instantiate certain header fields and options for particular sets of applications (see Section 9). A profile for audio and video data may be found in the companion Internet draft draft-ietf-avt-profile. This document defines a packet format shared by two protocols: o the real-time transport protocol (RTP), for exchanging data that has real-time properties. The RTP header consists of a fixed-length portion plus optional control fields; o the RTP control protocol (RTCP), for conveying information about the participants in an on-going session. RTCP consists of additional header options that may be ignored without affecting the ability to receive data correctly. RTCP is used for "loosely controlled" sessions, i.e., where there is no explicit membership control and set-up. Its functionality may be subsumed by a session control protocol, which is beyond the scope of this document. A discussion of real-time services and algorithms for their implementation and background on some of the RTP design decisions can be found in the current version of the companion Internet draft draft-ietf-avt-issues. The current Internet does not support the widespread use of real-time services. High-bandwidth services using RTP, such as video, can potentially seriously degrade 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-time services on the ------------------------------ 1. For most applications, RTP offers insufficient demultiplexing to run directly on IP. H. Schulzrinne/S. Casner Expires 12/31/93 [Page 4]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 Internet and other network services. 2 RTP Use Scenarios The following sections describe some aspects of the use of RTP. The examples were chosen to illustrate the basic operation of applications using RTP, not to limit what RTP may be used for. In these examples, RTP is carried on top of IP and UDP, and follows the conventions established by the profile for audio and video specified in the companion Internet draft draft-ietf-avt-profile. 2.1 Simple Multicast Audio Conference A working group of the IETF meets to discuss the latest protocol draft, using the IP multicast services of the Internet for voice communications. Through some allocation mechanism, the working group chair obtains a multicast group address; all participants use the destination UDP port specified by the profile. The multicast address and port are distributed, say, by electronic mail, to all intended participants. The mechanisms for discovering available multicast addresses and distributing the information to participants 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 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 seen by the source, so that, in our case, a chunk of audio is delivered to the speaker every 20 ms. The sequence number can also be used by the receiver to estimate how many packets are being lost. Each RTP packet also indicates what type of audio encoding (such as PCM, ADPCM or GSM) is being used, 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. Since members of the working group join and leave during the conference, it is useful to know who is participating at any moment. For that purpose, each instance of the audio application in the conference periodically multicasts the name, email address and other information of its user. Such control information is carried as RTCP SDES options within RTP messages, with or without audio data (see Section 6.2). These periodic messages also provide some indication as to whether the network connection is still functioning. A site sends the RTCP BYE (Section 6.3) option when it leaves a conference. The RTCP QOS (Section 6.4) option indicates how well H. Schulzrinne/S. Casner Expires 12/31/93 [Page 5]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 the current speaker is being received and may be used to control adaptive encodings. 2.2 Bridges So far, we have assumed that all sites want to receive audio 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, a bridge is placed near the low-bandwidth area. This bridge resynchronizes incoming audio packets to reconstruct the constant 20 ms spacing generated by the sender, mixes these reconstructed audio streams, translates the audio encoding to a lower-bandwidth one and forwards the lower-bandwidth packet stream to the low-bandwidth sites. After the mixing, the identity of the high-speed site that is speaking can no longer be determined from the network origin of the packet. Therefore, the bridge inserts a CSRC option (Section 5.2.1) into the packet containing a list of short, locally unique site identifiers to indicate which site(s) contributed to that mixed packet. An example of this is shown for bridge B1 in Fig. 1. As name and location information is received by the bridge in SDES options from the high-speed sites, that information is passed on to the receivers along with a mapping to the CSRC identifiers. This also works if an RTP packet is mixed through several bridges, with the CSRC value being mapped into a new locally unique value at each bridge. For example, in Fig. 1 bridge B3 maps CSRC value 3 for packets coming from B2 into CSRC value 1 for packets going to T2. 2.3 Translators Not all sites are directly accessible through IP multicast. For these sites, mixing may not necessary, but a translation of the underlying transport protocol is. RTP-level gateways that do not restore timing or mix packets from different sources are called translators in this document. Application-level firewalls, for example, will not let any IP packets pass. Two translators are installed, one on either side of the firewall, the outside one funneling all multicast packets received through the 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. Other examples include the connection of a group of hosts speaking only IP/UDP to a group of hosts that understand only ST-II. After RTP packets have passed through a translator, they all carry the network source address of the translator, making it impossible for the receiver to distinguish packets from different speakers based on network H. Schulzrinne/S. Casner Expires 12/31/93 [Page 6]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 [E1] [E6] | | E1:17 | E6:15 | | | E6:63/6 V B1:48 (1,2) B1:28/1 (1,2) V B1:63/5 (1,2) (B1)-------------><T1>-----------------><T2>--------------->[E7] ^ ^ E4:28/2 ^ E4:63/3 E2:1 | E4:47 | | B3:63/4 (1,4) | | | [E2] [E4] B3:89 (1,4) | | LEGEND: [E3] --------->(B2)----------->(B3)------------| [End system] E3:64 B2:12 (3) ^ (Bridge) | E5:45 <Translator> | [E5] source: port/SSRC (CSRCs) ------------------------> Figure 1: Sample RTP network with end systems, bridges and translators source addresses. Since each sending site has its own sequence number space and slightly offset timestamp space, the receiver could not properly mix the audio packets. (For video, it could not properly separate them into distinct displays.) Instead of forcing all senders to include some globally unique identifier in each packet, a translator inserts an SSRC option (Section 5.2.2) with a short identifier for the source that is locally unique to the translator. This also works if an RTP packet has to travel through several translators, with the SSRC value being mapped into a new locally unique value at each translator. An example is shown in Fig. 1, where hosts T1 and T2 are translators. The RTP packets from host E4 are identified with SSRC value 2, while those coming from bridge B1 are labeled with SSRC value 1. Similarly, translator T2 has labeled packets from E6, B1, E4 and B3 with SSRC values 6, 5, 3 and 4, respectively. 2.4 Security Conference participants would often like to ensure that nobody else can listen to their deliberations. Encryption, indicated by the presence of the ENC option (Section 5.5.1), provides that privacy. The encryption method and key can be changed during the conference by indexing into a table. For example, a meeting may go into executive session, protected by a different encryption key accessible only to a subset of the meeting participants. For authentication, a number of methods are provided, depending on needs and computational capabilities. All these message integrity check (MIC) options (Sections 5.5.3 and following) compute cryptographic checksums, also known H. Schulzrinne/S. Casner Expires 12/31/93 [Page 7]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 as message digests, over the RTP data. 3 Definitions Payload is the data following the RTP fixed header and any RTP/RTCP options. The payload format and interpretation are beyond the scope of this memo. RTP packets without payload are valid. Examples of payload include audio samples and video data. An RTP packet consists of the encapsulation specific to a particular underlying protocol, the fixed RTP header, RTP and RTCP options, if any, and the payload, if any. A single packet of the underlying protocol may contain several RTP packets if permitted by the encapsulation method. A (protocol) port is 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." [1] The transport selectors (TSEL) used by the OSI transport layer are equivalent to ports. A transport address denotes the combination of network address, e.g., the 4-octet IP Version 4 address, and the transport protocol port, e.g., the UDP port. In OSI systems, the transport address is called transport service access point or TSAP. The destination transport address may be a unicast or multicast address. A content source is the actual source of the data carried in an RTP packet, for example, the application that originally generated some audio data. Data from one or more content sources may be combined into a single RTP packet by a bridge, which becomes the synchronization source (see next paragraph). Content source identifiers carried in CSRC options identify the logical source of the data, for example, to highlight the current speaker in an audio conference; they have no effect on the delivery or playout timing of the data itself. In Fig. 1, E1 and E2 are the content sources of the data received by E7 from bridge B1, while B1 is the synchronization source. A synchronization source may be a single content source, or the combination of one or more content sources, produced by a bridge, with its own timing. Each synchronization source has its own sequence number space. The audio coming from a single microphone and the video from a camera are examples of synchronization sources. The receiver groups packets by synchronization source for playback. Typically a single synchronization source emits a single medium (e.g., audio or video). A synchronization source may change its data format, e.g., audio encoding, over time. Synchronization sources are identified by their transport address and the identifier carried in the SSRC option. If the SSRC option is absent, a value of zero is assumed for that identifier. H. Schulzrinne/S. Casner Expires 12/31/93 [Page 8]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 A transport source is the transport-level origin of the RTP packets as seen by the receiving end system. In Fig. 1, host T2, port 63 is the transport source of all packets received by end system E7. A channel comprises all synchronization sources sending to the same destination transport address using the same RTP channel ID. An end system generates the content to be sent in RTP packets and consumes the content of received RTP. An end system can act as one or more synchronization sources. (Most end systems are expected to be a single synchronization source.) When a packet is transmitted from an end system, the end system is the content source, synchronization source, and transport source at that point. An (RTP-level) bridge receives RTP packets from one or more sources, combines them in some manner and then forwards a new RTP packet. A bridge may change the data format. Since the timing among multiple input sources will not generally be synchronized, the bridge will make timing adjustments among the streams and generate its own timing for the combined stream. Therefore, when a packet is processed through a bridge, the bridge becomes the synchronization source as well as the transport source, but the originating end system remains the content source for that data. As the bridge combines packets from multiple content sources into a single outgoing packet, each of the contributing content sources is noted by the insertion of an identifier into the CSRC option in the outgoing packet. Audio bridges and media converters are examples of bridges. In Fig. 1, end systems E1 and E2 use the services of bridge B1. B1 inserts CSRC identifiers for E1 and E2 when they are active (e.g., talking in an audio conference). The RTP-level bridges described in this document are unrelated to the data link-layer bridges found in local area networks. If there is possibility for confusion, the term `RTP-level bridge' should be used. The name bridge follows common telecommunication industry usage. An (RTP-level) translator forwards RTP packets, but does not alter their sequence numbers or timestamps. Examples of its use include encoding conversion without mixing or retiming, conversion from multicast to unicast, and application-level filters in firewalls. A translator is neither a synchronization nor a content source, but does become the transport source for packets which flow through it. The properties of bridges and translators are summarized in Table 1. Checkmarks in parentheses designate possible, but unlikely actions. The RTP options are explained in Section 5.2, the RTCP options in Section 6. A synchronization unit consists of one or more packets that are emitted contiguously by the sender. The most common synchronization units are talkspurts for voice and frames for video transmission. During playout synchronization, the receiver must reconstruct exactly the time difference between packets within a synchronization unit (in the case of video, all the packets of a frame are typically given the same timestamp so there is no time difference). The time difference between synchronization units may be changed by the receiver to compensate for clock drift or to adjust to H. Schulzrinne/S. Casner Expires 12/31/93 [Page 9]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 end sys. bridge translator mix sources -- x -- change encoding -- x x encrypt x x (x) sign for authentication x x -- alter content x x x insert CSRC (RTP) -- x -- insert SSRC (RTP) (x) (x) x insert SDST (RTP) x x -- insert SDES (RTCP) x x -- Table 1: The properties of end systems, bridges and translators changing network delay jitter. For example, if audio packets are generated at fixed intervals during talkspurts, the receiver tries to play back packets with exactly the same spacing. However, if, for example, a silence period between synchronization units (talkspurts) lasts 600 ms, the receiver may adjust it to, say, 500 ms without this being noticed by the listener. Non-RTP mechanisms refers to other protocols and mechanisms that may be needed to provide a useable service. In particular, for multimedia conferences, a conference control application may distribute multicast addresses and keys for encryption and authentication, negotiate the encryption algorithm to be used, and determine the mapping from the RTP format field to the actual data format used. For simple applications, electronic mail or a conference database may also be used. The specification of such mechanisms is outside the scope of this memorandum. 4 Byte Order, Alignment, and Reserved Values 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 [2], Appendix A. Unless otherwise noted, numeric constants are in decimal (base 10). Numeric constants prefixed by `0x' are in hexadecimal. Fields within the fixed header and within options are aligned to the natural length of the field, i.e., 16-bit words are aligned on even addresses, 32-bit long words are aligned at addresses divisible by four, etc. Octets designated as padding have the value zero. Fields designated as "reserved" or R are set aside for future use; they should be set to zero by senders and ignored by receivers. Textual information is encoded accorded to the UTF-2 encoding of the ISO H. Schulzrinne/S. Casner Expires 12/31/93 [Page 10]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 standard 10646 (Annex F) [3,4]. US-ASCII is a subset of this encoding and requires no additional encoding. The presence of multi-octet encodings is indicated by setting the most significant bit to a value of one. An octet with a binary value of zero may be used as a string terminator for padding purposes. However, strings are not required to be zero terminated. 5 RTP Data Transfer Protocol 5.1 RTP Fixed Header Fields The RTP header has the following format: 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |Ver| ChannelID |P|S| format | sequence number | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | timestamp (seconds) | timestamp (fraction) | +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ | options ... | +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ The first eight octets are present in every RTP packet and have the following meaning: protocol version: 2 bits Identifies the protocol version. The version number of the protocol defined in this memo is one (1). channel ID: 6 bits The channel identifier field forms part of the tuple identifying a channel (see definition in Section 3) to provide an additional level of multiplexing at the RTP layer. The channel ID field is convenient if several different channels are to receive the same treatment by the underlying layers or if a profile allows for the concatenation of several RTP packets on different channels into a single packet of the underlying protocol layer (see Section 8.1). option present bit (P): 1 bit This flag has a value of one (1) if the fixed RTP header is followed by one or more options and a value of zero (0) otherwise. end-of-synchronization-unit (S): 1 bit This flag has a value of one in the last packet of a synchronization H. Schulzrinne/S. Casner Expires 12/31/93 [Page 11]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 unit, a value of zero otherwise. As shown in Appendix A, the beginning of a synchronization unit can be readily established from this flag.(2) format: 6 bits The format field forms an index into a table defined through the RTCP FMT option or non-RTP mechanisms (see Section 3). The mapping establishes the format of the RTP payload and determines its interpretation by the application. Formats defined through the FMT option must be kept in a separate mapping table per sender as there can be no guarantee that all senders will use the same table. If no mapping has been defined through these mechanisms, a standard mapping is specified by the profile in use by the application in question. An initial set of default mappings for audio and video is specified in the companion profile document RFC TBD, and may be extended in future editions of the Assigned Numbers RFC. sequence number: 16 bits The sequence number counts RTP packets. The sequence number increments by one for each packet sent. The sequence number may be used by the receiver to detect packet loss, to restore packet sequence and to identify packets to the application. timestamp: 32 bits The timestamp reflects the wall clock time when the RTP packet was generated. Several consecutive RTP packets may have equal timestamps if they are generated at once. The timestamp consists of the middle 32 bits of a 64-bit NTP timestamp, as defined in RFC 1305 [5]. That is, it counts time since 0 hours UTC, January 1, 1900, with a resolution of 65536 ticks per second. (UTC is Coordinated Universal Time, approximately equal to the historical Greenwich Mean Time.) The RTP timestamp wraps around approximately every 18 hours. The timestamp of the first packet within a synchronization unit is expected to closely reflect the actual sampling instant, measured by the local system clock. If possible, the local system clock should be controlled by a time synchronization protocol such as NTP. However, it is allowable to operate without synchronized time on those systems where it is not available, unless a profile or session protocol requires otherwise. It is not necessary to reference the local system clock to obtain the timestamp for the beginning of every synchronization unit, but the local clock should be referenced frequently enough so that clock drift between the synchronized system clock and the sampling clock can be compensated for gradually. Within one synchronization unit, it may be appropriate to compute timestamps based on the logical timing relationships between the packets. For audio samples, for example, it is more accurate to maintain the time ------------------------------ 2. If this flag were to signal the beginning of a synchronization unit instead, the end of a synchronization unit could not be established in real time. H. Schulzrinne/S. Casner Expires 12/31/93 [Page 12]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 within a synchronization unit in samples, incrementing by the number of samples per packet, and then converting to an RTP timestamp (see Appendix A.1). 5.2 The RTP Options The RTP fixed packet header may be followed by options and then the payload. Each option consists of the F (final) bit, the option type designation, a one-octet length field denoting the total number of 32-bit words comprising the option (including F bit, type and length), followed by any option-specific data. The last option before the payload has the F bit set to one; for all other options this bit has a value of zero. An application may discard options with types unknown to it. The option type number range is divided into four regions. Types 0 through 31 are general RTP options, their syntax and semantics independent of the format or any profile. Types 32 through 63 are RTCP options, again independent of format and profile. Types 64 through 95 are specific to a particular profile, i.e., valid for a range of formats. Types 96 through 126 are specific to a format as defined by the four-character name registered with the Internet Assigned Numbers Authority (IANA); these options may be described in a profile or format specification. Format-specific options are parsed according to the format selected by the format field in the fixed RTP header, as shown in Appendix A.4. Types 0 through 126 are reserved and registered with IANA. Type 127 is defined in Section 5.3 of this document; it allows application-specific extensions not registered with IANA. Unless otherwise noted, each option may appear only once per packet. Each packet may contain any number of options. Options may appear in any order, unless specifically restricted by the option description. In particular, the position of some security options has significance. 5.2.1 CSRC: Content source identifiers 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 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |F| CSRC = 0 | length | content source identifiers ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ The CSRC option, inserted only by bridges, lists all sources that contributed to the packet. For example, for audio packets, all sources that were mixed together to create a packet are enumerated, allowing correct talker indication at the receiver. The CSRC option may contain one or more 16-bit content source identifiers. The identifier values must be unique for all content sources received from a particular synchronization source on a particular channel; the value of binary zero is reserved and may not be H. Schulzrinne/S. Casner Expires 12/31/93 [Page 13]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 used. If the number of content sources is even, the two octets needed to pad the list to a multiple of four octets are set to zero. There should be no more than one CSRC option within a packet. If no CSRC option is present, the content source identifier is assumed to have a value of zero. CSRC options are not modified by translators. If a bridge receives a packet containing a CSRC option from another bridge located upstream, the identifier values in that CSRC option must be translated into new, locally unique values. A conformant RTP implementation does not have to be able to generate or interpret the CSRC option. 5.2.2 SSRC: Synchronization source identifier 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 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |F| SSRC = 1 | length = 1 | identifier | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ The SSRC option may be inserted by translators, end systems and bridges. It is typically used only by translators, but it may be used by an end system application to distinguish several sources sent with the same transport source address. If packets from multiple synchronization sources will be transmitted with the same transport source address (e.g., the same IP address and UDP port), an SSRC option must be inserted in each packet with a distinct identifier for the synchronization source. Conversely, synchronization sources that are distinguishable by their transport address do not require the use of SSRC options. The SSRC value zero is reserved and would not normally be transmitted; if received, the SSRC option should be treated as if not present. When no SSRC option is present, the transport source address is assumed to indicate the synchronization source. There must be no more than one SSRC option per packet; thus, a translator must remap the SSRC identifier of an incoming packet into a new, locally unique SSRC identifier. The SSRC option can be viewed as an extension of the source port number in protocols like UDP, ST-II or TCP. An RTP receiver must support the SSRC option. RTP senders only need to support this option if they intend to send more than one source to the same channel using the same source port. For example, a translator could use multiple source ports rather than insert SSRC options, but this is likely to be less convenient. H. Schulzrinne/S. Casner Expires 12/31/93 [Page 14]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 5.2.3 BOS: Beginning of synchronization unit 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 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |F| BOS = 3 | length = 1 | sequence number | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ The sequence number contained within this option is that of the first packet within the current synchronization unit. The BOS option allows the receiver to compute the offset of a packet with respect to the beginning of the synchronization unit, even if the last packet of the previous synchronization unit was lost. It is expected that many applications will be able to tolerate such a loss, and so will not use the BOS option but rely on the S bit. Those applications which do require the BOS option may use a profile that specifies it is always included. 5.3 APP: Application-specific option 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 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |F| APP = 127 | length | subtype | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | name (ASCII) | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | application-dependent data ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ subtype: 2 octets May be used as a subtype to allow a set of APP options to be defined under one unique name, or for any application-dependendent data. name: 4 octets A name chosen by the person defining the set of APP options to be unique with respect to other APP options 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 options 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. application-dependent data: variable length Application-dependent data may or may not appear in an APP option. It H. Schulzrinne/S. Casner Expires 12/31/93 [Page 15]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 is interpreted by the application and not RTP itself. The APP option is intended for experimental use as new applications and new features are developed, without requiring option type value registration. APP options with unrecognized names should be ignored. After testing and if wider use is justified, it is recommended that each APP option be redefined without the subtype and name fields and registered with the Internet Assigned Numbers Authority using an option type in the RTP, RTCP, profile-specific, or format-specific range as appropriate. 5.4 Reverse-Path Option With two-party (unicast) communications, having a receiver of data relay back control information to the sender is straightforward. Similarly, for multicast communications, control information can easily be sent to all members of the group. It may, however, be desirable to send a unicast message to a single member of a multicast group, for example to send a reception quality report. For such purposes, RTP includes a mechanism for sending so-called reverse RTP packets. The format of reverse RTP packets is exactly the same as for regular RTP packets and they can make use of any option defined in this memorandum, except SSRC, as appropriate. The support for and semantics of particular options are to be specified by a profile or an individual application. Additional options may be defined as prescribed in Section 5.2 as needed for a particular profile, format or application. Reverse RTP packets travel through the same translators as forward RTP packets. When RTP is carried in a protocol that provides transport-level addressing (ports), a site may distinguish reverse RTP packets from forward RTP packets by their arrival port. Reverse RTP packets arrive on the same port that the site uses as a source port for forward (data) RTP packets. Therefore, that port should be assigned uniquely; in particular, it should be different than the destination port used with the multicast address, and if the application is participating in several multicast groups, a distinct source port should be used to send to each group. If RTP is carried directly within IP or some other network-layer protocol that does not include port numbers, the reverse RTP packet must include an SDST option (defined next), and the presence of the SDST option signals that the packet is a reverse RTP packet. A receiver of reverse RTP packets cannot rely on sequence numbers being consecutive, as a sender is allowed to use the same sequence number space while communicating through this reverse path with several sites. In particular, a receiver of reverse RTP packets cannot tell by the sequence numbers whether it has received all reverse RTP packets sent to it. As a consequence, it is expected that reverse RTP packets would carry only options and no payload. The sequence number space of reverse RTP packets has to be completely separate from that used for RTP packets sent to the H. Schulzrinne/S. Casner Expires 12/31/93 [Page 16]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 multicast group. If the same sequence number space were used, the members of the multicast group not receiving reverse RTP packets would detect a gap in their received sequence number space. The sender of reverse RTP packets should ensure that sequence numbers are unique, modulo wrap-around, so that they can, if necessary, be used for matching request and response. (Currently, no such request-response mechanism has been defined.) As a hypothetical example, consider defining a request to pan the remote video camera. After completing the request, the receiver of the request would send a generic acknowledgement containing the sequence number of the request back to the requestor as an option (not as the packet sequence number in the fixed header). The timestamp should reflect the approximate sending time of the packet. The channel ID must be the same as that used in the corresponding forward RTP packets. If many receivers send a reverse RTP packet in response to a stimulus in the data stream, for example a request for retransmission of a particular data frame, the simultaneous delivery of a large number of packets back to the data source can cause congestion for both the network and the destination (this is known as an "ack implosion"). Thus reverse RTP packets should be used with care, perhaps with mechanisms such as response rate limiting and random delays to spread out the simultaneous delivery. 5.4.1 SDST: Synchronization destination identifier 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 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |F| SDST = 2 | length = 1 | identifier | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ The SDST option is only inserted by RTP end systems and bridges if they want to send a unicast packet to a particular site within the multicast group. These are called reverse RTP packets. Only reverse RTP packets may include the SDST option, but not all reverse RTP packets require it, as explained below. A reverse RTP packet must not contain an SSRC option. If a forward RTP packet carries SSRC identifier X when sent from A to B, where A and B may be either two translators or an end system and a translator, the unicast reverse RTP packet will carry an SDST option with identifier X from B to A. Consider the topology shown in Fig. 1. Assume that RTP is carried over a network or transport protocol that includes port numbers and that all forward RTP packets are addressed to destination port 8000. For the case that B1 wants to send a reverse packet to E1, B1 simply sends to the source address and port, that is, port 17 in this example. E1 can tell by the arrival on port 17 that the packet is a reverse packet rather than a regular H. Schulzrinne/S. Casner Expires 12/31/93 [Page 17]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 (forward) packet. No SDST option is required. The mechanism is somewhat more complicated when translators intervene. We focus on end system E7. E7 receives, say, video from a range of sources, E1 through E6 as indicated by the arrows. The transmission from T2 to E7 could be either multicast or unicast. Assume that E7 wants to send a retransmission request, a request to pan the camera, etc., to end system E4 alone. E7 may not be able to directly reach E4, as E4 may be using a network protocol unknown to E7 or be located behind a firewall. According to the figure, video transmissions from E4 reach E7 through T2 with source port 63 and SSRC identifier 3. For the reverse message, E7 sends a message to T2, with destination port 63 and SDST identifier 3. T2 can look up in its table that it sends forward data coming from T1 with that SSRC identifier 3. T2 also knows that those messages from T1 carry SSRC 2 and arrive with source port 28. Just like E7, T2 places the SSRC identifier, 2 in this case, into the SDST option and forwards the packet to T1 at port 28. Finally, translator T1 consults its table to find that it labels packets coming from E4, port 47 with SSRC value 2 and thus knows to forward the reverse packet to E4, port 47. T1 can either place value zero into the SDST option or remove the option. Note that E4 cannot directly determine that E7 sent the reverse packet, rather than, say, E6. If that is important, a global identifier as defined for the QOS option (Section 6.4) needs to be included in some option in the reverse packet. When reverse RTP packets are carried directly within IP or some other network-layer protocol that does not include port numbers, the SDST option is required to distinguish reverse RTP packets from forward RTP packets. In the case where no SSRC identifier needs to be placed in the SDST option, the value zero should be inserted. Only applications that need to send or receive reverse control RTP packets need to implement the SDST option. 5.5 Security Options The security options below offer message integrity, authentication and confidentiality and the combination of the three. Support for the security options is not mandatory, but the encryption option (ENC) should at least be recognized to avoid processing encrypted data. The four message integrity check options --- MIC, MICA, MICK and MICS --- are mutually exclusive, i.e., only one of them should be used in a single RTP packet. Multiple options are provided to satisfy varying security requirements and computational capabilities. A variety of security services may be provided with the encryption option, one of the message integrity check options, or the combination of the two options: H. Schulzrinne/S. Casner Expires 12/31/93 [Page 18]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 confidentiality: Confidentiality means that only the intended receiver(s) can decode the received RTP packets; for others, the RTP packet contains no useful information. Confidentiality of the content is achieved by encryption. The presence of encryption and the encryption initialization vector is indicated by the ENC option.(3) authentication and message integrity: These two security services are provided by the message integrity check options. The receiver can ascertain that the claimed originator is indeed the originator of the data (authentication) and that the data within an RTP packet has not been altered after leaving the sender (message integrity). Through examination of the timestamp and sequence number fields in the RTP header to verify that all the packets of a sequence are present and played in order, an implementation may also establish the integrity of that packet sequence. The services offered by MICA and MIC/MICK/MICS differ: With MIC/MICK/MICS, the receiver can only verify that the message originated within the group holding the secret key, rather than authenticate the sender of the message. The MICA option, in combination with certificates(4), affords true authentication of the sender. The certificates for MICA must be distributed through means outside of RTP. authentication, message integrity, and confidentiality: By carrying both the message integrity check and ENC option in RTP packets, the authenticity, message integrity and confidentiality of the packet can be assured (subject to the limitations discussed in the previous paragraph). For this combination of security features when group authentication is sufficient, the combination ENC and MIC is recommended (instead of MICS or MICK), as it yields the lowest processing overhead. All message integrity check options carry a message digest, which is a cryptographic hash function that transforms a message of any length to a fixed-length byte string, where the fixed-length string has the property that it is computationally infeasible to generate another, different message with the same digest. The message digest is computed over the fixed header, ------------------------------ 3. For efficiency reasons, this specification does not insist that content encryption only be used in conjunction with message integrity and authentication mechanisms, in which case there is no guarantee that the encrypted data has not been replayed or rearranged. This also means the receiving program may not be able to readily determine whether the data has been successfully decrypted, but in most cases, it will be obvious to the person receiving the data if he or she does not possess the right encryption key. 4. For a description of certificates see, for example, RFC 1422 or [6]. H. Schulzrinne/S. Casner Expires 12/31/93 [Page 19]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 a portion of the message integrity check option itself (the first two octets for MICA, the first four octets for MIC and MICS, or the whole option in the case of MICK), and the remaining header options and payload that will immediately follow the message integrity check option in the RTP packet. The fixed header is protected to foil replay attacks and reassignment to a different channel. When both a message integrity check option and the ENC option are to be included, the recommended ordering is that the message integrity check be applied first as described in the previous paragraph. Then the message integrity check option and the remaining header options and payload that follow it are encrypted using the shared secret key. The ENC option is prepended to the encrypted data; that is, the ENC option must be followed immediately by the message integrity check option, without any other options in between. The receiver first decrypts the octets following the ENC option and then authenticates the decrypted data using the message digest contained in the message integrity check option. For the MIC option in particular, this ordering must be used because the ENC option is required to provide confidentiality of the message digest. For the other message integrity check options, this ordering allows explicit detection of an encryption key mismatch. However, both the decryption and message integrity check functions must be performed before an invalid packet can be detected, which increases the potential for a denial-of-service attack. For those applications where this is a concern, the ordering may be reversed. The message integrity check options and the ENC option must not cover the SSRC or SDST option, i.e., SSRC or SDST must be inserted between the fixed header and the ENC or message integrity check option; SSRC and SDST are subject to change by translators that likely do not possess the necessary descriptor table (see below) and encryption keys. Trusted translators that have the necessary keys and descriptor translation table may modify the contents of the RTP packet, unless the MICA option is used (see MICA description in Section 5.5.3). All security options except MICA carry a one-octet descriptor field. These descriptors are indexes into two tables, one for the message integrity check options and one for the ENC option, established by non-RTP means to contain the digest algorithms (MD2, MD5, etc.), encryption algorithms (DES variants) and encryption keys or shared secrets (for the MICK option) to be used during a session. The descriptor value may change during a session, for example, to switch to a different encryption key. The tables must be established to be the same for all sources within the same channel; this reduces per-site state information. The descriptor value zero selects a set of default algorithms, namely, MD5 for the message digest algorithm and DES in CBC mode for encryption algorithm, so that basic security features may be implemented using simple non-RTP mechanisms to communicate a single shared secret (key). For example, the key might be communicated by telephone or (private) email and H. Schulzrinne/S. Casner Expires 12/31/93 [Page 20]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 entered manually. 5.5.1 ENC: Encryption 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 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |F| ENC = 8 | length = 3 | reserved | descriptor | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | DES (CBC) initialization vector, bytes 0 through 3 | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | DES (CBC) initialization vector, bytes 4 through 7 | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |F| ENC = 8 | length = 1 | reserved | descriptor | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ Every encrypted RTP packet must contain this option in one of the two forms shown. All octets in the packet following this option are encrypted, using the encryption key and symmetric encryption algorithm selected by the descriptor field. Note that the fixed header is specifically not encrypted because some fields must be interpreted by translators that will not have access to the key. The descriptor value may change over time to accommodate varying security requirements or limit the amount of ciphertext using the same key. For example, in a job interview conducted across a network, the candidate and interviewers could share one key, with a second key set aside for the interviewers only. For symmetric keys, source-specific keys offer no advantage. The descriptor value zero is reserved for a default mode using the Data Encryption Standard (DES) algorithm in CBC (cipher block chaining) mode, as described in Section 1.1 of RFC 1423 [7]. The padding specified in that section is to be used. In the first form of the ENC option, the 8-octet initialization vector (IV) is carried unencrypted within the option, but must be generated uniquely for each packet. In the second form (indicated by an option length of one), the ENC option does not contain an initialization vector and instead the fixed RTP header is used as the initialization vector. (Using the fixed RTP header as the initialization vector avoids regenerating the initialization vector for each packet and incurs less header overhead; it is unique for a period of at least 18 hours.) For details on the tradeoffs for CBC initialization vector use, see [8]. Support for encryption is not required. Implementations that do not support encryption should recognize the ENC option so that they can avoid processing encrypted messages and provide a meaningful failure indication. Implementations that support encryption should, at the minimum, always support the DES algorithm in CBC mode. H. Schulzrinne/S. Casner Expires 12/31/93 [Page 21]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 5.5.2 MIC: Messsage integrity check 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 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |F| MIC = 9 | length | reserved | descriptor | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | message digest (unencrypted) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ The MIC option option is used only in combination with the ENC option immediately preceding it to provide confidentiality, message integrity and group membership authentication. The message integrity check uses the digest algorithm selected by the descriptor field. The value zero implies the use of the MD5 message digest. Note that the MIC option is not separately encrypted. 5.5.3 MICA: Message integrity check, asymmetric encryption 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 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |F| MICA = 10 | length | message digest ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | (asymmetrically encrypted) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ The message digest is asymmetrically encrypted using the sender's private key according to the algorithm defined for Privacy Enhanced Mail, described in Section 4.2.1 of RFC 1423 (here "MIC" denotes the general term "message integrity check", not the RTP option): As described in PKCS #1, all quantities input as data values to the RSAEncryption process shall be properly justified and padded to the length of the modulus prior to the encryption process. In general, an RSAEncryption input value is formed by concatenating a leading NULL octet, a block type BT, a padding string PS, a NULL octet, and the data quantity D, that is, RSA input value = 0x00, BT, PS, 0x00, D. To prepare a MIC for RSAEncryption, the PKCS #1 "block type 01" encryption-block formatting scheme is employed. The block type BT is a single octet containing the value 0x01 and the padding string PS is one or more octets (enough octets to make the length of the complete RSA input value equal to the length of the modulus) each containing the value 0xFF. The data quantity D is comprised of the MIC and the MIC algorithm identifier which are ASN.1 encoded. H. Schulzrinne/S. Casner Expires 12/31/93 [Page 22]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 For the purposes of the MICA option, the encoding of data quantity D may be considered as a fixed binary sequence identifying the message integrity check algorithm, followed by the octets of the message digest. Currently, only the use of the MD2 and MD5 algorithms is defined, as described in RFC 1319 [9] (as corrected in Section 2.1 of RFC 1423) and RFC 1321 [10], respectively. For MD2, the fixed binary sequence (shown here in hexadecimal) is 0x30, 0x20, 0x30, 0x0C, 0x06, 0x08, 0x2A, 0x86, 0x48, 0x86, 0xF7, 0x0D, 0x02, 0x02, 0x05, 0x00, 0x04, 0x10, and for MD5 it is 0x30, 0x20, 0x30, 0x0C, 0x06, 0x08, 0x2A, 0x86, 0x48, 0x86, 0xF7, 0x0D, 0x02, 0x05, 0x05, 0x00, 0x04, 0x10. The appropriate sequence is followed immediately by the message digest, which is 16 octets long for both MD2 and MD5. For clarification of the octet ordering of the message digest, see RFC 1423, Sections 2.1 and 2.2. As an example, for an RSA encryption modulus length of 512 bits or 64 octets, the RSA input value would be: BT <---- PS ---> <------------ D -------------> 0x00 0x01 0xFF ... 0xFF 0x00 0x30 ... 0x10 [message digest] (29 octets) (18 octets) (16 octets) The length of the encrypted message digest will be equal to the modulus of the RSA encryption used, rounded to the next integral octet count. Contrary to what is specified in RFC 1423 for Privacy Enhanced Mail, the asymmetrically encrypted message digest is carried in binary, not represented in the printable encoding of RFC 1421, Section 4.3.2.4. The encrypted message digest is inserted into the MICA option immediately following the length octet, and is padded at the end to make the MICA option a multiple of four octets long. The value of the padding is left unspecified. The number of non-padding bits within the signature is known to the receiver as being equal to the key length. The modulus and public key are conveyed to the receivers by non-RTP means. After the message digest is decrypted, the message integrity check algorithm is identified through the octets prepended to the actual 16-octet digest. Asymmetric keys are used since symmetric keys would not allow authentication of the individual source in the multicast case. A translator is not allowed to modify the parts of an RTP packet covered by the MICA option as the receiver would have no way of establishing the identity of the translator and thus could not verify the integrity of the RTP packet. 5.5.4 MICK: Message integrity check, keyed 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 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ H. Schulzrinne/S. Casner Expires 12/31/93 [Page 23]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 |F| MICK = 11 | length | reserved | descriptor | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | message digest (including shared secret) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ This message integrity check option does not require encryption, but includes a shared secret in the computation of the message digest. The shared secret is equivalent to the key used for the MICS and ENC options, but is 16 octets long, padded if needed with binary zeroes. The shared secret is first placed into the MICK option where the message will later go, then the digest is computed over the fixed RTP header, the whole MICK option including the shared secret, and the remaining header options and payload that will immediately follow the MICK option in the RTP packet. The shared secret in the MICK option is then replaced by the computed 16-octet message digest for transmission. The receiver stores the message digest contained in the MICK option, replaces it with the shared secret key and computes the message digest in the same manner as the sender. If the RTP packet has not been tampered with and has originated with one of the holders of the shared secret, the computed message digest will agree with the digest found on reception in the MICK option.(5) The digest algorithm and shared secret are selected by the descriptor field. The value zero implies the use of the MD5 message digest and a single shared secret. 5.5.5 MICS: Message integrity check, symmetric-key encrypted 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 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |F| MICS = 12 | length | reserved | descriptor | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | message digest (symmetrically encrypted) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ This message integrity check encrypts the message digest using the DES algorithm in ECB mode as described in RFC 1423, Section 3.1. The digest ------------------------------ 5. This message integrity check follows the practice of SNMP Version 2, as described in RFC 1446, Section 1.5.1. Using the secret key in the computation of the message digest instead of encrypting the digest avoids the use of an encryption algorithm when only integrity and authentication are desired. However, the security of this approach has not been as well established as the authentication based on encrypting message digests used in the MICS, MIC and MICA options. H. Schulzrinne/S. Casner Expires 12/31/93 [Page 24]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 algorithm and symmetric key are selected by the descriptor field. The value zero implies the use of the MD5 message digest and a single key. 6 RTP Control Protocol --- RTCP The RTP control protocol (RTCP) conveys minimal control and advisory information during a session. It provides support for "loosely controlled" sessions, i.e., where participants enter and leave without membership control and parameter negotiation. The services provided by RTCP augment RTP, but an end system does not have to implement RTCP features to participate in sessions. There is one exception to this rule: if an application sends FMT options, the receiver has to decode these in order to properly interpret the RTP payload. RTCP does not aim to provide the services of a session control protocol and does not provide some of the services desirable for two-party conversations. If a session control protocol is in use, the services of RTCP should not be required. (As of the writing of this document, a session or conference control protocol has not been specified within the Internet.) RTCP options share the same structure and numbering space as RTP options, which are described in Section 5. Unless otherwise noted, control information is carried periodically as options within RTP packets, with or without payload. RTCP packets are sent to all members of a session, typically using multicast. These packets are part of the same sequence number space as RTP packets not containing RTCP options. The period should be varied randomly to avoid synchronization of all sources and its mean should increase with the number of participants in the session to limit the growth of the overall network and host interrupt load. The length of the period determines, for example, how long a receiver joining a session has to wait until it can identify the source. A receiver may remove from its list of active sites a site that it has not been heard from for a given time-out period; the time-out period may depend on the number of sites or the observed average interarrival time of RTCP messages. Note that not every periodic message has to contain all RTCP options; for example, the EMAIL part within the SDES option might only be sent every few messages. RTCP options should also be sent when information carried in RTCP options changes, but the generation of RTCP options should be rate-limited. 6.1 FMT: Format description 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 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |F| FMT = 32 | length |R|R| format | reserved | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ H. Schulzrinne/S. Casner Expires 12/31/93 [Page 25]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 | format name | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | format-dependent data ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ format: 6 bits The format field corresponds to the index value from the format field in the RTP fixed header, with values ranging from 0 to 63. format name: 4 octets The format name describes the format in an unambiguous way and is registered with the Internet Assigned Numbers Authority. The format name is interpreted as a sequence of four ASCII characters, with uppercase and lowercase characters treated as distinct. Format names beginning with the letter 'X' are reserved for experimental use and not subject to registration. format-dependent data: variable length Format-dependent data may or may not appear in a FMT option. It is interpreted by the application and not RTP itself. A FMT mapping changes the interpretation of a given format value carried in the fixed RTP header starting at the packet containing the FMT option. The new interpretation applies only to packets from the same synchronization source as the packet containing the FMT option. If format mappings are changed through the FMT option, the option should be sent periodically as otherwise sites that did not receive the FMT option due to packet loss or joining the session after the FMT option was sent will not know how to interpret the particular format value. 6.2 SDES: Source descriptor 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 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |F| SDES = 34 | length | source identifier | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | type=PORT (2) | length = 1 | port | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | type=ADDR (1) | length | reserved | address type | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | network-layer address ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ H. Schulzrinne/S. Casner Expires 12/31/93 [Page 26]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | type=PORT (2) | length > 1 | reserved | reserved | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | port ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | type=CNAME (4)| length | user and domain name ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | type=EMAIL (5)| length | electronic mail address ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | type=NAME (6) | length | common name of source ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | type=LOC (8) | length | geographic location of site ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | type=TXT (16) | length | text describing source ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | type=PRIV(255)| length > 1 | subtype | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | name (ASCII) | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | application-defined content ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ The SDES option is composed of the first four octets shown concatenated with one or more of the subsequent items as described individually below. SDES provides a mapping between a numeric source identifier and those items, which describe a particular source.(6) For those applications where the size of a multi-item SDES option would be a concern, multiple SDES options may be formed with subsets of the items to be sent in separate packets. When an SDES option originates from a content source (the actual source of the data), the identifier value is zero. If the data flows through a bridge, the bridge forwards the SDES information, but changes the SDES source identifier to the value used in the CSRC option when identifying ------------------------------ 6. Several attributes were combined into one option so that the receiver does not have to perform multiple mappings from identifiers to site data structures. H. Schulzrinne/S. Casner Expires 12/31/93 [Page 27]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 that content source. The bridge may choose to store the SDES information received from a content source and change then number of items sent together or the rate at which SDES information is sent. A bridge uses an identifier value of zero within an SDES option to describe itself rather than the content sources bridged by it, but if a bridge contributes local data to outgoing packets, it should select another non-zero source identifier for that traffic and send CSRC and SDES options for it as well. Translators do not modify or insert SDES options. The end system performs the same mapping it uses to identify the content sources (that is, the combination of transport source, SSRC identifier and the source identifier within this SDES option) to identify a particular source. SDES information is specific to traffic from a source on a particular channel, unless a profile or a higher-layer control protocol defines that the same SDES describes traffic from that source on some set of channels. Each item has a structure similar to that of RTP and RTCP options, that is, a type field followed by a length field, measured in multiples of four octets. No final bit (see Section 5.2) is needed since the overall length is known. Item types 0 through 127 apply to all profiles, while types 128 through 254 are allocated to profile-specific items; both ranges are reserved and registered with the Internet Assigned Numbers Authority (IANA). Item type 255 (PRIV) is provided for private or experimental extensions not registered with IANA. Items are independent of the format value. All of the SDES items are optional and unrecognized items may be ignored; however, if quality-of-service monitoring is to be used, receivers will require the PORT and ADDR items from the SDES option in order to construct the QOS option. Only the TXT item is expected to change during the duration of a session. Text items are encoded according to the rules in Section 4. Items are padded with the binary value zero to the next multiple of four octets. Each item may appear only once unless otherwise noted. A description of the content of these items follows: PORT/ADDR: The PORT item contains the source transport selector, such as the UDP source port number, and the ADDR item contains the network address of the source, for example, the IP version 4 address or an NSAP. Both are carried in binary form, not as "dotted decimal" or similar human-readable form. Address types are identified by the Domain Name Service Resource Record (RR) type, as specified in the current edition of the Assigned Numbers RFC. There must be no more than one PORT item in an SDES option. The PORT item should be followed immediately by an ADDR item. Concatenated, these two items serve as a globally unique identifier for the source which is returned in the QOS option. As far as RTP is concerned, this identifier is opaque, so it is unimportant which address is used for multi-homed hosts. Applications may find the address or port useful for debugging or monitoring, but should not assume that the combination can be used to communicate with the source process because it may be on H. Schulzrinne/S. Casner Expires 12/31/93 [Page 28]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 the other side of a firewall or using a different transport protocol. If it is useful to the application, it is permissible for a source to include additional ADDR items after the first to convey additional addresses if the source is multi-homed, or if the source's address may be represented in multiple schemes, for example during the transition from IPv4 to IPng.(7) The figure shows the PORT item in two forms. The first form shows the concatenated PORT and ADDR items as they would be used for the TCP and UDP protocols. For an IPv4 address, the length of the ADDR item would be 2, and the address type would be 1. The second form of the PORT item is indicated by a length field greater than one and is used when the transport selector (port number) is larger than two octets. Octets three and four of the item are reserved (zero) and the transport selector appears in words two and following of this item, in network byte order. CNAME: Canonical user and host identifier, e.g., "doe@sleepy.megacorp.com" or "sleepy.megacorp.com". The CNAME item must have the format "user@host" or "host", where "host" is the fully qualified domain name of the host from which the real-time data originates, formatted according to the rules specified in RFC 1034, RFC 1035 and Section 2.1 of RFC 1123. The "host" form may be used if a user name is not available, for example on single-user systems. 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 real-life name. Note that the host name is not necessarily identical to the electronic mail address of the participant. EMAIL: User's electronic mail address, formatted according to RFC 822, for example, "John.Doe@megacorp.com". NAME: Common name describing the source, e.g., "John Doe, Bit Recycler, Megacorp". This name 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 (profiles may establish such priorities). ------------------------------ 7. The ordering simplifies processing at the receiver, as the consecutive octet string of PORT followed by the first ADDR can be stored as the globally unique identifier. H. Schulzrinne/S. Casner Expires 12/31/93 [Page 29]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 LOC: Geographic user location. 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. TXT: Text describing the source, e.g., "out for lunch". PRIV: Private, unregistered items. The subtype and name fields are to be used in the same manner as in the APP option (Section 5.3). The format and content of the octets following the name field are determined by the application defining the item. 6.3 BYE: Goodbye 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 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |F| BYE = 35 | length = 1 | content source identifier | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ The BYE option indicates that a particular session participant is no longer active. When a BYE option originates from a content source (the actual source of the data), the identifier value is zero. If the message flows through a bridge, the bridge forwards the BYE message, but changes the identifier to be the (non-zero) value used in the CSRC option when identifying that content source. If a bridge shuts down, it should first send BYE options for all content sources it handles, followed by a BYE option with an identifier value of zero. An RTCP message may contain more than one BYE option. Multiple identifiers in a single BYE option are not allowed, to avoid ambiguities between the special value of zero and any necessary padding. 6.4 QOS: Quality of service measurement 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 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |F| QOS = 36 | length | reserved | reserved | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | packets expected | H. Schulzrinne/S. Casner Expires 12/31/93 [Page 30]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | packets received | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | minimum delay (seconds) | minimum delay (fraction) | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | maximum delay (seconds) | maximum delay (fraction) | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | average delay (seconds) | average delay (fraction) | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | type=PORT (2) | length | transport address ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | type=ADDR (1) | length | reserved | address type | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | network-layer address ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ The QOS option conveys statistics on the reception of packets from a single synchronization source on a single channel. These statistics are the number of packets received, the number of packets expected, the minimum delay, the maximum delay and the average delay. The expected number of packets may be computed according to the algorithm in Section A.5. The delay measures are in units of 1/65536 of a second, that is, with the same resolution as the timestamp in the fixed RTP header. The synchronization source to which these statistics correspond is identified by appending to the fixed-length part of the QOS option the PORT and ADDR items, in that order, as received in an SDES option from that source. Together, the PORT and ADDR items form a globally unique identifier (as described with the SDES option, Section 6.2). If the source has supplied more than one ADDR item, only the first one from the SDES (the one immediately following the PORT item) is used. If no SDES option, or none with PORT and ADDR items, has been received from a particular source, the QOS option cannot be sent unless the PORT and ADDR items are known by some other mechanism. The QOS option may be sent in either normal (forward) or reverse RTP packets. In the first case, the channel to which these statistics correspond is same as the channel on which the QOS option is sent; that is, the channel is identified by the destination (multicast or unicast) address, destination port and channel ID. If the QOS option is sent in a reverse RTP packet, the channel is identified by the channel ID in the header and the destination port number as the packet arrives at the synchronization source, which will be the same port that the source uses to send data on that channel, as described in Section 5.4. Sending the QOS option by multicast has the advantage that all participants in the session can compare their reception to that of others, and allows participants to adjust the rate at which QOS is sent based on the number of participants. A single RTCP packet may contain several QOS options for different sources. It is left to the implementor to decide how often to transmit QOS options H. Schulzrinne/S. Casner Expires 12/31/93 [Page 31]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 and which sources are to be included. 7 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. For example, the SDES fields 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 [11]. Therefore, the use of security mechanisms with RTP is important. As a first step, RTP options make it easy for all participants in a session to identify themselves; if deemed important for a particular application, it is the responsibility of the application writer to make listening without identification difficult. It should be noted, however, that privacy of the payload can generally be assured only by encryption. The security options described in Section 5.5 can be used to implement message integrity, authentication and confidentiality and the combination of the three. These security services might also be provided at the IP layer as security mechanisms are developed for that layer. The periodic transmission of SDES options from sources that are otherwise idle may make it possible to detect denial-of-service attacks, as the receiver can detect the absence of these expected messages. The messages that are received must be verified for integrity and authenticated before being accepted for this purpose. Unlike for other data, ciphertext-only attacks may be more difficult for compressed audio and video sources. Such data is very close to white noise, making statistics-based ciphertext-only attacks difficult. Even without message integrity check options, it may be difficult for an attacker to detect automatically when he or she has found the secret cryptographic key since the bit pattern after correct decryption may not look significantly different from one decrypted with the wrong key. However, the session information is more or less constant and predictable, allowing known-plaintext attacks. Chosen-plaintext attacks appear, in general, to be difficult. The integrity of the timestamp in the fixed RTP header can be protected by the message integrity options. If clocks are known to be synchronized, an attacker only has a very limited time window of maybe a few seconds every 18 hours to replay recorded RTP without detection by the receiver. H. Schulzrinne/S. Casner Expires 12/31/93 [Page 32]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 Key distribution and certificates are outside the scope of this document. 8 RTP over Network and Transport Protocols This section describes issues specific to carrying RTP packets within particular network and transport protocols. 8.1 Defaults The following rules apply unless superseded by protocol-specific subsections in this section. The rules apply to both forward and reverse RTP packets. RTP packets contain no length field or other delineation, therefore a framing mechanism is needed if they are carried in underlying protocols that provide the abstraction of a continuous bit stream rather than messages (packets). TCP is an example of such a protocol. Framing is also needed if the underlying protocol may contain padding so that the extent of the RTP payload cannot be determined. For these cases, each RTP packet is prefixed by a 32-bit framing field containing the length of the RTP packet measured in octets, not including the framing field itself. If an RTP packet traverses a path over a mixture of octet-stream and message-oriented protocols, each RTP-level bridge between these protocols is responsible for adding and removing the framing field. A profile may specify that this framing method is 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. 8.2 ST-II When used in conjunction with RTP, ST-II [12] service access ports (SAPs) have a length of 16 bits. The next protocol field (NextPCol, Section 4.2.2.10 in RFC 1190) is used to distinguish two encapsulations of RTP over ST-II. The first uses NextPCol value TBD and directly places the RTP packet into the ST-II data area. If NextPCol value TBD is used, the RTP header is preceded by a 32-bit header shown below. The octet count determines the number of octets in the RTP header and payload to be checksummed, allowing the checksum to cover only the header if it is preferred to ignore errors in H. Schulzrinne/S. Casner Expires 12/31/93 [Page 33]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 the data. The 16-bit checksum uses the TCP and UDP checksum algorithm. 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 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | count of octets to be checked | checksum | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | RTP packet (fixed header, options and payload) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 9 RTP Profiles 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 defining a profile to select the appropriate choices for a particular class of applications and environment. A profile for audio and video applications may be found in the companion Internet draft draft-ietf-avt-profile. Within this specification, the following possible uses of a profile have been identified, but this list is not meant to be exclusive: o Define a set of formats (e.g., media encodings) and a default mapping of those formats to format values. o Define new, application-class-specific options, or specify that an option, such as BOS, should be included in every packet. o Specify the support for and semantics of particular options to be used in Reverse Path messages. o Define new application-class-specific SDES items, or the data format, preferred use, or required use of particular SDES items. o Define when SDES applies to some grouping of channels rather than just a single channel. o Specify that globally synchronized time is required for operation of an application. o Specify that a particular underlying network or transport layer protocol will be used to carry RTP packets. o Specify that the RTP header is always to be prefixed by the framing field to allow carrying multiple RTP packets (perhaps for different media) in one lower-layer packet. H. Schulzrinne/S. Casner Expires 12/31/93 [Page 34]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 o Describe the application of the quality-of-service option. 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. For example, additional options or formats can be defined and registered through IANA for publication in the Assigned Numbers RFC as an alternative to publishing a new profile specification. It is recommended that a profile specify a default port number to be used with that profile so that applications that support operation under multiple profiles can use the port number to select the profile. A Implementation Notes We describe aspects of the receiver implementation in this section. 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. The following definitions are used for all examples; the structure definitions are valid for 32-bit big-endian architectures only. Bit fields are assumed to be packed tightly, with no additional padding. #include <sys/types.h> typedef double CLOCK_t; /* the definitions below are valid for 32-bit architectures and will have to be changed for 16- or 64-bit architectures */ typedef u_long u_int32; typedef u_short u_int16; typedef enum { RTP_CSRC = 0, RTP_SSRC = 1, RTP_SDST = 2, RTP_BOS = 3, RTP_ENC = 8, RTP_MIC = 9, RTP_MICA = 10, RTP_MICK = 11, RTP_MICS = 12, RTP_FMT = 32, RTP_SDES = 34, RTP_BYE = 35, H. Schulzrinne/S. Casner Expires 12/31/93 [Page 35]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 RTP_QOS = 36, RTP_MINFMT = 96, RTP_MAXFMT = 126, RTP_APP = 127 } rtp_option_t; typedef struct { unsigned int ver:2; /* version number */ unsigned int channel:6; /* channel id */ unsigned int p:1; /* option present */ unsigned int s:1; /* sync bit */ unsigned int format:6; /* format of payload */ u_int16 seq; /* sequence number */ u_int32 ts; /* timestamp */ } rtp_hdr_t; typedef union { struct { /* generic first 16 bits of options */ unsigned int final:1; /* final option */ unsigned int type:7; /* option type */ } generic; struct { unsigned int final:1; /* final option */ unsigned int type:7; /* option type */ u_char length; /* length, including type/length */ u_int16 id[1]; /* content source identifier */ } csrc; /* ... */ } rtp_t; A.1 Timestamp Recovery For some applications it is useful to have the receiver reconstruct the sender's high-order bits of the NTP timestamp from the received 32-bit RTP timestamp. The following code uses double-precision floating point numbers for whole numbers with a 48-bit range. Other type definitions of CLOCK_t may be appropriate for different operating environments, e.g., 64-bit architectures or systems with slow floating point support. The routine applies to any clock frequency, not just the RTP value of 65,536 Hz, and any clock starting point. It will reconstruct the correct high-order bits as long as the local clock now is within one half of the wrap-around time of the 32-bit timestamp, e.g., approximately 9.1 hours for RTP timestamps. #include <math.h> #define MOD32bit 4294967296. #define MAX31bit 0x7fffffff H. Schulzrinne/S. Casner Expires 12/31/93 [Page 36]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 CLOCK_t clock_extend(ts, now) u_int32 ts; /* in: timestamp, low-order 32 bits */ CLOCK_t now; /* in: current local time */ { u_int32 high, low; /* high and low order bits of 48-bit clock */ low = fmod(now, MOD32bit); high = now / MOD32bit; if (low > ts) { if (low - ts > MAX31bit) high++; } else { if (ts - low > MAX31bit) high--; } return high * MOD32bit + ts; } /* extend_timestamp */ Using the full timestamp internally has the advantage that the remainder of the receiver code does not have to be concerned with modulo arithmetic. The current local time does not have to be derived directly from the system clock for every packet. For audio samples, for example, it is more accurate to maintain the time within a synchronization unit in samples, incrementing by the number of samples per packet, and then converting to an RTP timestamp. The following code implements the conversion from a 8 KHz sample clock into an RTP timestamp. This assumes that the sample clock is also started at the RTP clock epoch (January 1, 1970). If not, the appropriate offset has to be added. CLOCK_t t; /* 8-kHz sample clock */ CLOCK_t RTP_ts; /* RTP timestamp */ RTP_ts = floor(t * 8.192 + 0.5); The whole seconds within NTP timestamps can be obtained by adding 2208988800 to the value of the standard Unix clock (generated, for example, by the gettimeofday system call), which starts from the year 1970. For the RTP timestamp, only the least significant 16 bits of the seconds are used. A.2 Detecting the Beginning of a Synchronization Unit RTP packets contain a bit flag indicating the end of a synchronization unit. The following code fragment determines, based on sequence numbers, if a H. Schulzrinne/S. Casner Expires 12/31/93 [Page 37]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 packet is the beginning of a synchronization unit. It assumes that the packet header has been converted to host byte order. static u_int32 seq_eos; rtp_hdr_t *h; static int flag; if (h->s) { flag = 1; seq_eos = h->seq; } /* handle wrap-around of sequence number */ else if (flag && (h->seq - seq_eos < 32768)) { flag = 0; /* code here to handle beginning of synchronization unit */ } A.3 Demultiplexing and Locating the Synchronization Source The combination of destination address, destination port and channel ID determines the channel. For each channel, the receiver maintains a list of all sources, content and synchronization sources alike, in a table or other suitable data structure. Synchronization sources are stored with a content source identifier of zero. When an RTP packet arrives, the receiver determines its network source address and port (from information returned by the operating system), synchronization source identifier (SSRC option) and content source identifier(s) (CSRC option). To locate the table entry containing timing information, mapping from content descriptor to actual encoding, etc., the receiver sets the content source identifier to zero and locates a table entry based on the tuple (transport source address, synchronization source identifier, 0). The receiver identifies the contributors to the packet (for example, the speaker who is heard in the packet) through the list of content source identifiers carried in the CSRC option. To locate the table entry, it matches on the triple (network address and port, synchronization source identifier, content source identifier). A.4 Parsing RTP Options The following code segment walks through the RTP options, preventing infinite loops due to zero and invalid length fields. Structure definitions are valid for big-endian architectures only. H. Schulzrinne/S. Casner Expires 12/31/93 [Page 38]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 u_int32 len; /* length of RTP packet in bytes */ u_int32 *pt; /* pointer */ rtp_hdr_t *h; /* fixed header */ rtp_t *r; /* options */ if (h->p) { pt = (u_int32 *)(h+1); do { r = (rtp_t *)pt; pt += r->generic.length; /* point to end of option */ /* invalid length field */ if ((char *)pt - (char *)h > len || r->generic.length == 0) return -1; switch(r->generic.type) { case RTP_BYE: /* handle BYE option */ break; case RTP_CSRC: /* handle CSRC option */ break; /* ... */ default: if (r->generic.type >= RTP_MINFMT && r- >generic.type <= RTP_MAXFMT) { /* call option handler particular to this format */ (parse_format_options[h->format])(r); } else break; /* ignore undefined options */ } } while (!r->generic.final); } A.5 Determining the Expected Number of RTP Packets The number of packets expected can be computed by the receiver by tracking the first sequence number received (seq0), the last sequence number received, seq, and the number of complete sequence number cycles: expected = cycles * 65536 + seq - seq0 + 1; The cycle count is updated for each packet, where seq_prior is the sequence number of the prior packet: H. Schulzrinne/S. Casner Expires 12/31/93 [Page 39]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 unsigned long seq, seq_prior; if (seq - seq_prior > 65536) cycle++; else if (seq - seq_prior > 32768) cycle--; seq_prior = seq; Acknowledgments This memorandum is based on discussions within the IETF Audio/Video Transport working group chaired by Stephen Casner. The current protocol has its origins in the Network Voice Protocol and the Packet Video Protocol (Danny Cohen and Randy Cole) and the protocol implemented by the vat application (Van Jacobson and Steve McCanne). Stuart Stubblebine (ISI) helped with the security aspects of RTP. Ron Frederick (Xerox PARC) provided extensive editorial assistance. B Addresses of Authors Stephen Casner USC/Information Sciences Institute 4676 Admiralty Way Marina del Rey, CA 90292-6695 telephone: +1 310 822 1511 (extension 153) electronic mail: casner@isi.edu Henning Schulzrinne AT&T Bell Laboratories MH 2A244 600 Mountain Avenue Murray Hill, NJ 07974-0636 telephone: +1 908 582 2262 facsimile: +1 908 582 5809 electronic mail: hgs@research.att.com References [1] D. E. Comer, Internetworking with TCP/IP, vol. 1. Englewood Cliffs, New Jersey: Prentice Hall, 1991. H. Schulzrinne/S. Casner Expires 12/31/93 [Page 40]
INTERNET-DRAFT draft-ietf-avt-rtp-04.txt October 20, 1993 [2] J. Postel, "Internet protocol," Network Working Group Request for Comments RFC 791, Information Sciences Institute, Sept. 1981. [3] 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. [4] The Unicode Consortium, The Unicode Standard. New York, New York: Addison-Wesley, 1991. [5] D. L. Mills, "Network time protocol (version 3) -- specification, implementation and analysis," Network Working Group Request for Comments RFC 1305, University of Delaware, Mar. 1992. [6] S. Kent, "Understanding the Internet certification system," in Proceedings of the International Networking Conference (INET), (San Francisco, California), pp. BAB--1 -- BAB--10, Internet Society, Aug. 1993. [7] D. Balenson, "Privacy enhancement for internet electronic mail: Part III: Algorithms, modes, and identifiers," Network Working Group Request for Comments RFC 1423, IETF, Feb. 1993. [8] V. L. Voydock and S. T. Kent, "Security mechanisms in high-level network protocols," ACM Computing Surveys, vol. 15, pp. 135--171, June 1983. [9] J. Kaliski, Burton S., "The MD2 message-digest algorithm," Network Working Group Request for Comments RFC 1319, RSA Laboratories, Apr. 1992. [10] R. Rivest, "The MD5 message-digest algorithm," Network Working Group Request for Comments RFC 1321, IETF, Apr. 1992. [11] S. Stubblebine, "Security services for multimedia conferencing," in 16th National Computer Security Conference, (Baltimore, MD), pp. 391--395, Sept. 1993. [12] C. Topolcic, S. Casner, C. Lynn, Jr., P. Park, and K. Schroder, "Experimental internet stream protocol, version 2 (ST-II)," Network Working Group Request for Comments RFC 1190, BBN Systems and Technologies, Oct. 1990. H. Schulzrinne/S. Casner Expires 12/31/93 [Page 41]
