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Internet Engineering Task Force Audio-Video Transport WG INTERNET-DRAFT H. Schulzrinne/S. Casner AT&T/ISI July 30, 1993 Expires: 10/01/93 RTP: A Real-Time Transport Protocol 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 draft describes a real-time transport protocol (RTP) suitable for the network transport of real-time data, such as audio, video or simulation data for both multicast and unicast transport services. The data transport is enhanced by a control protocol (RTCP) designed to provide minimal control and identification functionality particularly in multicast networks. 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. Within multicast associations, sites can direct control messages to individual sites. This specification is a product of the Audio-Video Transport working group within the Internet Engineering Task Force. Comments are solicited and should be addressed to the working group's mailing list at rem-conf@es.net and/or the authors. INTERNET-DRAFT RTP July 30, 1993 Contents 1 Introduction 2 2 Protocol Conventions 3 3 Real-time Data Transfer Protocol -- RTP 4 3.1 Definitions . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4 3.2 RTP Fixed Header Fields . . . . . . . . . . . . . . . . . . . . . . 6 3.3 The RTP Options . . . . . . . . . . . . . . . . . . . . . . . . . . 8 3.4 Reverse-Path Option . . . . . . . . . . . . . . . . . . . . . . . . 9 3.5 Security Options . . . . . . . . . . . . . . . . . . . . . . . . . 11 3.6 The Use of the Security Options . . . . . . . . . . . . . . . . . . 15 4 Real Time Control Protocol --- RTCP 17 5 Security Considerations 22 6 RTP over network and transport protocols 23 6.1 Defaults . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23 6.2 ST-II . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24 A Implementation Notes 24 A.1 Timestamp recovery . . . . . . . . . . . . . . . . . . . . . . . . 24 A.2 Detecting the Beginning of a Synchronization Unit . . . . . . . . . 25 A.3 Demultiplexing and Locating the Synchronization Source . . . . . . 26 B Addresses of Authors 27 1 Introduction This draft concisely specifies a real-time transport protocol. A discussion of the design decisions can be found in the current version of the companion H. Schulzrinne/S. Casner Expires 10/01/93 [Page 2] INTERNET-DRAFT RTP July 30, 1993 Internet draft draft-ietf-avt-issues.txt. The transport protocol provides end-to-end delivery services for one or more s_t_r_e_a_m_s_ of data with real-time characteristics, for example, interactive audio and video. It does n_o_t_ 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 may 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. [For most applications, RTP offers insufficient demultiplexing to run directly on IP.] RTP transfers data in a single direction, possibly to multiple destinations if supported by the underlying network. A mechanism for indicating a return path for control 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. A profile for audio and video data may be found in the companion Internet draft draft-ietf-avt-profile.txt. This document defines two packet formats and protocols: o the real-time transport protocol (RTP) for exchanging data with real-time properties. o the real-time control protocol (RTCP) for conveying information about the sites in an on-going association. RTCP options may be ignored without affecting the ability to correctly receive data. RTCP is used for loosely controlled conferences, i.e., where there is no explicit admission control and set-up. Its functionality may be subsumed by a conference control protocol (which is beyond the scope of this document). 2 Protocol Conventions Control fields (options) for RTP and RTCP share the same structure and numbering space and are carried within the same packet. Options may appear in any order, unless specifically restricted by the option description. [The position of some security options may have significance.] Each option consists of the final bit, the option type designation, a one-octet length field denoting the total number of 32-bit long words comprising the option (including final bit, type and length), and finally any option-specific data. The last option before the packet data portion (``payload'') has the 'F' (final) bit set to one, for all other options this field has a value of H. Schulzrinne/S. Casner Expires 10/01/93 [Page 3] INTERNET-DRAFT RTP July 30, 1993 zero. 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. Options unknown to the RTP implementation or the application are to be ignored. Options with option types having values from 64 to 127 inclusive are to be used for private extensions. Fields designated as 'reserved' or 'R' are set aside for future use; they should be set to zero by senders and ignored by receivers. All integer fields are carried in network byte order, that is, most significant byte (octet) first. The transmission order is described in detail in [1], Appendix A. Unless otherwise noted, numeric constants are in decimal (base 10). Numeric constants prefixed by '0x' are in hexadecimal. Textual information is encoded accorded to the UTF-2 encoding of the ISO standard 10646 (Annex F) [2,3]. US-ASCII is a subset of this encoding and requires no additional encoding. The presence of multi-byte encodings is indicated by setting the most significant bit to a value of one. A byte with a binary value of zero may be used as a string terminator for padding purposes. [Text in square brackets is intended to motivate the design decisions made.] 3 Real-time Data Transfer Protocol -- RTP 3.1 Definitions P_a_y_l_o_a_d_ is the data following the RTP fixed header and the RTP/RTCP options. The payload format and interpretation are beyond the scope of this memo. A valid RTP packet may carry no payload. An R_P_D_U_ stands for RTP protocol data unit. It 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 s_y_n_c_h_r_o_n_i_z_a_t_i_o_n_ s_o_u_r_c_e_ is the combination of one or more content sources with its own timing. The RPDUs emitted by a synchronization source have non-decreasing sequence numbers and time stamps (modulo their field lengths). The audio coming from a microphone or the video from a source are examples of synchronization sources. Typically, a single source emits a single medium (e.g., audio or video). A synchronization source is a member of exactly one channel, as defined below. A synchronization source may change its data format over time. Synchronization sources are identified by their source network address, source transport address (e.g., UDP source port) and the value of SSRC identifier carried in the SSRC option. If the H. Schulzrinne/S. Casner Expires 10/01/93 [Page 4] INTERNET-DRAFT RTP July 30, 1993 SSRC option is not present, a value of zero for that identifier is assumed. A c_o_n_t_e_n_t_ s_o_u_r_c_e_ is the actual source of the data carried, for example, the user and host that originally generated the audio data. One or more content sources may contribute data for one synchronization source. Content sources are used for identifying the logical source of the data; they have no effect on the delivery of the data itself. A n_e_t_w_o_r_k_ s_o_u_r_c_e_ is the network-level origin of the RPDUs as seen by the receiving end system. All sources sending to the same destination network address and transport-level address using the same RTP flow identifier belong to same c_h_a_n_n_e_l_. An e_n_d_ s_y_s_t_e_m_ generates the content to be used in RTP packets and delivers the content of received RTP packets to the user application. An end system can act as one or more synchronization sources. (Most end systems are expected to be a single synchronization source.) An (RTP-level) b_r_i_d_g_e_ 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 source will not generally be synchronized, the bridge will make timing adjustments among the streams and generate its own timing for the combined stream. Therefore, bridges are synchronization sources, with each of the sources whose packets were combined into an outgoing RTP packet as the content sources for that outgoing packet. Audio bridges and media converters are examples of bridges. Example: assume SMITH@FOO and JONES@BAR are using a bridge to translate their audio from one encoding to another. The bridge mixes audio packets from Smith and Jones together and forwards the mixed packets. If, say, Smith was talking, she is indicated as the content source of the outgoing packet, allowing the receiver to properly display the current speaker rather than just the bridge that mixed the audio. For an end system receiving RTP packets from that bridge, the bridge is the synchronization source and Smith the content source. 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 usage.] An (RTP-level) t_r_a_n_s_l_a_t_o_r_ does not alter the timing of packets. 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. The properties of bridges and translators are summarized in Table 1. Checkmarks in parentheses designate possible, but unlikely actions. A s_y_n_c_h_r_o_n_i_z_a_t_i_o_n_ u_n_i_t_ consists of one or more packets that, as a group, share a common fixed delay between generation and playout of each part of the group, or can only be scheduled as a whole. The delay may change at the H. Schulzrinne/S. Casner Expires 10/01/93 [Page 5] INTERNET-DRAFT RTP July 30, 1993 end sys. bridge translator mix sources -- x -- change encoding N/A x x encrypt x x (x) sign for authentication x x -- touch content x x (x) insert CSRC -- x -- insert SSRC x x x insert SDST x x -- insert SDES x x -- Table 1: The properties of end systems, bridges and translators beginning of such a synchronization unit. The most common synchronization units are talkspurts for voice and frames for video transmission. N_o_n_-_R_T_P_ m_e_c_h_a_n_i_s_m_s_ 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 encryption and authentication keys, negotiate the encryption algorithm to be used, 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 the mechanism itself is outside the scope of this memorandum. 3.2 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| FlowID |P|S| format | sequence number | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | timestamp (seconds) | timestamp (fraction) | +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ | options ... | +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ The fields in the first eight octets are present in every RTP packet and have the following meaning: protocol version: 2 bits H. Schulzrinne/S. Casner Expires 10/01/93 [Page 6] INTERNET-DRAFT RTP July 30, 1993 Defines the protocol version. The version number of the protocol defined in this memo is one. FlowID: 6 bits The value of the field is the flow identifier, which forms part of the tuple identifying a channel (see definition above). [The flow 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 RPDUs on different channels into a single protocol data unit of the underlying protocol layer.] option present bit (P): 1 bit This flag has a value of one if the fixed RTP header is followed by one or more options and a value of zero otherwise. end-of-synchronization-unit (S): 1 bit This flag has a value of one in the last packet of a synchronization unit, a value of zero otherwise. [As shown in Section A, the beginning of a synchronization unit can be readily established from this flag. If this flag were to signal to the beginning of a synchronization unit, the end of a synchronization unit could not be established in real time.] format: 6 bits The 'format' field forms an index into a table defined through the RTCP FMT option or non-RTP mechanisms (see Section 3.1. The mapping establishes the format of the RTP payload and determines its interpretation by higher layers. If no mapping has been defined in this manner, a standard mapping is specified by the companion profile document, RFC TBD. Also, default formats may be defined by the current edition of the Assigned Numbers RFC. sequence number: 16 bits The sequence number counts RPDUs. 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 wallclock time when the RPDU was generated. The timestamp consists of the middle 32 bits of a 64-bit NTP timestamp, as defined in RFC 1305 [4]. Several consecutive packets may have equal timestamps. The timestamp of the first packet(s) within a synchronization unit is expected to closely reflect the actual sampling instant, measured by the local system clock. The local system clock should be controlled by a time synchronization protocol such as NTP if such a service is available. It is not expected that the local system clock be referenced to obtain the timestamp for the beginning of every synchronization unit, but the local clock should be referenced H. Schulzrinne/S. Casner Expires 10/01/93 [Page 7] INTERNET-DRAFT RTP July 30, 1993 frequently enough so that clock drift between synchronized system clock and 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, the nominal sampling interval may be used. If the clock quality field of the CDES option does not indicate otherwise, it is assumed that the timestamp at the beginning of a synchronization unit is derived from a synchronized system clock. 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. 3.3 The RTP Options The packet header may be followed by options and the payload. Options are summarized below. Unless otherwise noted, each option may appear only once per packet. Each packet may contain any number of options. A conforming implementation of RTP has to support the RTP options listed here, unless otherwise noted. CSRC 0 Content source identifiers. The content source option is inserted only by bridges and identifies all sources that contributed to the packet. For example, for audio packets, all sources that were mixed together to create this packet are listed, allowing correct talker indication at the receiver. Each CSRC option may contain one or more content source identifiers, each 16 bits long. The identifier values must be unique for all content sources received through a particular synchronization source (bridge) on a particular channel; the value of binary zero is reserved and may not be 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 only be a single 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 RTP-level translators. A conformant RTP implementation does not have to be able to generate or interpret the CSRC 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| CSRC | length | content source identifier ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ SSRC 1 Synchronization source identifier. The SSRC option may be inserted by RTP-level translators, end systems and bridges. It is typically used only by translators, but it may be used by an H. Schulzrinne/S. Casner Expires 10/01/93 [Page 8] INTERNET-DRAFT RTP July 30, 1993 end system application to distinguish several sources sent with the same lower-layer source address. Each synchronization source with the same lower-layer address (e.g., the same IP address and UDP port) must have a distinct SSRC. Synchronization sources that are distinguishable by their lower-layer address do not require the use of SSRC options. The SSRC value zero is reserved and must not be used. If no SSRC option is present, the network source is assumed to indicate the synchronization source. There must be no more than one SSRC identifier per packet; thus, a translator must remap the SSRC identifier of an incoming packet into a new, locally unique SSRC identifier. The SSRC option may be considered in functionality as an extension of the source port number in protocols like UDP, ST-II or TCP. A 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. 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 | length = 1 | identifier | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ BOP 3 (beginning of playout unit) 16-bit sequence number designating the first packet within the current playout 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| BOP | length = 1 | sequence number | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 3.4 Reverse-Path Option With two-party (unicast) communications, relaying back control information to the sender is easy. For multicast communications, control information can be sent to all members of the group. It may, however, be desirable to send a message to an individual member of a multicast group, for example to request retransmission of a particular data frame or to request/send a reception quality report. For this particular use, we introduce a mechanism for sending so-called reverse RPDUs. The RPDU format of reverse RPDUs is exactly the same as for regular messages and they can make use of all the options defined in this memorandum. Reverse RPDUs travel through the same translators as other RPDUs. The receiver distinguishes reverse RPDUs by their arrival on a different transport selector (e.g., a different UDP port), namely the same one which is used as a source transport selector H. Schulzrinne/S. Casner Expires 10/01/93 [Page 9] INTERNET-DRAFT RTP July 30, 1993 (e.g., UDP source port) for forward RPDUs. A receiver of reverse RDPUs cannot rely on any sequence number ordering, as a sender may use the same sequence number space while communicating through this reverse mechanism with several receivers. The sequence number space of reverse RPDUs has to be completely separate from that used for RPDUs sent to the multicast group. If the same sequence number space were used, the members of the multicast group not receiving reverse RPDUs would detect a gap in their received sequence number space. SDST 2 Synchronization destination identifier. The SDST option is only inserted by RTP end systems and bridges if they want to send unicast information to a particular site within the multicast group. Packets containing an SDST option must not contain an SSRC option and vice versa. The identifier value zero is allowed, unlike for SSRC options (see example below). Denote the the end system that wants to return a unicast message by S and the desired destination end system of that unicast message by D. If the multicast packets received by S from D contain no SSRC option, S and D must be directly connected, without an intervening translator. No SDST option is need in this case. If the multicast packet received by S from D contain an SSRC option, S inserts an SDST option using the identifier contained in the SSRC option received from D. D then forwards the packet to the source network and transport address found in the multicast packets coming from D. The packet will thus reach the translator on the path between S and D closest to S. The arrival on that transport address tells the translator that the packet is a unicast reverse control packet. The translator determines which source it maps into the identifier contained in the SDST option and replaces the SDST identifier by that value. In other words: if a forward RTP packet carries SSRC identifier X between two systems (either two translators or an end system and a translator), the unicast reverse control packet will carry SDST with identifier X between those two systems. Example for UDP: T1 and T2 are translators between end systems S and D. In the forward direction, D sends regular RTP packets with no SSRC to (among other multicast group members) translator T2 with destination port 3456 and source port 5678; T2 inserts SSRC identifier 13 and forwards to translator T1 on source port 4590 and destination port 3456; T1 translates SSRC 13 into SSRC 8 and forwards to S using destination port 3456 and source port 12789. In the unicast reverse RPDU, site S sends the packet to translator T1, destination port 12789 with SDST value 8. T1 replaces SDST value 8 with SDST value 13 and forwards to translator T2 with destination port 4590. T2 finally sends the message with SDST value 0 to site D at destination port 5678. By its arrival port, site D determines that the RPDU is a reverse RPDU and treat it accordingly. H. Schulzrinne/S. Casner Expires 10/01/93 [Page 10] INTERNET-DRAFT RTP July 30, 1993 [Reverse control unicast packets are already identified by their destination transport address, so SSRC could be used for reverse control packets. A separate option is used to limit confusion.] Only applications that need to send or receive unicast control information flow need to implement the SDST 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| SDST | length = 1 | identifier | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 3.5 Security Options The security options below offer message integrity, authentication and privacy and the combination of the three. Support for the security options is not mandatory, but see the discussion for the ENC option. The four message integrity check options --- MIC, MICA, MICK and MICS --- are mutually exclusive, i.e., only one of them should be used for a single RPDU. All message integrity check options are computed over the fixed header, the ENC option preceding the message integrity check option (if present), the first four octets of the message integrity check option and the data (remaining header and payload) following the message integrity check option. The message integrity check options and the ENC option shall not cover the SSRC and SDST options, i.e., SSRC and SDST must be inserted between the fixed header and the ENC or message integrity check options, as SSRC and SDST are subject to change by translators that are likely not in possession of the necessary descriptor table (see below) and encryption keys. Translators that have the necessary keys and descriptor translation table may modify the contents of the RPDU, unless the MICA option is used (see MICA description). All security options carry a one-octet descriptor field. This descriptor is an index into two tables, one for the message integrity check options, one for the ENC option, established by non-RTP means, containing digest algorithms (MD2, MD5, etc.), encryption algorithms (DES variants) and encryption keys or shared secrets (for the MICK option). All sources within the same channel share the same table. The descriptor value may change during a session, for example, to use a different set of encryption keys. The descriptor value zero describes a set of default algorithms to be used: MD5 for the message digest algorithm, DES CBC for the encryption algorithm. H. Schulzrinne/S. Casner Expires 10/01/93 [Page 11] INTERNET-DRAFT RTP July 30, 1993 The MIC, MICK and MICS message integrity checks offer g_r_o_u_p_ a_u_t_h_e_n_t_i_c_a_t_i_o_n_, that is, the receiver can ascertain that the RPDU originated from a member of the group of sites sharing a common secret, but the receiver cannot authenticate which of the sources among that group sent the data. The receiver can also be assured that nobody outside the group tampered with the RPDU. ENC 8 All packet data after this option, but not the fixed header, is encrypted, using the encryption key and symmetric encryption algorithm specified by the descriptor field. The descriptor value may change over time to accomodate varying security requirements or reduce the amount of ciphertext using the same key. [For example, in a network interview, 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 [5]. The padding specified in that section is to be used. The 8-octet initialization vector (IV) may be carried unencrypted within the ENC option or generated anew for each packet. If the ENC option does not contain an initialization vector (indicated by an option length of 1), the fixed RTP header is used as the IV. [Using the fixed RTP header as the IV avoids regenerating the IV for each packet and incurs less header overhead.] For details on the tradeoffs for CBC IV use, see [6]. 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 CBC 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 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |F| ENC | length = 3 | reserved | descriptor | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | DES (CBC) initialization vector, bytes 0 through 3 | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | DES (CBC) initialization vector, bytes 4 through 7 | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |F| ENC | length = 1 | reserved | descriptor | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ H. Schulzrinne/S. Casner Expires 10/01/93 [Page 12] INTERNET-DRAFT RTP July 30, 1993 MIC 9 Messsage integrity check. The MIC option option is used only in combination with the ENC option immediately preceding it to provide privacy and group membership authentication. The message integrity check uses the digest algorithm specified by the descriptor field. The value zero implies the use of the MD5 message digest. Note that the MIC option is not separately 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| MIC | length | reserved | descriptor | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | message digest (unencrypted) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ MICA 10 Message integrity check, asymmetric encryption. Currently, only the use of the MD2 and MD5 message digest algorithms is defined, as described in RFC 1319 [7] (as corrected in Section 2.1 of RFC 1423) and RFC 1321 [8], respectively. The MD2 and MD5 message digests are 16 octets long. ``To avoid any potential ambiguity regarding the ordering of the octets of an MD2 message digest that is input as a data value to another encryption process (e.g., RSAEncryption), the following holds true. The first (or left-most displayed, if one thinks in terms of a digest's "print" representation) octet of the digest (i.e., digest[0] as specified in RFC 1319), when considered as an RSA data value, has numerical weight 2**120. The last (or right-most displayed) octet (i.e., digest[15] as specified in RFC 1319) has numerical weight 2**0.'' [RFC 1423, Section 2.1] ``To avoid any potential ambiguity regarding the ordering of the octets of a MD5 message digest that is input as an RSA data value to the RSA encryption process, the following holds true. The first (or left-most displayed, if one thinks in terms of a digest's "print" representation) octet of the digest (i.e., the low-order octet of A as specified in RFC 1321), when considered as an RSA data value, has numerical weight 2**120. The last (or right-most displayed) octet (i.e., the high-order octet of D as specified in RFC 1321) has numerical weight 2**0.'' [RFC 1423, Section 2.2] The message digest is encrypted, using asymmetric keys, with the sender's private key using the algorithm described in Section 4.2.1 of RFC 1423: ``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 H. Schulzrinne/S. Casner Expires 10/01/93 [Page 13] INTERNET-DRAFT RTP July 30, 1993 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.''. The encoding is described in detail in RFC 1423. For encrypting MD2 and MD5, the data quantity D is comprised of the 16-byte checksum, preceded by the binary sequences shown here in hexadecimal: 0x30, 0x20, 0x30, 0x0C, 0x06, 0x08, 0x2A, 0x86, 0x48, 0x86, 0xF7, 0x0D, 0x02, 0x02, 0x05, 0x00, 0x04, 0x10 for MD2 and 0x30, 0x20, 0x30, 0x0C, 0x06, 0x08, 0x2A, 0x86, 0x48, 0x86, 0xF7, 0x0D, 0x02, 0x05, 0x05, 0x00, 0x04, 0x10 for MD5. The signature is padded as necessary. The value of the padding is left unspecified. [Note: The number of non-padding bits within the signature is known to the receiver as being equal to the key length. The MIC algorithm is identified through the bytes prepended to the actual 16-byte signature.] Contrary to what is specified in RFC 1423 for privacy enhanced mail, the asymmetrically signed MIC is carried in binary, NOT represented in the printable encoding of RFC 1421, Section 4.3.2.4. The encrypted length of the signature will be equal to the modulus of the RSA encryption used, rounded to the next integral byte count. The modulus and public key is conveyed to the receivers by non-RTP means. [Note: Asymmetric keys are used since symmetric keys would not allow authentication of the individual source in the multicast case.] A translator that receives an RPDU is not allowed to modify the parts of the RPDU 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 RDPU. Support for sending or interpreting MICA options is not required. 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 | length | encrypted message-digest ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ MICK 11 Message integrity check, keyed. This message integrity check does not require encryption. In addition to the RPDU parts to be included in the message digest according to the introduction to this section, the shared secret is placed in the MICK option and included in the message digest. (The shared secret is equivalent to the key used for the MICS and ENC options, but is 16 octets long, if H. Schulzrinne/S. Casner Expires 10/01/93 [Page 14] INTERNET-DRAFT RTP July 30, 1993 necessary by padding with binary zeroes.) The shared secret in the MICK option is then replaced by the computed 128-bit digest. The receiver saves the message digest contained in the MICK option, replacing it with the shared secret key and computes the message digest in the same manner as the sender. If the RPDU has not been tampered and originated with one of the holders of the secret key, the computed message digest will agree with the digest found on reception in the MICS option. [The message integrity check follows the practice of SNMP Version 2, as described in RFC 1446, Section 1.5.1. The MICS option itself is covered by the digest in order to detect tampering with the descriptor field itself. Using the secret key in the signature instead of encrypting the MD5 message digest avoids the use of an encryption algorithm when only authentication is desired. However, the security of this approach has not been as well established as that based on encrypting message digests, as used in the MICS, MIC and MICA options.] 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 | length | reserved | descriptor | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | encrypted message digest ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ MICS 12 Message integrity check, symmetric-key encrypted. This message integrity check encrypts the message digest using DES ECB mode as described in RFC 1423, Section 3.1. 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 | length | reserved | descriptor | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | encrypted message digest ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 3.6 The Use of the Security Options Combinations of the message integrity check and ENC security options can be used to provide a variety of security services: H. Schulzrinne/S. Casner Expires 10/01/93 [Page 15] INTERNET-DRAFT RTP July 30, 1993 confidentiality: Confidentiality here means that only the intended receiver(s) can decode the received RTP packets; for others, the data contains no useful information. Confidentiality of the content is achieved by encryption using DES. The presence of encryption and the initialization vector is indicated by the ENC option. [Note: for efficiency reasons, this specification does not insist that content encryption only be used in connection with message integrity and authentication mechanisms. In most all cases, it will be obvious to the person receiving the data if he or she does not possess the right encryption key.] authentication and message integrity: In combination with certificates, the receiver can ascertain that the claimed originator is indeed the originator of the data (authentication) and that the data has not been altered after leaving the sender (message integrity). These two security services are provided by the message integrity check options. Certificates for MICA must be distributed through means outside of RTP. The services offered by MICA and MIC/MICK/MICS differ: 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, while the MICA option affords true authentication of the sender. 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). The message integrity check is applied first to the all parts of the outgoing packet to be authenticated, and the message integrity check option is prepended to those parts. Then, the packet including the message integrity check option is encrypted using the shared secret key. 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 signature contained in the message integrity check option. For this combination of security features and group authentication, the combination ENC and MIC is recommended (instead of MICS or MICK) as it yields the lowest processing overhead. A message integrity check option followed by an ENC option should not be used. H. Schulzrinne/S. Casner Expires 10/01/93 [Page 16] INTERNET-DRAFT RTP July 30, 1993 4 Real Time Control Protocol --- RTCP The real-time control protocol (RTCP) conveys minimal control and advisory information during a conference. It provides support for loosely controlled conferences, i.e., where participants enter and leave without admission control and parameter negotiation. The services provided by RTCP services enhance RTP, but an end system does not have to implement RTCP features to participate in conferences(1). RTCP does not aim to provide the services of a conference control protocol and does not provide some of the services desirable for two-party conversations. If a conference control protocol is in use, the services of RTCP should not be required. (Note: as of the writing of this document, a conference or session control protocol has not been specified within the Internet.) Unless otherwise noted, control information is carried periodically as options within RPDUs, with or without payload. RTCP packets are sent to all members of a conference. These packets are part of the same sequence nubmer 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 conference 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 conference has to wait in the worst case until it can identify the source. A receiver may remove from its list of active sites a site that it has not heard from for a given time-out period; he 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 MAIL part within the SDES option might only be sent every few messages. The item types are defined below: FMT 32 Format description. format: 6 bits The 'format' field corresponds to the index value from the 'format' RTP fixed header field, with values ranging from 0 to 63. Clock quality: 8 bits Provides an indication as to the sender-perceived quality of the timestamps in the RTP header. The octet is interpreted as a quantity indicating the maximum dispersion to a root time server measured in fractions of a second and expressed as a ------------------------------ 1. There is one exception to that rule: if an application sends FMT options, the receiver has to decode these in order to properly interpret the RTP payload. H. Schulzrinne/S. Casner Expires 10/01/93 [Page 17] INTERNET-DRAFT RTP July 30, 1993 power of two. If a source is known to be synchronized to standard time, but with an unknown dispersion, or the dispersion is greater than TBD, the value TBD is used. If the clock is based on the nominal sample rate of the source, a value of TBD is used. The clock quality indication can be used to judge how the delay measurements reported by the QOS option can be interpreted (as absolute delay or only as delay variation). It is also useful for determining to what extent several sources with different clocks can be synchronized. Format-dependent data: variable Format-dependent data may or may not appear in a FMT option. It is passed to the next layer and not interpreted by RTP. A FMT mapping changes the interpretation of a given 'format' value (as carried in the fixed RTP header) starting at the packet containing the FMT option. The new interpretation applies only to packets from the synchronization source of this packet. A sender should refrain from changing the mappings between the RTP format field and the other fields in the FMT option that have been established through a conference registry, a conference announcement protocol or otherwise. Dynamic changes to these values may result in misinterpretation of RTP payload if the packet(s) containing the FMT option are lost. 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 | length |R|R| format | clock quality | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | format-dependent data ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ SDES 33 This option provides a mapping between a numeric source identifier and one or more identifying attributes. [Several attributes were combined into one option to avoid multiple mappings from identifiers to the receiver site data structure.] For those applications where the size of a multipart SDES option would be a concern, multiple SDES options may be formed with subsets of the parts to be sent in separate packets. An end system or a bridge uses an identifier value of zero to identify itself. For each contributor, a bridge forwards the SDES information received from that contributor, but changes the SDES source identifier to correspond to the value used in the CSRC option when identifying this contributor. A bridge that contributes data H. Schulzrinne/S. Casner Expires 10/01/93 [Page 18] INTERNET-DRAFT RTP July 30, 1993 to outgoing packets should use a CSRC and select another non-zero source identifier for that traffic and send CSRC and SDES options for it. 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 network source, synchronization source and the source number within this SDES option) to identify a particular source. SDES information is specific to a particular flow identifier, unless a higher-layer control protocol defines that all packets with the same source identifier (network and transport-level source addresses and the optional SSRC value) from a set of channels defined by the control protocol are described by the same SDES. Currently, the following items are defined. Each has a structure similar to that of RTCP and RTP options, that is, a type field followed by a length field (measured in multiples of four octets). No final bit is needed since the overall length is known. All of the SDES items are optional; however, if quality-of-service monitoring is to be used, the ADDR and TSEL items need to be provided (see QOS option). type value description ADDR 1 network address of source TSEL 2 transport address CNAME 4 canonical user and host identifier, e.g., ``doe@sleepy.megacorp.com'' or ``sleepy.megacorp.com'' MAIL 5 user's electronic mail address e.g., ``John.Doe@megacorp.com'' LOC 8 geographic user location, e.g., ``Rm. 2A244, Berkeley Heights, NJ'' TXT 16 text describing the source, e.g.,``John Doe, Bit Recycler, Megacorp'' 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 more description of the content of some of these types follows: ADDR: A source may send several network addresses, but only one for each address type value. Address types are identified by the Domain Name Service Resource Record (RR) type, as specified in the current edition of the Assigned Numbers RFC. For NSAP addresses, the NSEL byte is not included. TSEL: The protocol identifier uses the IP Protocol Numbers defined H. Schulzrinne/S. Casner Expires 10/01/93 [Page 19] INTERNET-DRAFT RTP July 30, 1993 in the current edition of the Assigned Numbers RFC. The figure shows the use of the TSEL item for the TCP and UDP protocols. There must be no more than one TSEL item in an SDES option. The TSEL item should precede any address information. [Multiple concurrent transport addresses are not meaningful. The ordering simplifies processing at the receiver.] CNAME: The CNAME item must have the format ``user@host'' or ``host'', where ``host'' is the fully qualified domain name of the host where the real-time data originates from, 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. The latter is provided through the MAIL item. LOC: Depending on the application, different degrees of detail are appropriate for this item. For conference applications, a string like ``Tampere, Finland'' may be sufficient, while for an active badge system, strings like ``Room 2A244, AT&T BL MH'' might be appropriate. 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 | length | source identifier | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | type = ADDR | length | reserved | address type | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | network-layer address ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | type = ADDR | length = 2 | reserved | addr. type = 1| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | IPv4 address | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | type = TSEL | length | reserved | transport pro.| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | transport-address (port number) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ H. Schulzrinne/S. Casner Expires 10/01/93 [Page 20] INTERNET-DRAFT RTP July 30, 1993 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | type = TSEL | length | reserved | protocol = 6 | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | reserved | TCP port number | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | type = TSEL | length | reserved | protocol = 17 | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | reserved | UDP port number | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | type = CNAME | length | user and domain name ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | type = MAIL | length | electronic mail address ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | type = LOC | length | geographic location of site ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | type = TXT | length | text describing source ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ BYE 35 The BYE option indicates that a particular site is no longer active. A bridge sends BYE options with a (non-zero) content source value. An identifier value of zero indicates that the source indicated by the synchronization source (SSRC) option and network address is no longer active. 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. Each RTCP message can contain one or more BYE messages. [Multiple identifiers in a single BYE option are not allowed to avoid ambiguities between the special value of zero and any necessary padding.] 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 | length = 1 | content source identifier | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ QOS 36 Quality of service measurement. The QOS options describes statistics of a single synchronization source. The synchronization source is identified by one of the ADDR items from the SDES option together with the TSEL item from the SDES option. The SDES items H. Schulzrinne/S. Casner Expires 10/01/93 [Page 21] INTERNET-DRAFT RTP July 30, 1993 are appended directly to the fixed-length part of the QOS option, with TSEL following ADDR. For a description of these items, see the SDES option. The other fields of the option contains the number of packets received (32 bits), the number of packets expected (32 bits), the minimum delay, the maximum delay and the average delay. The delay measures are encoded as 16/16 NTP timestamps, that is, 16 bits encode the number and seconds and 16 bits the fraction of a second. A single RTCP packet may contain several QOS options. It is left to the implementor to decide how often to transmit QOS options and which sources are to be included. [The timestamp format is identical to the one used in the fixed RTP header. The quality-of-service information is identical to that carried in the reverse control 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| QOS | length | synchronization source | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | packets expected | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | packets received | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | minimum delay (seconds) | minimum delay (fraction) | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | maximum delay (seconds) | maximum delay (fraction) | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | average delay (seconds) | average delay (fraction) | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | type = ADDR | length | reserved | address type | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | network-layer address ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | type = TSEL | length | reserved | transport pro.| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | transport-address (port number) ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 5 Security Considerations IP multicast provides no direct means for a sender to know all the receivers of the data sent. RTP options make it easy for all participants in a conference to identify themselves; if deemed important for a particular H. Schulzrinne/S. Casner Expires 10/01/93 [Page 22] INTERNET-DRAFT RTP July 30, 1993 application, it is the responsibility of the application writer to make listening without identification difficult. It should be noted, however, that within an internet, privacy of the payload can generally only be assured by encryption. The periodic transmission of session messages may make it possible to detect denial-of-service attacks. For many types of payload expected to be carried in RTP packets, such as compressed audio and video, the data is very close to white noise, making statistics-based ciphertext-only attacks difficult. Without MICS/MICA options, it may even be difficult to detect automatically when the code has been broken. However, the session information is more or less constant and predictable, allowing known-plaintext attacks. Chosen-plaintext attacks appear to be difficult. Since the timestamp in the RTP header is protected by the message integrity check options, some replay attacks can be detected if the receiver can bound the maximum packet delay and clock offset of the sender. Without authentication, the SDES fields may be used to impersonate another site. Impersonation and denial-of-service attacks can be made more difficult by providing digital signatures for all or parts of a message. The MICA or MICS and ENC RTP options described in Section 3 support privacy within group communications. The issues of key distribution and a certification hierarchy are outside the scope of this document. A direct mapping of all PEM header fields into RTCP option types would be straightforward and would allow reuse of existing PEM implementations. However, it is questionable whether loose conference control is the appropriate mechanism for distributing key and certificate information. 6 RTP over network and transport protocols This section describes issues specific to carrying RPDUs over particular network and transport protocols. 6.1 Defaults The following rules apply unless superseded by protocol-specific subsections in this section. If RTP protocol data units (RPDU) are carried over underlying protocols that provide the abstraction of a continuous bit stream rather than messages, each RPDU is prefixed by a 32-bit framing field containing the length of the RPDU measured in octets, not including the framing field itself. If an RPDU 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 determine that framing H. Schulzrinne/S. Casner Expires 10/01/93 [Page 23] INTERNET-DRAFT RTP July 30, 1993 is to be used for protocols that do provide framing in order to allow carrying several RPDUs in one underlying protocol data unit. [Carrying several RPDUs in one network or transport packet reduces header overhead and may ease synchronization between different streams.] 6.2 ST-II 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 RPDU 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 byte count determines the number of bytes in the RTP header and payload to be checksummed. 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 bytes to be checked | check sum | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ ... RTP header ... +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 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. A.1 Timestamp recovery A fully specified NTP timestamp with 32 bits of full seconds and 16 bits of resolution for the fractional seconds can be easily recovered from the RTP timestamp. The following code stores timestamps as the 48-bit whole part of a double precision floating point number: #include <math.h> typedef double CLOCK_t; typedef unsigned long u_long; H. Schulzrinne/S. Casner Expires 10/01/93 [Page 24] INTERNET-DRAFT RTP July 30, 1993 #define MAX32_bit 4294967296. #define MAX31 0x7fffffff CLOCK_t extend_timestamp(t, now) u_long t; /* in: timestamp, low-order 32 bits */ double now; /* in: current local time */ { u_long high, low; /* high and low order bits of 48-bit clock */ low = fmod(x, MAX_32bit); high = now / MAX_32bit; if ((low > t) && (low - t > MAX31)) high++; else if ((low < t) && (t - low > MAX31)) high--; return high * MAX_32bit + t; } /* 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; a clock based on samples, e.g., incremented by the nominal audio frame duration, is sufficient. 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 if a packet is the beginning of a synchronization unit: CLOCK_t eos_t, t, now; int flag; struct { unsigned int ver:2; /* version number */ unsigned int flow:6; /* flow */ unsigned int o:1; /* option present */ unsigned int s:1; /* sync bit */ unsigned int format:6; /* content type */ u_short seq; /* sequence number */ u_long ts; /* time stamp */ } *h; t = extend_timestamp(h->ts, now); if (h->s) { H. Schulzrinne/S. Casner Expires 10/01/93 [Page 25] INTERNET-DRAFT RTP July 30, 1993 flag = 1; eos_t = t; } else if (flag && t > eot_t) { flag = 0; /* handle beginning of synchronization unit */ } (The structure definition has to be changed for little endian systems.) A.3 Demultiplexing and Locating the Synchronization Source For a combination of multicast or destination unicast address, destination port, the flow 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 datastructure. Synchronization sources are stored with a content source value 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 (SSRC option) and content source(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 to zero and locates a table entry based on the triple (network address and port, 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 sources carried in the CSRC option. To locate the table entry, it matches on the triple (network address and port, synchronization source identifier, content source). Note that since network addresses are only generated locally at the receiver, the receiver can choose whatever format seems most appropriate for matching. For example, a Berkeley Unix-based system may use struct sockaddr data types if it expects network sources with non-IP addresses. Acknowledgments This draft is based on discussion 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 Frederic (Xerox PARC) provided extensive editorial assistance. H. Schulzrinne/S. Casner Expires 10/01/93 [Page 26] INTERNET-DRAFT RTP July 30, 1993 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 telephone: +1 908 582 2262 electronic mail: hgs@research.att.com References [1] J. Postel, ``Internet protocol,'' Network Working Group Request for Comments RFC 791, Information Sciences Institute, Sept. 1981. [2] 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. [3] The Unicode Consortium, T_h_e_ U_n_i_c_o_d_e_ S_t_a_n_d_a_r_d_. New York, New York: Addison-Wesley, 1991. [4] 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. [5] 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. [6] V. L. Voydock and S. T. Kent, ``Security mechanisms in high-level network protocols,'' A_C_M_ C_o_m_p_u_t_i_n_g_ S_u_r_v_e_y_s_, vol. 15, pp. 135--171, June 1983. [7] J. Kaliski, Burton S., ``The md2 message-digest algorithm,'' Network Working Group Request for Comments RFC 1319, RSA Laboratories, Apr. 1992. [8] R. Rivest, ``The MD5 message-digest algorithm,'' Network Working Group H. Schulzrinne/S. Casner Expires 10/01/93 [Page 27] INTERNET-DRAFT RTP July 30, 1993 Request for Comments RFC 1321, IETF, Apr. 1992. [9] P. Mockapetris, ``Domain names -- concepts and facilities,'' Network Working Group Request for Comments RFC 1034, ISI, Nov. 1987. [10] P. Mockapetris, ``Domain names -- implementation and specification,'' Network Working Group Request for Comments RFC 1035, ISI, Nov. 1987. H. Schulzrinne/S. Casner Expires 10/01/93 [Page 28]