idnits 2.17.1 draft-ietf-avt-audio-t140c-00.txt: Checking boilerplate required by RFC 5378 and the IETF Trust (see https://trustee.ietf.org/license-info): ---------------------------------------------------------------------------- ** It looks like you're using RFC 3978 boilerplate. You should update this to the boilerplate described in the IETF Trust License Policy document (see https://trustee.ietf.org/license-info), which is required now. -- Found old boilerplate from RFC 3978, Section 5.5 on line 901. -- Found old boilerplate from RFC 3979, Section 5, paragraph 2 on line 876. -- Found old boilerplate from RFC 3979, Section 5, paragraph 3 on line 882. ** The document seems to lack an RFC 3978 Section 5.1 IPR Disclosure Acknowledgement -- however, there's a paragraph with a matching beginning. Boilerplate error? ** This document has an original RFC 3978 Section 5.4 Copyright Line, instead of the newer IETF Trust Copyright according to RFC 4748. ** This document has an original RFC 3978 Section 5.5 Disclaimer, instead of the newer disclaimer which includes the IETF Trust according to RFC 4748. ** The document seems to lack an RFC 3979 Section 5, para. 1 IPR Disclosure Acknowledgement -- however, there's a paragraph with a matching beginning. Boilerplate error? Checking nits according to https://www.ietf.org/id-info/1id-guidelines.txt: ---------------------------------------------------------------------------- ** The document seems to lack a 1id_guidelines paragraph about 6 months document validity -- however, there's a paragraph with a matching beginning. Boilerplate error? == No 'Intended status' indicated for this document; assuming Proposed Standard Checking nits according to https://www.ietf.org/id-info/checklist : ---------------------------------------------------------------------------- No issues found here. Miscellaneous warnings: ---------------------------------------------------------------------------- == The copyright year in the RFC 3978 Section 5.4 Copyright Line does not match the current year == Line 470 has weird spacing: '... second inter...' -- The document seems to lack a disclaimer for pre-RFC5378 work, but may have content which was first submitted before 10 November 2008. If you have contacted all the original authors and they are all willing to grant the BCP78 rights to the IETF Trust, then this is fine, and you can ignore this comment. If not, you may need to add the pre-RFC5378 disclaimer. (See the Legal Provisions document at https://trustee.ietf.org/license-info for more information.) -- The document date (August 2004) is 7156 days in the past. Is this intentional? Checking references for intended status: Proposed Standard ---------------------------------------------------------------------------- (See RFCs 3967 and 4897 for information about using normative references to lower-maturity documents in RFCs) == Unused Reference: '15' is defined on line 854, but no explicit reference was found in the text -- Possible downref: Non-RFC (?) normative reference: ref. '1' -- Possible downref: Non-RFC (?) normative reference: ref. '5' ** Obsolete normative reference: RFC 2327 (ref. '7') (Obsoleted by RFC 4566) ** Obsolete normative reference: RFC 2733 (ref. '8') (Obsoleted by RFC 5109) -- Obsolete informational reference (is this intentional?): RFC 2833 (ref. '14') (Obsoleted by RFC 4733, RFC 4734) -- Obsolete informational reference (is this intentional?): RFC 2793 (ref. '15') (Obsoleted by RFC 4103) Summary: 8 errors (**), 0 flaws (~~), 4 warnings (==), 9 comments (--). Run idnits with the --verbose option for more detailed information about the items above. -------------------------------------------------------------------------------- 2 Network Working Group G. Hellstrom 3 Internet Draft Omnitor AB 4 P. Jones 5 Expires: February 2005 Cisco Systems, Inc. 6 August 2004 8 RTP Payload for Text Conversation interleaved in an audio stream 10 Status of this Memo 12 By submitting this Internet-Draft, we certify that any applicable 13 patent or other IPR claims of which we are aware have been 14 disclosed, and any of which we become aware will be disclosed, in 15 accordance with RFC 3668 (BCP 79). 17 By submitting this Internet-Draft, we accept the provisions of 18 Section 3 of RFC 3667 (BCP 78). 20 Internet-Drafts are working documents of the Internet Engineering 21 Task Force (IETF), its areas, and its working groups. Note that 22 other groups may also distribute working documents as Internet- 23 Drafts. 25 Internet-Drafts are draft documents valid for a maximum of six 26 months and may be updated, replaced, or obsoleted by other 27 documents at any time. It is inappropriate to use Internet-Drafts 28 as reference material or cite them other than as "work in 29 progress". 31 The list of current Internet-Drafts can be accessed at 32 http://www.ietf.org/ietf/1id-abstracts.txt 34 The list of Internet-Draft Shadow Directories can be accessed at 35 http://www.ietf.org/shadow.html 37 This document is a submission of the IETF AVT WG. Comments should 38 be directed to the AVT WG mailing list, avt@ietf.org. 40 Abstract 42 This memo describes how to carry real time text conversation 43 session contents in RTP packets. Text conversation session contents 44 are specified in ITU-T Recommendation T.140. 46 One payload format is described for transmitting audio and text 47 data within one single RTP session. 49 This RTP payload description recommends a method to include 50 redundant text from already transmitted packets in order to reduce 51 the risk of text loss caused by packet loss. 53 Table of Contents 55 1. Introduction...................................................3 56 2. Conventions used in this document..............................4 57 3. Usage of RTP...................................................4 58 3.1 Motivations and rationale..................................4 59 3.2 Payload Format for Transmission of audio/t140c Data........4 60 3.3 The "T140block"............................................4 61 3.4 Synchronization of Text with Other Media...................5 62 3.5 Synchronization considerations for the audio/t140c format..5 63 3.6 RTP packet header..........................................6 64 4. Protection against loss of data................................6 65 4.1 Payload Format when using Redundancy.......................7 66 4.2 Using redundancy with the audio/t140c format...............7 67 5. Recommended Procedure..........................................8 68 5.1 Recommended Basic Procedure................................8 69 5.2 Transmission before and after "Idle Periods"...............8 70 5.3 Detection of Lost Text Packets.............................9 71 5.4 Compensation for Packets Out of Order......................9 72 6. Parameter for Character Transmission Rate.....................10 73 7. Examples......................................................10 74 7.1 RTP Packetization Examples for the audio/t140c format.....10 75 7.2 SDP Examples..............................................12 76 8. Security Considerations.......................................12 77 8.1 Confidentiality...........................................12 78 8.2 Integrity.................................................13 79 8.3 Source authentication.....................................13 80 9. Congestion Considerations.....................................13 81 10. IANA considerations..........................................14 82 10.1 Registration of MIME Media Type audio/t140c..............15 83 10.2 SDP mapping of MIME parameters...........................16 84 10.3 Offer/Answer Consideration...............................16 85 11. Authors' Addresses...........................................16 86 12. Acknowledgements.............................................17 87 13. Normative References.........................................17 88 14. Informative References.......................................17 89 15. Intellectual Property Statement..............................18 90 16. Copyright Statement..........................................18 92 [Notes to RFC Editor: 93 1. All references to RFC XXXX are to be replaced by references to 94 the RFC number of this memo, when published. ] 96 1. Introduction 98 This document defines a payload type for carrying text conversation 99 session contents in RTP [2] packets. Text conversation session 100 contents are specified in ITU-T Recommendation T.140 [1]. Text 101 conversation is used alone or in connection to other conversational 102 facilities such as video and voice, to form multimedia conversation 103 services. Text in multimedia conversation sessions is sent 104 character-by-character as soon as it is available, or with a small 105 delay for buffering. 107 The text is intended to be entered by human users from a keyboard, 108 handwriting recognition, voice recognition or any other input 109 method. The rate of character entry is usually at a level of a few 110 characters per second or less. In general, only one or a few new 111 characters are expected to be transmitted with each packet. Small 112 blocks of text may be prepared by the user and pasted into the user 113 interface for transmission during the conversation, occasionally 114 causing packets to carry more payload. 116 T.140 specifies that text and other T.140 elements must be 117 transmitted in ISO 10646-1[5] code with UTF-8 [6] transformation. 118 That makes it easy to implement internationally useful applications 119 and to handle the text in modern information technology 120 environments. The payload of an RTP packet following this 121 specification consists of text encoded according to T.140 without 122 any additional framing. A common case will be a single ISO 10646 123 character, UTF-8 encoded. 125 T.140 requires the transport channel to provide characters without 126 duplication and in original order. Text conversation users expect 127 that text will be delivered with no or a low level of lost 128 information. 130 Therefore a mechanism based on RTP is specified here. It gives text 131 arrival in correct order, without duplication, and with detection 132 and indication of loss. It also includes an optional possibility to 133 repeat data for redundancy to lower the risk of loss. Since packet 134 overhead is usually much larger than the T.140 contents, the 135 increase in bandwidth with the use of redundancy is minimal. 137 By using RTP for text transmission in a multimedia conversation 138 application, uniform handling of text and other media can be 139 achieved in, as examples, conferencing systems, firewalls, and 140 network translation devices. This, in turn, eases the design and 141 increases the possibility for prompt and proper media delivery. 143 This document introduces a method of transporting text interleaved 144 with voice within the same RTP session. 146 2. Conventions used in this document 148 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 149 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in 150 this document are to be interpreted as described in RFC 2119 [4]. 152 3. Usage of RTP 154 The payload format for real-time text transmission with RTP [2] 155 described in this memo is intended for use between PSTN gateways 156 and is called audio/t140c. 158 3.1 Motivations and rationale 160 The audio/t140c payload specification is intended to allow gateways 161 that are interconnecting two PSTN networks to interleave, through a 162 single RTP session, audio and text data received on the PSTN 163 circuit. This is comparable to the way in which DTMF is extracted 164 and transmitted within an RTP session [14]. 166 The audio/t140c format SHALL NOT be used for other applications 167 than PSTN gateway applications. In such applications, a specific 168 profiling document MAY make it REQUIRED for a specific application. 169 The reason to prefer to use audio/t140c could be for gateway 170 application where the ports are a limited and scarce resource. 172 3.2 Payload Format for Transmission of audio/t140c Data 174 An audio/t140c conversation RTP payload format consists of a 16-bit 175 "T140block counter" carried in network byte order (see RFC 791 [11] 176 Annex B), followed by one and only one "T140block" (see section 177 3.3). The fields in the RTP header are set as defined in section 178 3.6. 180 The T140block counter MUST be initialized to zero the first time 181 that a packet containing a T140block is transmitted and MUST be 182 incremented by 1 each time that a new block is transmitted. Once 183 the counter reaches the value 0xFFFF, the counter is reset to 0 the 184 next time the counter is incremented. This T140block counter is 185 used to detect lost blocks and to avoid duplication of blocks. 187 For the purposes of readability, the remainder of this document 188 only refers to the T140block without making explicit reference to 189 the T140block counter. Readers should understand that when using 190 the audio/t140c format, the T140block counter MUST always precede 191 the actual T140block, including redundant data transmissions. 193 3.3 The "T140block" 195 T.140 text is UTF-8 coded as specified in T.140 with no extra 196 framing. The T140block contains one or more T.140 code elements as 197 specified in [1]. Most T.140 code elements are single ISO 10646 198 [5] characters, but some are multiple character sequences. Each 199 character is UTF-8 encoded [6] into one or more octets. Each block 200 MUST contain an integral number of UTF-8 encoded characters 201 regardless of the number of octets per character. Any composite 202 character sequence (CCS) SHOULD be placed within one block. 204 3.4 Synchronization of Text with Other Media 206 Usually, each medium in a session utilizes a separate RTP stream. 207 As such, if synchronization of the text and other media packets is 208 important, the streams MUST be associated when the sessions are 209 established and the streams MUST share the same reference clock 210 (refer to the description of the timestamp field as it relates to 211 synchronization in section 5.1 of RFC 3550). Association of RTP 212 streams can be done through the CNAME field of RTCP SDES function. 213 It is dependent on the particular application and is outside the 214 scope of this document. 216 3.5 Synchronization considerations for the audio/t140c format. 218 The audio/t140c packets are generally transmitted as interleaved 219 packets between voice packets or other kinds of audio packets with 220 the intention to create one common audio signal in the receiving 221 equipment to be used for alternating between text and voice. The 222 audio/t140c payload is then used to play out audio signals 223 according to a PSTN textphone coding method (usually a modem). 225 One should observe the RTP timestamps of the voice, text, or other 226 audio packets in order to reproduce the stream correctly when 227 playing out the audio. Note also, that incoming text from a PSTN 228 circuit might be at a higher bit-rate than can be played out on an 229 egress PSTN circuit. As such, it is possible that, on the egress 230 side, a gateway may not complete the play out of the text packets 231 before it is time to play the next voice packet. Given that this 232 application is primarily for the benefit of users of PSTN textphone 233 devices, it is strongly RECOMMENDED that all received text packets 234 be properly reproduced on the egress gateway before considering any 235 other subsequent audio packets. 237 If necessary, voice and other audio packets should be discarded in 238 order to properly reproduce the text signals on the PSTN circuit, 239 even if the text packets arrive late. 241 The PSTN textphone users commonly use turn-taking indicators in the 242 text stream, so it can be expected that as long as text is 243 transmitted, it is valid text and should be given priority over 244 voice. 246 Note that the usual RTP semantics apply with regards to switching 247 payload formats within an RTP session. A sender MAY switch between 248 "audio/t140c" and some other format within an RTP session, but MUST 249 NOT send overlapping data using two different audio formats within 250 an RTP session. This does not prohibit an implementation from being 251 split into two logical parts to send overlapping data, each part 252 using a different SSRC and sending its own RTP and RTCP (such an 253 end point will appear to others in the session as two participants 254 with different SSRC, but the same RTCP SDES CNAME). Further details 255 around using multiple payloads in an RTP session can be found in 256 RFC 3550 [2]. 258 3.6 RTP packet header 260 Each RTP packet starts with a fixed RTP header. The following 261 fields of the RTP fixed header are specified for T.140 text 262 streams: 264 Payload Type (PT): The assignment of an RTP payload type is 265 specific to the RTP profile under which this payload format is 266 used. For profiles that use dynamic payload type number 267 assignment, this payload format can be identified by the MIME 268 type "audio/t140c" (see section 10). If redundancy is used per 269 RFC 2198, another payload type number needs to be provided for 270 the redundancy format. The MIME type for identifying RFC 2198 is 271 available in RFC 3555. 273 Sequence number: The definition of sequence numbers is available in 274 RFC 3550 [2]. Character loss is detected through the T140block 275 counter when using the audio/t140c payload format. 277 Timestamp: The RTP Timestamp encodes the approximate instance of 278 entry of the primary text in the packet. For audio/t140c, the 279 clock frequency MAY be set to any value, and SHOULD be set to the 280 same value as for any audio packets in the same RTP stream in 281 order to avoid RTP timestamp rate switching. The value SHOULD be 282 set by out of band mechanisms. Sequential packets MUST NOT use 283 the same timestamp. Since packets do not represent any constant 284 duration, the timestamp cannot be used to directly infer packet 285 loss. 287 M-bit: The M-bit MUST be included. The first packet in a session, 288 and the first packet after an idle period, SHOULD be 289 distinguished by setting the marker bit in the RTP data header to 290 one. The marker bit in all other packets MUST be set to zero. 291 The reception of the marker bit MAY be used for refined methods 292 for detection of loss. 294 4. Protection against loss of data 296 Consideration must be devoted to keeping loss of text caused by 297 packet loss within acceptable limits. (See ITU-T F.703 [16]) 298 The default method that MUST be used when no other method is 299 explicitly selected is redundancy in accordance with RFC 2198 [3]. 300 When this method is used, the original text and two redundant 301 generations SHOULD be transmitted if the application or end-to-end 302 conditions do not call for other levels of redundancy to be used. 304 Other protection methods MAY be used. Forward Error Correction 305 mechanisms as per RFC 2733 [8] or any other mechanism with the 306 purpose of increasing the reliability of text transmission MAY be 307 used as an alternative or complement to redundancy. Text data MAY 308 be sent without additional protection if end-to-end network 309 conditions allow the text quality requirements specified in ITU-T 310 F.703 [16] to be met in all anticipated load conditions. 312 4.1 Payload Format when using Redundancy 314 When using the format with redundant data, the transmitter may 315 select a number of T140block generations to retransmit in each 316 packet. A higher number introduces better protection against loss 317 of text but marginally increases the data rate. 319 The RTP header is followed by one or more redundant data block 320 headers, one for each redundant data block to be included. Each of 321 these headers provides the timestamp offset and length of the 322 corresponding data block plus a payload type number indicating the 323 payload format audio/t140c. 325 After the redundant data block headers follows the redundant data 326 fields carrying T140blocks from previous packets, and finally the 327 new (primary) T140block for this packet. 329 Redundant data that would need a timestamp offset higher than 16383 330 due to its age at transmission MUST NOT be included in transmitted 331 packets. 333 4.2 Using redundancy with the audio/t140c format 335 Since sequence numbers are not provided in the redundant header and 336 since the sequence number space is shared by all audio payload 337 types within an RTP session, a sequence number in the form of a 338 T140block counter is added to the T140block for transmission. This 339 allows the redundant T140block data corresponding to missing 340 primary data to be retrieved and used properly into the stream of 341 received T140block data when using the audio/t140c payload format. 343 All non-empty redundant data block MUST contain the same data as a 344 T140block previously transmitted as primary data, and be identified 345 with a T140block counter equating to the original T140block counter 346 for that T140block. 348 The T140block counters preceding the text in the T140block, enables 349 the ordering by the receiver. If there is a gap in the T140block 350 counter value of received audio/t140c packets, and if there are 351 redundant T140blocks with T140block counters matching those that 352 are missing, the redundant T140blocks may be substituted for the 353 missing T140blocks. 355 The value of the length field in the redundant header indicates the 356 length of the concatenated T140block counter and the T140block. 358 5. Recommended Procedure 360 This section contains RECOMMENDED procedures for usage of the 361 payload format. Based on the information in the received packets, 362 the receiver can: 364 - reorder text received out of order. 365 - mark where text is missing because of packet loss. 366 - compensate for lost packets by using redundant data. 368 5.1 Recommended Basic Procedure 370 Packets are transmitted when there is valid T.140 data to transmit. 372 T.140 specifies that T.140 data MAY be buffered for transmission 373 with a maximum buffering time of 500 ms. A buffering time of 300 ms 374 is RECOMMENDED, when the application or end-to-end network 375 conditions are not known to require another value. 377 If no new data is available for a longer period than the buffering 378 time, the transmission process is in an idle period. 380 When new text is available for transmission after an idle period, 381 it is RECOMMENDED to send it as soon as possible. After this 382 transmission, it is RECOMMENDED to buffer T.140 data in buffering 383 time intervals, until next idle period. This is done in order to 384 keep the maximum bit rate usage for text at a reasonable level. The 385 buffering time MUST be selected so that text users will perceive a 386 real time text flow. 388 5.2 Transmission before and after "Idle Periods". 390 When valid T.140 data has been sent and no new T.140 data is 391 available for transmission after the selected buffering time, an 392 empty T140block SHOULD be transmitted. This situation is regarded 393 to be the beginning of an idle period. The procedure is recommended 394 in order to more rapidly detect potentially missing text before an 395 idle period or when the audio stream switches from the transmission 396 of audio/t140c to some other form of audio. 398 An empty T140block contains no data, neither T.140 data nor a 399 T140block counter. 401 When redundancy is used, transmission continues with a packet at 402 every transmission timer expiration and insertion of an empty 403 T.140block as primary, until the last non-empty T140block has been 404 transmitted as primary and as redundant data with all intended 405 generations of redundancy. The last packet before an idle period 406 will contain only one non-empty T140block as redundant data, and 407 the empty primary T140block. 409 When using the audio/t140c payload format, empty T140blocks sent as 410 primary data SHOULD NOT be included as redundant T140blocks, as it 411 would simply be a waste of bandwidth to send them and it would 412 introduce a risk of false detection of loss. 414 After an idle period, the transmitter SHOULD set the M-bit to one 415 in the first packet with new text. 417 5.3 Detection of Lost Text Packets 419 Receivers detect the loss of an audio/t140c packet by observing the 420 value of the T140block counter in a subsequent audio/t140c packet. 422 Missing data SHOULD be marked by insertion of a missing text marker 423 in the received stream for each missing T140block, as specified in 424 ITU-T T.140 Addendum 1 [1]. 426 Procedures based on detection of the packet with the M-bit set to 427 one MAY be used to reduce the risk for introducing false markers of 428 loss. False detection will also be avoided when using audio/t140c 429 by observing the value of the T140block counter value. 431 If two successive packets have the same number of redundant 432 generations, it SHOULD be treated as the general redundancy level 433 for the session. Change of the general redundancy level SHOULD only 434 be done after an idle period. 436 5.4 Compensation for Packets Out of Order 438 For protection against packets arriving out of order, the following 439 procedure MAY be implemented in the receiver. If analysis of a 440 received packet reveals a gap in the sequence and no redundant data 441 is available to fill that gap, the received packet SHOULD be kept 442 in a buffer to allow time for the missing packet(s) to arrive. It 443 is RECOMMENDED that the waiting time be limited to 1 second. 445 If a packet with a T140block belonging to the gap arrives before 446 the waiting time expires, this T140block is inserted into the gap 447 and then consecutive T140blocks from the leading edge of the gap 448 may be consumed. Any T140block which does not arrive before the 449 time limit expires should be treated as lost and a missing text 450 marker inserted ( see section 5.3 ). 452 6. Parameter for Character Transmission Rate 454 In some cases, it is necessary to limit the rate at which 455 characters are transmitted. For example, when a PSTN gateway is 456 interworking between an IP device and a PSTN textphone, it may be 457 necessary to limit the character rate from the IP device in order 458 to avoid throwing away characters in case of buffer overflow at the 459 PSTN gateway. 461 To control the character transmission rate, the MIME parameter 462 "cps" in the "fmtp" attribute [7] is defined (see section 10 ). It 463 is used in SDP with the following syntax: 465 a=fmtp: cps= 467 The field is populated with the payload type that is used 468 for text. The field contains an integer representing the 469 maximum number of characters that may be received per second. The 470 value shall be used as a mean value over any 10 second interval. 471 The default value is 30. 473 Examples of use in SDP are found in section 7.2. 475 In receipt of this parameter, devices MUST adhere to the request by 476 transmitting characters at a rate at or below the specified 477 value. 479 7. Examples 481 7.1 RTP Packetization Examples for the audio/t140c format 483 Below is an example of an audio/t140c RTP packet without 484 redundancy. 486 0 1 2 3 487 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 1 488 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 489 |V=2|P|X| CC=0 |M| T140c PT | sequence number | 490 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 491 | timestamp (8000Hz) | 492 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 493 | synchronization source (SSRC) identifier | 494 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 495 | T140block counter | T.140 encoded data | 496 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +---------------+ 497 | | 498 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 500 Below is an example of an RTP packet with one redundant T140block 501 using audio/t140c payload format. The primary data block is 502 empty, which is the case when transmitting a packet for the 503 sole purpose of forcing the redundant data to be transmitted 504 in the absence of any new data. Note that since this is the 505 audio/t140c payload format, the redundant block of T.140 data is 506 immediately preceded with a T140block counter. 508 0 1 2 3 509 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 1 510 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 511 |V=2|P|X| CC=0 |M| "RED" PT | sequence number of primary | 512 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 513 | timestamp of primary encoding "P" | 514 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 515 | synchronization source (SSRC) identifier | 516 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 517 |1| T140c PT | timestamp offset of "R" | "R" block length | 518 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 519 |0| T140c PT | "R" T140block counter | | 520 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + 521 | "R" T.140 encoded redundant data | 522 + +---------------+ 523 | | 524 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 526 As a follow-on to the previous example, the example below shows 527 the next RTP packet in the sequence which does contain a new real 528 T140block when using the audio/t140c payload format. This 529 example has 2 levels of redundancy and one primary data block. 530 Since the previous primary block was empty, no redundant data 531 is included for that block. This is because when using the 532 audio/t140c payload format, any previously transmitted "empty" 533 T140blocks are NOT included as redundant data in subsequent 534 packets. 536 0 1 2 3 537 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 1 538 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 539 |V=2|P|X| CC=0 |M| "RED" PT | sequence number of primary | 540 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 541 | timestamp of primary encoding "P" | 542 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 543 | synchronization source (SSRC) identifier | 544 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 545 |1| T140c PT | timestamp offset of "R1" | "R1" block length | 546 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 547 |0| T140c PT | "R1" T140block counter | | 548 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ + 549 | "R1" T.140 encoded redundant data | 550 + +---------------+ 551 | | "P" T140block | 552 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 553 | counter | "P" T.140 encoded primary data | 554 +-+-+-+-+-+-+-+-+ + 555 | | 556 + +---------------+ 557 | | 558 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 560 7.2 SDP Examples 562 Below is an example of SDP describing RTP text interleaved with 563 G.711 audio packets within the same RTP session from port 7200 and 564 at a maximum text rate of 6 characters per second: 566 m=audio 7200 RTP/AVP 0 98 567 a=rtpmap:98 t140c/8000 568 a=fmtp:98 cps=6 570 Below is an example using RFC 2198 to provide the recommended two 571 levels of redundancy to the text packets in an RTP session with 572 interleaving text and G.711 at a text rate no faster than 20 573 characters per second: 575 m=audio 7200 RTP/AVP 0 98 100 576 a=rtpmap:98 t140c/8000 577 a=fmtp:98 cps=20 578 a=rtpmap:100 red/8000 579 a=fmtp:100 98/98/98 581 Note - While these examples utilize the RTP/AVP profile, it is not 582 intended to limit the scope of this memo to use with only that 583 profile. Rather, any appropriate profile may be used in 584 conjunction with this memo. 586 8. Security Considerations 588 All of the security considerations from section 14 of RFC 3550 [2] 589 apply. 591 8.1 Confidentiality 593 Since the intention of the described payload format is to carry 594 text in a text conversation, security measures in the form of 595 encryption are of importance. The amount of data in a text 596 conversation session is low and therefore any encryption method MAY 597 be selected and applied to T.140 session contents or to the whole 598 RTP packets. SRTP [13] provides a suitable method for ensuring 599 confidentiality. 601 8.2 Integrity 603 It may be desirable to protect the text contents of an RTP stream 604 against manipulation. SRTP [13] provides methods for providing 605 integrity that MAY be applied. 607 8.3 Source authentication 609 Measures to make sure that the source of text is the intended one 610 can be accomplished by a combination of methods. 612 Text streams are usually used in a multimedia control environment. 613 Security measures for authentication are available and SHOULD be 614 applied in the registration and session establishment procedures, 615 so that the identity of the sender of the text stream is reliably 616 associated with the person or device setting up the session. Once 617 established, SRTP [13] mechanisms MAY be applied to ascertain that 618 the source is maintained the same during the session. 620 9. Congestion Considerations 622 The congestion considerations from section 10 of RFC 3550 [2], 623 section 6 of RFC 2198 [3] and any used profile, e.g. the section 624 about congestion in chapter 2 of RFC 3551 [10] apply with the 625 following application specific considerations. 627 Automated systems MUST NOT use this format to send large amounts of 628 text at a rate significantly above that which a human user could 629 enter. 631 Even if the network load from users of text conversation is usually 632 very low, for best-effort networks an application MUST monitor the 633 packet loss rate and take appropriate actions to reduce its sending 634 rate if this application sends at higher rate than what TCP would 635 achieve over the same path. The reason is that this application, 636 due to its recommended usage of two or more redundancy levels, is 637 very robust against packet loss. At the same time, due to the low 638 bit-rate of text conversations, if one considers the discussion in 639 RFC 3714 [12], this application will experience very high packet 640 loss rates before it needs to perform any reduction in the sending 641 rate. 643 If the application needs to reduce its sending rate, it SHOULD NOT 644 reduce the number of redundancy levels below the default amount 645 specified in section 4. Instead, the following actions are 646 RECOMMENDED in order of priority: 648 - Increase the shortest time between transmissions described in 649 section 5.1 from the recommended 300 ms to 500 ms that is the 650 highest value allowable according to T.140. 652 - Limit the maximum rate of characters transmitted. 654 - Increase the shortest time between transmissions to a higher 655 value, not higher than 5 seconds. This will cause unpleasant 656 delays in transmission, beyond what is allowed according to 657 T.140, but text will still be conveyed in the session with some 658 usability. 660 - Exclude participants from the session. 662 Please note that if the reduction in bit-rate achieved through the 663 above measures are not sufficient, the only remaining action is to 664 terminate the session. 666 As guidance, some load figures are provided here as examples based 667 on use of IPv4, including the load from IP, UDP and RTP headers 668 without compression. 670 -Experience tells that a common mean character transmission rate 671 during a complete PSTN text telephony session in reality is around 672 2 characters per second. 674 -A maximum performance of 20 characters per second is enough even 675 for voice to text applications. 677 -With the (unusually high) load of 20 characters per second, in a 678 language that make use of three octets UTF-8 characters, two 679 redundant levels and 300 ms between transmissions, the maximum load 680 of this application is 3500 bits/s. 682 -When the restrictions mentioned above are applied, limiting 683 transmission to 10 characters per second, using 5 s between 684 transmissions, the maximum load of this application in a language 685 that uses one octet per UTF-8 character is 300 bits/s. 687 Note also, that this payload can be used in a congested situation 688 as a last resort to maintain some contact when audio and video 689 media need to be stopped. The availability of one low bit-rate 690 stream for text in such adverse situations may be crucial for 691 maintaining some communication in a critical situation. 693 10. IANA considerations 695 This document defines one RTP payload format named "t140" and an 696 associated MIME type "audio/t140c", to be registered by IANA. 698 10.1 Registration of MIME Media Type audio/t140c 700 MIME media type name: audio 702 MIME subtype name: t140c 704 Required parameters: 705 rate: The RTP timestamp clock rate, which is equal to the 706 sampling rate. This parameter SHOULD have the same value as 707 for any audio codec packets interleaved in the same RTP 708 stream. 710 Optional parameters: 711 cps: The maximum number of characters that may be received 712 per second. The default value is 30. 714 Encoding considerations: T.140 text can be transmitted with RTP 715 as specified in RFC XXXX. 717 Security considerations: See section 8 of RFC XXXX. 719 Interoperability considerations: None 721 Published specification: ITU-T T.140 Recommendation. 722 RFC XXXX. 724 Applications which use this media type: 725 Text communication systems and text conferencing tools that 726 transmit text associated with audio and within the same RTP 727 session as the audio, such as PSTN gateways that transmit 728 audio and text signals between two PSTN textphone users 729 over an IP network. 731 Additional information: This type is only defined for transfer 732 via RTP. 734 Magic number(s): None 735 File extension(s): None 736 Macintosh File Type Code(s): None 738 Person & email address to contact for further information: 739 Paul E. Jones 740 E-mail: paulej@packetizer.com 742 Intended usage: COMMON 744 Author / Change controller: 745 Paul E. Jones | IETF avt WG 746 paulej@packetizer.com | 748 10.2 SDP mapping of MIME parameters 750 The information carried in the MIME media type specification has a 751 specific mapping to fields in the Session Description Protocol 752 (SDP) [7], which is commonly used to describe RTP sessions. When 753 SDP is used to specify sessions employing the audio/t140c format, 754 the mapping is as follows: 756 - The MIME type ("audio") goes in SDP "m=" as the media name. 758 - The MIME subtype (payload format name) goes in SDP "a=rtpmap" 759 as the encoding name. For audio/t140c, the clock rate MAY be 760 set to any value, and SHOULD be set to the same value as for 761 any audio packets in the same RTP stream. 763 - The parameter "cps" goes in SDP "a=fmtp" attribute. 765 - When the payload type is used with redundancy according to 766 RFC 2198, the level of redundancy is shown by the number of 767 elements in the slash-separated payload type list in the 768 "fmtp" parameter of the redundancy declaration as defined in 769 RFC 2198 [3]. 771 10.3 Offer/Answer Consideration 773 In order to achieve interoperability within the framework of the 774 offer/answer model [9], the following consideration should be made: 776 - The "cps" parameter is declarative. Both sides may provide a 777 value, which is independent of the other side. 779 11. Authors' Addresses 781 Gunnar Hellstrom 782 Omnitor AB 783 Renathvagen 2 784 SE-121 37 Johanneshov 785 Sweden 786 Phone: +46 708 204 288 / +46 8 556 002 03 787 Fax: +46 8 556 002 06 788 E-mail: gunnar.hellstrom@omnitor.se 790 Paul E. Jones 791 Cisco Systems, Inc. 792 7025 Kit Creek Rd. 793 Research Triangle Park, NC 27709 794 USA 795 Phone: +1 919 392 6948 796 E-mail: paulej@packetizer.com 798 12. Acknowledgements 800 The authors want to thank Stephen Casner, Magnus Westerlund and 801 Colin Perkins for valuable support with reviews and advice on 802 creation of this document, to Mickey Nasiri at Ericsson Mobile 803 Communication for providing the development environment, Michele 804 Mizarro for verification of the usability of the payload format for 805 its intended purpose, and Andreas Piirimets for editing support. 807 13. Normative References 809 [1] ITU-T Recommendation T.140 (1998) - Text conversation protocol 810 for multimedia application, with amendment 1, (2000). 812 [2] Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson, 813 "RTP: A Transport Protocol for Real-Time Applications", RFC 814 3550, July 2003. 816 [3] Perkins, C., Kouvelas, I., Hardman, V., Handley, M. and J. 817 Bolot, "RTP Payload for Redundant Audio Data", RFC 2198, 818 September 1997. 820 [4] Bradner, S., "Key words for use in RFCs to Indicate 821 Requirement Levels", BCP 14, RFC 2119, March 1997. 823 [5] ISO/IEC 10646-1: (1993), Universal Multiple Octet Coded 824 Character Set. 826 [6] Yergeau, F., "UTF-8, a transformation format of ISO 10646", 827 RFC 3629, December 2003. 829 [7] Handley, M., Jacobson, V., "SDP: Session Description 830 Protocol", RFC 2327, April 1998. 832 [8] Rosenberg, J., Schulzrinne, H., "An RTP Payload Format for 833 Generic Forward Error Correction", RFC 2733, December 1999. 835 [9] Rosenberg, J., Schulzrinne, H., "An Offer/Answer Model with 836 the Session Description Protocol (SDP)", RFC 3264, June 2002. 838 [10] Schultzrinne, J., Perkins, C., "RTP Profile for Audio and 839 Video Conference with Minimal Control", RFC 3551, July 2003. 841 [11] Postel, J.,"Internet Protocol", RFC 791, 1981. 843 14. Informative References 845 [12] Floyd, S., Kempf, J., IAB Concerns Regarding Congestion 846 Control for Voice Traffic in the Internet, RFC 3714,March 2004 848 [13] Baugher, McGrew, Carrara, Naslund, Norrman, The Secure Real- 849 Time Transport Protocol (SRTP), RFC 3711, March 2004. 851 [14] Schulzrinne, H., Petrack, S., "RTP Payload for DTMF Digits, 852 Telephony Tones and Telephony Signals", RFC 2833, May 2000. 854 [15] Hellstrom, G., "RTP Payload for text conversation.", RFC2793, 855 2000 857 [16] ITU-T Recommendation F.703, Multimedia Conversational 858 Services, Nov 2000. 860 15. Intellectual Property Statement 862 The IETF takes no position regarding the validity or scope of any 863 Intellectual Property Rights or other rights that might be claimed 864 to pertain to the implementation or use of the technology described 865 in this document or the extent to which any license under such 866 rights might or might not be available; nor does it represent that 867 it has made any independent effort to identify any such rights. 868 Information on the IETF's procedures with respect to rights in IETF 869 Documents can be found in RFC 3667 (BCP 78) and RFC 3668 (BCP 79). 871 Copies of IPR disclosures made to the IETF Secretariat and any 872 assurances of licenses to be made available, or the result of an 873 attempt made to obtain a general license or permission for the use 874 of such proprietary rights by implementers or users of this 875 specification can be obtained from the IETF on-line IPR repository 876 at http://www.ietf.org/ipr. 878 The IETF invites any interested party to bring to its attention any 879 copyrights, patents or patent applications, or other proprietary 880 rights that may cover technology that may be required to implement 881 this standard. Please address the information to the IETF at 882 ietf-ipr@ietf.org. 884 16. Copyright Statement 886 Copyright (C) The Internet Society (2004). 888 This document is subject to the rights, licenses and restrictions 889 contained in BCP 78, and except as set forth therein, the authors 890 retain all their rights. 892 Disclaimer of Validity 894 This document and the information contained herein are provided on 895 an "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE 896 REPRESENTS OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY AND 897 THE INTERNET ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, 898 EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT 899 THE USE OF THE INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR 900 ANY IMPLIED WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A 901 PARTICULAR PURPOSE.