+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| timestamp of primary encoding "P" |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| synchronization source (SSRC) identifier |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|1| T140c PT | timestamp offset of "R" | "R" block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|0| T140c PT | "R" T140block counter | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +
| "R" T.140 encoded redundant data |
+ +---------------+
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
As a follow-on to the previous example, the example below shows the
next RTP packet in the sequence that does contain a new real
T140block when using the audio/t140c payload format. This example
has 2 levels of redundancy and one primary data block. Since the
previous primary block was empty, no redundant data is included for
that block. This is because when using the audio/t140c payload
format, any previously transmitted "empty" T140blocks are NOT
included as redundant data in subsequent packets.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P|X| CC=0 |M| "RED" PT | sequence number of primary |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| timestamp of primary encoding "P" |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| synchronization source (SSRC) identifier |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|1| T140c PT | timestamp offset of "R1" | "R1" block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|0| T140c PT | "R1" T140block counter | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +
| "R1" T.140 encoded redundant data |
+ +---------------+
| | "P" T140block |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| counter | "P" T.140 encoded primary data |
+-+-+-+-+-+-+-+-+ +
| |
+ +---------------+
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
7.2. SDP Examples
Below is an example of SDP describing RTP text interleaved with G.711
audio packets within the same RTP session from port 7200 and at a
maximum text rate of 6 characters per second:
m=audio 7200 RTP/AVP 0 98
a=rtpmap:98 t140c/8000
a=fmtp:98 cps=6
Below is an example using RFC 2198 to provide the recommended two
levels of redundancy to the text packets in an RTP session with
interleaving text and G.711 at a text rate no faster than 20
characters per second:
m=audio 7200 RTP/AVP 0 98 100
a=rtpmap:98 t140c/8000
a=fmtp:98 cps=20
a=rtpmap:100 red/8000
a=fmtp:100 98/98/98
Note: While these examples utilize the RTP/AVP profile, it is not
intended to limit the scope of this memo to use with only that
profile. Rather, any appropriate profile may be used in conjunction
with this memo.
8. Security Considerations
All of the security considerations from section 14 of RFC 3550 [2]
apply.
8.1. Confidentiality
Since the intention of the described payload format is to carry text
in a text conversation, security measures in the form of encryption
are of importance. The amount of data in a text conversation session
is low, and therefore any encryption method MAY be selected and
applied to T.140 session contents or to the whole RTP packets.
Secure Realtime Transport Protocol (SRTP) [13] provides a suitable
method for ensuring confidentiality.
8.2. Integrity
It may be desirable to protect the text contents of an RTP stream
against manipulation. SRTP [13] provides methods for providing
integrity that MAY be applied.
8.3. Source Authentication
Measures to make sure that the source of text is the intended one can
be accomplished by a combination of methods.
Text streams are usually used in a multimedia control environment.
Security measures for authentication are available and SHOULD be
applied in the registration and session establishment procedures, so
that the identity of the sender of the text stream is reliably
associated with the person or device setting up the session. Once
established, SRTP [13] mechanisms MAY be applied to ascertain that
the source is maintained the same during the session.
9. Congestion Considerations
The congestion considerations from section 10 of RFC 3550 [2],
section 6 of RFC 2198 [3], and any used profile (e.g., the part about
congestion in section 2 of RFC 3551 [10]) apply with the following
application-specific considerations.
Automated systems MUST NOT use this format to send large amounts of
text at a rate significantly above that which a human user could
enter.
Even if the network load from users of text conversation is usually
very low, for best-effort networks an application MUST monitor the
packet loss rate and take appropriate actions to reduce its sending
rate if this application sends at higher rate than what TCP would
achieve over the same path. The reason is that this application, due
to its recommended usage of two or more redundancy levels, is very
robust against packet loss. At the same time, due to the low bit-
rate of text conversations, if one considers the discussion in RFC
3714 [12], this application will experience very high packet loss
rates before it needs to perform any reduction in the sending rate.
If the application needs to reduce its sending rate, it SHOULD NOT
reduce the number of redundancy levels below the default amount
specified in section 4. Instead, the following actions are
RECOMMENDED in order of priority:
- Increase the shortest time between transmissions described in
section 5.1 from the recommended 300 ms to 500 ms that is the
highest value allowable according to T.140.
- Limit the maximum rate of characters transmitted.
- Increase the shortest time between transmissions to a higher value,
not higher than 5 seconds. This will cause unpleasant delays in
transmission, beyond what is allowed according to T.140, but text
will still be conveyed in the session with some usability.
- Exclude participants from the session.
Please note that if the reduction in bit-rate achieved through the
above measures is not sufficient, the only remaining action is to
terminate the session.
As guidance, some load figures are provided here as examples based on
use of IPv4, including the load from IP, UDP, and RTP headers without
compression.
- Experience tells that a common mean character transmission rate
during a complete PSTN text telephony session in reality is around
2 characters per second.
- A maximum performance of 20 characters per second is enough even
for voice-to-text applications.
- With the (unusually high) load of 20 characters per second, in a
language that make use of three-octet UTF-8 characters, two
redundant levels, and 300 ms between transmissions, the maximum
load of this application is 3500 bits/s.
- When the restrictions mentioned above are applied, limiting
transmission to 10 characters per second, using 5 s between
transmissions, the maximum load of this application in a language
that uses one octet per UTF-8 character is 300 bits/s.
Note also, that this payload can be used in a congested situation as
a last resort to maintain some contact when audio and video media
need to be stopped. The availability of one low bit-rate stream for
text in such adverse situations may be crucial for maintaining some
communication in a critical situation.
10. IANA Considerations
This document defines one RTP payload format named "t140" and an
associated MIME type "audio/t140c". They have been registered by the
IANA.
10.1. Registration of MIME Media Type audio/t140c
MIME media type name: audio
MIME subtype name: t140c
Required parameters:
rate: The RTP timestamp clock rate, which is equal to the
sampling rate. This parameter SHOULD have the same value as
for any audio codec packets interleaved in the same RTP
stream.
Optional parameters:
cps: The maximum number of characters that may be received
per second. The default value is 30.
Encoding considerations: T.140 text can be transmitted with RTP
as specified in RFC 4351.
Security considerations: See section 8 of RFC 4351.
Interoperability considerations: None
Published specification: ITU-T T.140 Recommendation.
RFC 4351.
Applications which use this media type:
Text communication systems and text conferencing tools that
transmit text associated with audio and within the same RTP
session as the audio, such as PSTN gateways that transmit
audio and text signals between two PSTN textphone users
over an IP network.
Additional information: This type is only defined for transfer
via RTP.
Magic number(s): None
File extension(s): None
Macintosh File Type Code(s): None
Person & email address to contact for further information:
Paul E. Jones
E-mail: paulej@packetizer.com
Intended usage: COMMON
Author / Change controller:
Paul E. Jones | IETF avt WG delegated from the IESG
paulej@packetizer.com |
10.2. SDP Mapping of MIME Parameters
The information carried in the MIME media type specification has a
specific mapping to fields in the Session Description Protocol (SDP)
[7], which is commonly used to describe RTP sessions. When SDP is
used to specify sessions employing the audio/t140c format, the
mapping is as follows:
- The MIME type ("audio") goes in SDP "m=" as the media name.
- The MIME subtype (payload format name) goes in SDP "a=rtpmap" as
the encoding name. For audio/t140c, the clock rate MAY be set
to any value, and SHOULD be set to the same value as for any
audio packets in the same RTP stream.
- The parameter "cps" goes in SDP "a=fmtp" attribute.
- When the payload type is used with redundancy according to RFC
2198, the level of redundancy is shown by the number of elements
in the slash-separated payload type list in the "fmtp" parameter
of the redundancy declaration as defined in RFC 2198 [3].
10.3. Offer/Answer Consideration
In order to achieve interoperability within the framework of the
offer/answer model [9], the following consideration should be made:
- The "cps" parameter is declarative. Both sides may provide a
value, which is independent of the other side.
11. Acknowledgements
The authors want to thank Stephen Casner, Magnus Westerlund, and
Colin Perkins for valuable support with reviews and advice on
creation of this document; Mickey Nasiri at Ericsson Mobile
Communication for providing the development environment; Michele
Mizarro for verification of the usability of the payload format for
its intended purpose; and Andreas Piirimets for editing support.
12. Normative References
[1] ITU-T Recommendation T.140 (1998) - Text conversation protocol
for multimedia application, with amendment 1, (2000).
[2] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson,
"RTP: A Transport Protocol for Real-Time Applications", STD 64,
RFC 3550, July 2003.
[3] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V., Handley, M.,
Bolot, J., Vega-Garcia, A., and S. Fosse-Parisis, "RTP Payload
for Redundant Audio Data", RFC 2198, September 1997.
[4] Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997.
[5] ISO/IEC 10646-1: (1993), Universal Multiple Octet Coded
Character Set.
[6] Yergeau, F., "UTF-8, a transformation format of ISO 10646", STD
63, RFC 3629, November 2003.
[7] Handley, M. and V. Jacobson, "SDP: Session Description
Protocol", RFC 2327, April 1998.
[8] Rosenberg, J. and H. Schulzrinne, "An RTP Payload Format for
Generic Forward Error Correction", RFC 2733, December 1999.
[9] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
Session Description Protocol (SDP)", RFC 3264, June 2002.
[10] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video
Conferences with Minimal Control", STD 65, RFC 3551, July 2003.
[11] Postel, J., "Internet Protocol", STD 5, RFC 791, September 1981.
13. Informative References
[12] Floyd, S. and J. Kempf, "IAB Concerns Regarding Congestion
Control for Voice Traffic in the Internet", RFC 3714, March
2004.
[13] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC
3711, March 2004.
[14] Schulzrinne, H. and S. Petrack, "RTP Payload for DTMF Digits,
Telephony Tones and Telephony Signals", RFC 2833, May 2000.
[15] Hellstrom, G. and P. Jones, "RTP Payload for Text Conversation",
RFC 4103, June 2005.
[16] ITU-T Recommendation F.703, Multimedia Conversational Services,
Nov 2000.
[17] Casner, S. and P. Hoschka, "MIME Type Registration of RTP
Payload Formats", RFC 3555, July 2003.
Authors’ Addresses
Gunnar Hellstrom
Omnitor AB
Renathvagen 2
SE-121 37 Johanneshov
Sweden
Phone: +46 708 204 288 / +46 8 556 002 03
Fax: +46 8 556 002 06
EMail: gunnar.hellstrom@omnitor.se
Paul E. Jones
Cisco Systems, Inc.
7025 Kit Creek Rd.
Research Triangle Park, NC 27709
USA
Phone: +1 919 392 6948
EMail: paulej@packetizer.com
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