Real-time text

Real-time text in Asterisk

The SIP channel has support for real-time text conversation calls in Asterisk (T.140). During a call sometimes there are losses of T.140 packets and a solution to that is to use redundancy. See and RFC 5194 for more information.

Real-time text redundancy support is available in asterisk trunk since revision 116237 and in videocaps since revision 118927.

Supported real-time text codec is t.140.

ITU-T T.140

You can find more information about T.140 at RFC 4103 is used to transport T.140 is used. More information can be found at the IETF homepage

This codec is implemented in Asterisk trunk and Videocaps branch

In order to enable real-time text redundancy in Asterisk trunk or Videocaps, modify sip.conf to add:

videosupport=yes ; needed for proper SDP handling even if only text and voice calls are handled
allow=h263 ; at least one video codec as H.261, H.263 or H.263+ is needed.

General information about real-time text support in Asterisk

With the configuration above, calls will be supported with any combination of real-time text, audio and video.

Text for both t140 and t140red is handled on channel and application level in Asterisk conveyed in Text frames, with the subtype "t140". Text is conveyed in such frames usually only containing one or a few characters from the real-time text flow. The packetization interval is 300 ms, handled on lower RTP level, and transmission redundancy level is 2, causing one original and two redundant transmissions of all text so that it is reliable even in high packet loss situations. Transmitting applications do not need to bother about the transmission interval. The t140red support handles any buffering needed during the packetization intervals.

Asterisk version 1.8.0 supports T.140 natively

Clients known to support text, audio/text or audio/video/text calls with Asterisk:

  • Omnitor Allan eC - SIP audio/video/text softphone
  • AuPix APS-50 - audio/video/text softphone.
  • France Telecom Conf –audio/video/text softphone.
  • SIPcon1 – open source SIP audio/text softphone available in Sourceforge.


Here is a patch for videocaps branch revision 151045 that fixes timestamp offset, clock frequency and m-bit

A known general problem with Asterisk is that when a client which uses audio/video/T.140 calls to a Asterisk with T.140 media offered but video support not specified. In this case Asterisk handles the sdp media description (m=) incorrectly, and the sdp response is not created correctly. To solve this problem, turn on video support in Asterisk.

Modify sip.conf to add;

allow=h263 ; at least one video codec as H.261, H.263 or H.263+ is needed.

The problem with sdp is a bug and is reported to Asterisk bugtracker, it has id 0012434.


Asterisk real-time text redundancy support is developed by Omnitor.
Asterisk real-time text support is developed by AuPix.

The work with Asterisk real-time text redundancy was supported with funding from the National Institute on Disability and Rehabilitation Research (NIDRR), U.S. Department of Education, under grant number H133E040013 as part of a co-operation between the Telecommunication Access Rehabilitation Engineering Research Center of the University of Wisconsin – Trace Center joint with Gallaudet University, and Omnitor. 

Created by: pejo, Last modification: Mon 28 of May, 2012 (10:53 UTC) by dominic16y
Please update this page with new information, just login and click on the "Edit" or "Discussion" tab. Get a free login here: Register Thanks! - Find us on Google+