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Real-time text in Asterisk
--------------------------
The SIP channel has support for real-time text conversation calls in Asterisk (T.140).
This is a way to perform text based conversations in combination with other media,
most often video. The text is sent character by character as a media stream.
The supported real-time text codec is t.140.
Real-time text redundancy support is now available in Asterisk.
ITU-T T.140
-----------
You can find more information about T.140 at www.itu.int. RTP is used for the transport T.140,
as specified in RFC 4103.
How to enable T.140
-------------------
In order to enable real-time text with redundancy in Asterisk, modify sip.conf to add:
[general]
disallow=all
allow=ulaw
allow = alaw
allow=t140
textsupport=yes
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.
The codec settings may change, depending on your phones. The important settings here are to allow
t140 to enable text support.
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 (t140) is handled on channel and application level in Asterisk conveyed in
text frames, with the subtype "t140". Text conveyed in such frames usually only contains 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.
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 eConf –audio/video/text softphone.
- SIPcon1 - open source SIP audio/text softphone available in Sourceforge.
Limitations
-----------
A known general problem with Asterisk is that when a client which uses audio/video/T.140 calls to
an 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
[general]
videosupport=yes
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.
Credits
-------
- Asterisk real-time text support is developed by AuPix
- Asterisk real-time text redundancy support (in trunk) is developed by Omnitor
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.
Olle E. Johansson, Edvina AB, has been a liason between the Asterisk project and this project.
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