Asterisk Documentation 1.6.1 realtimetext.txt

-=NOTE: These pages are automatically updated once per
day from the Asterisk subversion repository when the repository changes revisions. Any
changes made to this page will be automatically overwritten with the
latest version from http://svn.digium.com/view/asterisk/branches/.

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.

During a call sometimes there are losses of T.140 packets and a solution to that is to 
use redundancy in the media stream (RTP).
See  "http://en.wikipedia.org/wiki/Text_over_IP"http://en.wikipedia.org/wiki/Text_over_IP
and RFC 5194 for more information.

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
	allow=t140red
	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 and 140red and 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 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.

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 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.



-=NOTE: These pages are automatically updated once per
day from the Asterisk subversion repository when the repository changes revisions. Any
changes made to this page will be automatically overwritten with the
latest version from http://svn.digium.com/view/asterisk/branches/.

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.

During a call sometimes there are losses of T.140 packets and a solution to that is to 
use redundancy in the media stream (RTP).
See  "http://en.wikipedia.org/wiki/Text_over_IP"http://en.wikipedia.org/wiki/Text_over_IP
and RFC 5194 for more information.

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
	allow=t140red
	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 and 140red and 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 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.

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 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.



Created by: josiahbryan, Last modification: Tue 25 of May, 2010 (08:44 UTC)
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