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85 lines
3.8 KiB
Plaintext
85 lines
3.8 KiB
Plaintext
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Real-time text in Asterisk
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--------------------------
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The SIP channel has support for real-time text conversation calls in Asterisk (T.140).
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This is a way to perform text based conversations in combination with other media,
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most often video. The text is sent character by character as a media stream.
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During a call sometimes there are losses of T.140 packets and a solution to that is to
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use redundancy in the media stream (RTP).
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See "http://en.wikipedia.org/wiki/Text_over_IP"http://en.wikipedia.org/wiki/Text_over_IP
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and RFC 5194 for more information.
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The supported real-time text codec is t.140.
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Real-time text redundancy support is now available in Asterisk.
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ITU-T T.140
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-----------
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You can find more information about T.140 at www.itu.int. RTP is used for the transport T.140,
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as specified in RFC 4103.
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How to enable T.140
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-------------------
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In order to enable real-time text with redundancy in Asterisk, modify sip.conf to add:
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[general]
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disallow=all
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allow=ulaw
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allow = alaw
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allow=t140
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allow=t140red
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textsupport=yes
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videosupport=yes ; needed for proper SDP handling even if only text and voice calls are handled
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allow=h263 ; at least one video codec as H.261, H.263 or H.263+ is needed.
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The codec settings may change, depending on your phones. The important settings here are to allow
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t140 and 140red and enable text support.
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General information about real-time text support in Asterisk
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------------------------------------------------------------
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With the configuration above, calls will be supported with any combination of real-time text,
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audio and video.
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Text for both t140 and t140red is handled on channel and application level in Asterisk conveyed in
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Text frames, with the subtype "t140". Text is conveyed in such frames usually only containing one or
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a few characters from the real-time text flow. The packetization interval is 300 ms, handled on lower
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RTP level, and transmission redundancy level is 2, causing one original and two redundant transmissions
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of all text so that it is reliable even in high packet loss situations. Transmitting applications do not
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need to bother about the transmission interval. The t140red support handles any buffering needed during
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the packetization intervals.
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Clients known to support text, audio/text or audio/video/text calls with Asterisk:
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----------------------------------------------------------------------------------
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- Omnitor Allan eC - SIP audio/video/text softphone
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- AuPix APS-50 - audio/video/text softphone.
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- France Telecom eConf –audio/video/text softphone.
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- SIPcon1 - open source SIP audio/text softphone available in Sourceforge.
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Limitations
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-----------
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A known general problem with Asterisk is that when a client which uses audio/video/T.140 calls to
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an Asterisk with T.140 media offered but video support not specified. In this case Asterisk handles
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the sdp media description (m=) incorrectly, and the sdp response is not created correctly.
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To solve this problem, turn on video support in Asterisk.
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Modify sip.conf to add
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[general]
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videosupport=yes
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allow=h263 ; at least one video codec as H.261, H.263 or H.263+ is needed.
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The problem with sdp is a bug and is reported to Asterisk bugtracker, it has id 0012434.
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Credits
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-------
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- Asterisk real-time text support is developed by AuPix
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- Asterisk real-time text redundancy support is developed by Omnitor
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The work with Asterisk real-time text redundancy was supported with funding from the National Institute
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on Disability and Rehabilitation Research (NIDRR), U.S. Department of Education, under grant number
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H133E040013 as part of a co-operation between the Telecommunication Access Rehabilitation Engineering
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Research Center of the University of Wisconsin – Trace Center joint with Gallaudet University, and Omnitor.
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Olle E. Johansson, Edvina AB, has been a liason between the Asterisk project and this project.
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