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diff --git a/doc/realtimetext.txt b/doc/realtimetext.txt new file mode 100644 index 000000000..1f706fe90 --- /dev/null +++ b/doc/realtimetext.txt @@ -0,0 +1,76 @@ +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. + |