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implementing T.140 support in RTP.
T.140/RFC 4351 is TDD over IP - text telephony for hearing impaired.
It defines a realtime text chat, much like the old "talk" application
in Unix.
T.140 is character by character in real time. It's not
the same as our current MESSAGE format - that is more like IM, but
can be gatewayed to MESSAGE with a text "codec" if needed.
More patches will follow, as soon as we've separated this code from
the video capabilities functions in the videocaps branch.
Code by John Martin, Aupix (disclaimer on file)
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documentation...
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THANK YOU!
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46154 | kpfleming | 2006-10-24 19:26:17 -0500 (Tue, 24 Oct 2006) | 2 lines
add passthrough and file format support for G.722 16KHz audio (issue #5084, original patch by andrew, updated by mithraen)
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option for setting our own packetization as apposed to just doing
what is asked.
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for uses in cases where you *know* that it will do no good. This patch was
inspired by file for use in some work of his on mixmonitor/chanspy.
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Russell Bryant and the RTP portion done by myself, Muffinlicious Joshua Colp. This has gone through so many discussions/revisions it's not funny but we finally have it!
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There are some situations in Asterisk where ast_frame and/or iax_frame
structures are rapidly allocatted and freed (at least 50 times per second
for one call).
This code significantly improves the performance of ast_frame_header_new(),
ast_frdup(), ast_frfree(), iax_frame_new(), and iax_frame_free() by keeping
a thread-local cache of these structures and using frames from the cache
whenever possible instead of calling malloc/free every time.
This commit also converts the ast_frame and iax_frame structures to use the
linked list macros.
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use this way
add infrastructure for whispering onto a channel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38422 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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versions of Asterisk will still work properly
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split support for G726-32 into RFC3551 and AAL2 packing orders, since both are in use
change "G726-32" to be RFC3551 packing order, in spite of devices that use AAL2 order with this MIME type
add ability to directly transcode between packing orders
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a new implementation of a fixed size jitterbuffer, as well as support for the
existing adaptive jitterbuffer implementation. (issue #3854, Slav Klenov)
Thank you very much to Slav Klenov of Securax and all of the people involved
in the testing of this feature for all of your hard work!
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support sending control frames with payload
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in app_dial to support the use of this mode.
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datalen of a frame, which is a very common operation.
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const-ify some more APIs
remove 'type' field from ast_channel, in favor of the one in the channel's tech structure
allow string field module users to specify the 'chunk size' for pool allocations
update chan_alsa to be compatible with recent const-ification patches
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have to allocate one on the stack
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return a chain of frames
cleanup code in ast_read()
add AST_FRAME_DTMF_BEGIN and AST_FRAME_DTMF_END so that variable-length DTMF events can be supported
teach chan_zap to send DTMF_BEGIN and DTMF_END when appropriate
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old local rtp stuff...
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This should prevent us from unintentionally changing variable
values when they're returned from pbx_builtin_getvar_helper.
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documentation cleanup
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adjust txgain/rxgain if they are able (only Zap channels at this time)
modify app_chanspy to use new gain option
reformat app_chanspy to match coding guidelines
add user-controlled volume adjustment to app_meetme (issue #4170, heavily modified to actually work on Zap channels)
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remove deprecated config_old.c/config_old.h
remove unused cvsid.h
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support non-SLINEAR moh streams (issue #4908)
add external app to feed TCP stream into Asterisk for moh (issue #4908)
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(bug #4058)
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#3865)
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