r48273 dealt with the comments and such, this deals with the code itself.
(This couldn't have been easily done if it weren't for 48273 - thanks again for that merbanan)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48276 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48258 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- remove coef_in.h and coef_out.h that was only included as data definitions in fskmodem.c
If you understand spanish, please help us translate the comments in fskmodem.c. Thanks!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48205 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Encapsulate RTP timers to the RTP structure, so we have one set for video and one for audio
- Document RTP keepalive configuration option
- Cleanup and document the monitor support function to hangup on RTP timeouts
- Add RTP keepalive to SIP show settings
Imported from 1.4 with modifications for trunk.
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in both the append and non-append modes. Also, always truncating the partial
write makes the behavior of the function more consistent, where in any type of
write, no partial result is left in the buffer.
Thanks for the feedback, luigi
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48109 65c4cc65-6c06-0410-ace0-fbb531ad65f3
and also add a more verbose comment explaining why it is only needed in the
case of appending to the string for any curious readers that come along in the
future.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48103 65c4cc65-6c06-0410-ace0-fbb531ad65f3
an extra allocation on a path where we have way too many already.
Unfortunately the AMI-over-HTTP requires multiple copies,
because we need to generate a header, then the raw output to
an intermediate buffer, then convert it to html/xml, and
finally copy everything into a malloc'ed buffer because
that's what the generic_http_callback interface expects.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48090 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ast_dynamic_str_thread_build_va_couldnt_we_choose_a_shorter_name()
I am unsure whether the truncation of the string in case of a failed
attempt should be done unconditionally. See the XXX mark.
Russel, ideas ?
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
to simplify the body of the main loop of the accepting thread.
Rename purge_unused() to purge_events() so one knows what the
function does.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48081 65c4cc65-6c06-0410-ace0-fbb531ad65f3
+ use a wrapper around ast_carefulwrite(), used in two places,
to make life easier when we decide to use a different interface
to the socket.
+ put an ast_verbose() message on astman_append on a case that
should never happen now that we use a temporary file for
AMI-over-HTTP sessions
+ document and slightly simplify process_events() by removing
unnecessary parentheses.
+ in get_input(), use ast_wait_for_input() instead of poll().
We may want to move to a completely non-blocking
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48080 65c4cc65-6c06-0410-ace0-fbb531ad65f3
causing the http threads to do busy waiting around the socket...
Fix the mistake, sorry for the inconvenience!
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Because https is more secure than http, it usually
makes sense to keep this service more open than the
one on the unencrypted port.
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function doing work on the socket. This is another generalization
to provide a generic mechanism to open TCP/TLS socket with a thread
managing the accpet and children threads managing the individual
sessions.
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and then run a service on it.
The code in manager.c does essentially the same things,
so we will be able to reuse the code in here (probably
moving it to netsock.c or another appropriate library file).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48008 65c4cc65-6c06-0410-ace0-fbb531ad65f3
have a clear understanding of the frame allocation/deallocation, so I just mark this
for investigation. (Reported by Ed Guy). We're trying to see if a free() hurts...
- Doxygen comments on p2p rtp bridge stuff. I am a bit worried about shortcutting
rtcp this way, but will need feedback from rtcp gurus. This should work for
video calls too, and possibly UDPTL.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48003 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Document cause codes
- Document a bit more on channel variables - global, predefined and local
- Fix some doxygen in channel.h. Adding one comment for two definitions does not
work. They won't be copied to each.
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