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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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- 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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48168 | file | 2006-11-30 16:18:24 -0500 (Thu, 30 Nov 2006) | 2 lines
Do not do a partial bridge for Google Talk since we need to handle STUN. (issue #8448 reported by phsultan)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48107 | file | 2006-11-29 11:50:33 -0500 (Wed, 29 Nov 2006) | 10 lines
Merged revisions 48106 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48106 | file | 2006-11-29 11:47:10 -0500 (Wed, 29 Nov 2006) | 2 lines
If the frame was duplicated before writing out then we need to free it. (issue #8429 reported by edguy3)
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freed, but I don't
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.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47944 | file | 2006-11-22 16:47:43 -0500 (Wed, 22 Nov 2006) | 2 lines
Video will never reach Packet2Packet bridging and can do more harm then good.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47897 | file | 2006-11-21 12:32:27 -0500 (Tue, 21 Nov 2006) | 2 lines
If we have the non standard G726-32 setting turned on we want to return G726-32 to the SDP, not our AAL2 string. (issue #8330 reported by voipgate)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47852 | file | 2006-11-20 10:58:50 -0500 (Mon, 20 Nov 2006) | 2 lines
Only remove/destroy the RTCP I/O item if it exists.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47645 | file | 2006-11-14 23:45:24 -0500 (Tue, 14 Nov 2006) | 2 lines
If NAT detection is turned on or already detected then say NAT is active when setting the remote RTP peer when doing early bridging. (issue #8365 reported by marcelbarbulescu)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47639 | file | 2006-11-14 19:14:07 -0500 (Tue, 14 Nov 2006) | 2 lines
Turn notice about unknown RTCP packet type into a debug message instead.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47053 | tilghman | 2006-11-02 17:49:13 -0600 (Thu, 02 Nov 2006) | 2 lines
More changes making the CLI more consistent with "category verb arguments" (continuation of issue 8236)
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can't do anything with it yet. Ideally a first step would be a
passthrough mode.
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I currently don't see this as a bug that needs to be fixed in 1.4/1.2 too,
but feel free to backport if you see it that way. RTCP now binds to
ALL IP addresses on the host, RTP to a specific address.
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r45452 | file | 2006-10-17 23:02:08 -0400 (Tue, 17 Oct 2006) | 2 lines
Don't segfault if you're using a channel driver that doesn't turn RTCP on
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r44628 | file | 2006-10-06 17:08:54 -0400 (Fri, 06 Oct 2006) | 2 lines
Remove the seqno check for RFC2833, the handler is smart enough to not need it.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r44605 | file | 2006-10-06 14:46:28 -0400 (Fri, 06 Oct 2006) | 2 lines
When the sequence number rolls over then reset the recorded sequence number for DTMF (issue #8106 reported by bungalow)
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patch provided in bugnote, with minor changes.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r44090 | pcadach | 2006-10-01 01:20:38 +0600 (Вск, 01 Окт 2006) | 1 line
Allow one-way RTP streams (device->Asterisk)
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r43798 | file | 2006-09-27 15:10:59 -0400 (Wed, 27 Sep 2006) | 2 lines
Compensate for out of order packets better if RFC2833 compensation is turned on.
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minor - abstract early bridging into the technology so that we don't always assume they use RTP and try it.
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(issue 7983).
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going off hold.
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bridge to occur
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option for setting our own packetization as apposed to just doing
what is asked.
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instead of Packet2Packet since video isn't supported there yet. (reported by PCadach in #asterisk-bugs)
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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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bridged to, and also use it in chan_sip so we know to ignore the no RTP activity checking
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upon switching to P2P callback bridging. Once P2P callback bridging has ended, then restore callback mode.
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a way that under a straight bridge (ie: no transcoding and no DTMF detection) the core is not touched at all and no frames pass through (not even null frames). This is accomplished by stealing the file descriptors from the channel and using the provided IO context with a custom callback. When a channel is placed on hold this bridge is broken so audio can flow from the core to the other side. When a channel is off hold this bridge is re-established.
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that's what RTP-level packet bridging is all about!
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- restructured build tree and makefiles to eliminate recursion problems
- support for embedded modules
- support for static builds
- simpler cross-compilation support
- simpler module/loader interface (no exported symbols)
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