e2a50de88f
Clarify why we are reading in a frame in the Packet2Packet bridge.
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2006-12-30 18:27:13 +00:00
c6c83cf01e
Merged revisions 49066 via svnmerge from
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r49066 | file | 2006-12-30 00:46:57 -0500 (Sat, 30 Dec 2006) | 2 lines
If the Packet2Packet bridge is being broken because of a masquerade then attempt to read a frame in so the masquerade actually happens. Otherwise weirdness will occur. (issue #8696 reported by kjotte)
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2006-12-30 05:49:17 +00:00
adca0ff14b
Merged revisions 49006 via svnmerge from
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r49006 | kpfleming | 2006-12-27 16:06:56 -0600 (Wed, 27 Dec 2006) | 2 lines
since these variables all have static duration, none of them need initializers (they default to zero anyway)
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2006-12-27 22:14:33 +00:00
7f61b822c1
Merged revisions 48964 via svnmerge from
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r48964 | file | 2006-12-25 23:31:58 -0500 (Mon, 25 Dec 2006) | 2 lines
Add an API call that initializes an RTP structure. We need this because chan_sip is cheeky and uses a temporary RTP structure for codec purposes, and the API calls that are used rely on the lock. (Pointed out on asterisk-dev by Andy Wang)
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2006-12-26 04:34:07 +00:00
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Merged revisions 48506 via svnmerge from
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r48506 | file | 2006-12-15 14:55:28 -0500 (Fri, 15 Dec 2006) | 2 lines
Turn payload_lock into bridge_lock and make it encompass all RTP structure contents that may relate to bridge information, including who we are bridged to.
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2006-12-15 19:57:04 +00:00
f6649ae0af
Merged revisions 48472 via svnmerge from
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r48472 | file | 2006-12-14 12:36:12 -0500 (Thu, 14 Dec 2006) | 2 lines
Payload values on the RTP structure can change AFTER a bridge has started. This comes from the packet handling of the SIP response when indication that it was answered has been sent. Therefore we need to protect this data with a lock when we read/write. (issue #8232 reported by tgrman)
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2006-12-14 17:39:16 +00:00
1c4c365377
Merged revisions 48461 via svnmerge from
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r48461 | file | 2006-12-13 22:33:30 -0500 (Wed, 13 Dec 2006) | 2 lines
Remove direct RTCP bridging. I've come to the conclusion that we should handle this through the core and not just forward it on. Should solve a few bugs.
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2006-12-14 03:39:39 +00:00
c3052f7a7e
Merged revisions 48381 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48381 | file | 2006-12-11 00:36:45 -0500 (Mon, 11 Dec 2006) | 2 lines
Merge in my latest RTP changes. Break out RTP and RTCP callback functions so they no longer share a common one.
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2006-12-11 05:38:57 +00:00
17a2888d2e
Staticize one, and Constify a bunch of usage strings for CLI commands.
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2006-12-06 07:28:56 +00:00
fe53552f41
Doxygen updates
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2006-12-05 20:39:13 +00:00
00bf07b12e
Well, yes...
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2006-12-05 11:09:23 +00:00
b8fcae6d75
Reserving flags for coming code (currently in the "videocaps" branch)
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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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2006-12-05 10:52:53 +00:00
c23bc46089
- Disable RTP timeouts during T.38 transmission
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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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2006-12-02 12:05:40 +00:00
869101028b
Merged revisions 48168 via svnmerge from
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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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2006-11-30 21:22:01 +00:00
2bee4aba4d
Change logging for p2p rtp bridge mode
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2006-11-29 19:44:06 +00:00
d44b349211
Merged revisions 48107 via svnmerge from
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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
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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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2006-11-29 16:53:27 +00:00
7991366506
- Adding comment on suspicious memory allocation. Seems like it's never freed, but I don't
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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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2006-11-25 09:45:57 +00:00
b50fc7a502
Merged revisions 47944 via svnmerge from
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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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2006-11-22 21:49:11 +00:00
a69ac09748
Merged revisions 47897 via svnmerge from
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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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2006-11-21 17:34:22 +00:00
03a7adf8ce
Use RTP/RTCP fds on the RTP structure, don't bother storing them.
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2006-11-20 16:06:10 +00:00
b2b966eda8
Merged revisions 47852 via svnmerge from
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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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2006-11-20 16:04:14 +00:00
993c6823e6
Merged revisions 47645 via svnmerge from
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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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2006-11-15 04:47:52 +00:00
5861048fb6
Merged revisions 47639 via svnmerge from
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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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2006-11-15 00:15:38 +00:00
79f75ec09a
Merged revisions 47053 via svnmerge from
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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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2006-11-02 23:55:59 +00:00
2cb07fbaa4
In debug mode, recognize that someone is sending zrtp, even though we
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can't do anything with it yet. Ideally a first step would be a
passthrough mode.
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2006-10-30 16:59:02 +00:00
52a5d63a2d
Bind RTCP to the same IP as RTP.
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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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2006-10-29 20:21:33 +00:00
0ca6a42d7e
fix various spelling mistakes in comments (issue #8237 , jmls)
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2006-10-26 17:52:15 +00:00
88efcea05e
Merged revisions 46154 via svnmerge from
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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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2006-10-25 00:32:23 +00:00
bb8926d50c
Merged revisions 45452 via svnmerge from
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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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2006-10-18 03:03:37 +00:00
62e6417b21
Merged revisions 44628 via svnmerge from
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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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2006-10-06 21:10:42 +00:00
85625f3505
Merged revisions 44605 via svnmerge from
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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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2006-10-06 18:47:49 +00:00
ae8cc3e18b
bug #8076 check option_debug before printing to debug channel.
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patch provided in bugnote, with minor changes.
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2006-10-03 15:53:07 +00:00
3cea4702a3
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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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2006-09-30 19:23:59 +00:00
6df7c274d8
Merged revisions 43798 via svnmerge from
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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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2006-09-27 19:12:40 +00:00
04cf782862
Small Cisco's RTP DTMF update
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2006-09-24 12:15:49 +00:00
1d50a8e881
Correct behavior on Cisco's DTMF
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2006-09-23 18:25:13 +00:00
2b55678e1f
Remove deprecated CLI apps from the core
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2006-09-21 21:17:39 +00:00
1c764935f2
SS7 marked the start of an open season for trunk again but here's something minor - abstract early bridging into the technology so that we don't always assume they use RTP and try it.
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2006-09-21 19:27:26 +00:00
70af28270d
Constify the result of a config retrieval function, to avoid mutilation (issue 7983).
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2006-09-20 20:40:39 +00:00
3c6d5053ba
Totally break a P2P bridge upon going on hold, and re-establish it upon going off hold.
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2006-09-20 17:08:44 +00:00
659d467720
Expand codec check so that raw formats must be equal for a Packet2Packet bridge to occur
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2006-09-20 16:55:09 +00:00
465adf2bf1
allow for packetization on rtp channel drivers, need to add
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option for setting our own packetization as apposed to just doing
what is asked.
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2006-09-18 23:32:57 +00:00
fcb999c01c
merge qwell's CLI verbification work
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2006-09-18 19:54:18 +00:00
d2a359e57f
Optimize a bit
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2006-09-09 19:12:52 +00:00
10e361763c
It's another round of RTP updates!
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2006-09-09 18:07:59 +00:00
f912c9e69f
Unbridge the RTP streams at the correct place
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2006-09-01 18:57:10 +00:00
0be2884d80
If we are doing video and we can't reinvite, then resort to generic bridging instead of Packet2Packet since video isn't supported there yet. (reported by PCadach in #asterisk-bugs)
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2006-09-01 17:54:22 +00:00
f2b836ff4f
Tweak the DTMF muting stuff a bit to take into account VLDTMF and compensation.
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2006-08-31 21:00:16 +00:00
29ee02bfce
Only write a received packet out if we are actually bridged to something
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2006-08-31 14:46:46 +00:00
c6977b9983
Merge in VLDTMF support with Zaptel/Core done by the ever great Darumkilla 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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2006-08-31 01:59:02 +00:00
Joshua Colp
Kevin P. Fleming
Russell Bryant
Olle Johansson
Tilghman Lesher
Matt O'Gorman
Paul Cadach