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${ noResults }
69 Commits (21b53af31dcf5e881bf2e6617d7605653119c2f4)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Joshua Colp
|
dcdc6c0bc6 |
Change RTP protos list to be read/write. Most of the time it's only going to be read so making it use mutex locks was a waste.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51467 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
|
39d3580ee4 |
Make the RTP stack better conform to coding guidelines. (issue #8679 reported by johann8384)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51463 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
Russell Bryant
|
dcca8f345f |
Merged revisions 51311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines Merge the changes from the /team/group/vldtmf_fixup branch. The main bug being addressed here is a problem introduced when two SIP channels using SIP INFO dtmf have their media directly bridged. So, when a DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk would try to emulate a digit of some length by first sending a DTMF BEGIN frame and sending a DTMF END later timed off of incoming audio. However, since there was no audio coming in, the DTMF_END was never generated. This caused DTMF based features to no longer work. To fix this, the core now knows when a channel doesn't care about DTMF BEGIN frames (such as a SIP channel sending INFO dtmf). If this is the case, then Asterisk will not emulate a digit of some length, and will instead just pass through the single DTMF END event. Channel drivers also now get passed the length of the digit to their digit_end callback. This improves SIP INFO support even further by enabling us to put the real digit duration in the INFO message instead of a hard coded 250ms. Also, for an incoming INFO message, the duration is read from the frame and passed into the core instead of just getting ignored. (issue #8597, maybe others...) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51314 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
Luigi Rizzo
|
e7c5029d23 |
in the interest of portability, avoid using %zd when all
we need is to print is an integer that fits in 16 bits. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51310 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
|
461d49d2bd |
Merged revisions 51211 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51211 | file | 2007-01-17 19:18:44 -0500 (Wed, 17 Jan 2007) | 2 lines Pass data as well for hold/unhold/vidupdate frames. (issue #8840 reported by mdu113) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51212 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
|
3e6d6e0e62 |
Merged revisions 51182 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51182 | file | 2007-01-17 01:36:41 -0500 (Wed, 17 Jan 2007) | 2 lines Return the correct result when directly writing out a packet so that the core doesn't then decide to handle it the regular way again. (issue #8833 reported by rcourtna) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51183 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
Jason Parker
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9ca780a271 |
Merged revisions 51170 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51170 | qwell | 2007-01-16 18:20:56 -0600 (Tue, 16 Jan 2007) | 4 lines Fix issue with dtmf continuation packets when the dtmf digit is 0... Issue 8831 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51171 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
|
4942fd94d2 |
Merged revisions 50466 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r50466 | file | 2007-01-11 00:19:39 -0500 (Thu, 11 Jan 2007) | 2 lines Add support to see whether NAT was detected (yay symmetric RTP) and also add a check in chan_sip so that if NAT has been detected and the reinvite behind nat option has been turned off, then just do partial bridge. (issue #8655 reported by mnicholson) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@50467 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
|
ee137a5eaa |
Make callback declaration match one used in trunk.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@50056 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
|
91a7ca8df7 |
Merged revisions 50032 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r50032 | file | 2007-01-08 13:21:31 -0500 (Mon, 08 Jan 2007) | 2 lines Disable the more intense packet2packet bridging until the bugs can be worked out. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@50033 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
Olle Johansson
|
68ff3c3575 |
Issue #8663 - Add passthrough support for MPEG4 (neutrino88).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49968 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
|
e2a50de88f |
Clarify why we are reading in a frame in the Packet2Packet bridge.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49072 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
|
c6c83cf01e |
Merged revisions 49066 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49067 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
Kevin P. Fleming
|
adca0ff14b |
Merged revisions 49006 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49008 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
|
7f61b822c1 |
Merged revisions 48964 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48965 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
|
915647d267 |
Merged revisions 48506 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48507 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
|
f6649ae0af |
Merged revisions 48472 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48473 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
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1c4c365377 |
Merged revisions 48461 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48462 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
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c3052f7a7e |
Merged revisions 48381 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48383 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Russell Bryant
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17a2888d2e |
Staticize one, and Constify a bunch of usage strings for CLI commands.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48303 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Olle Johansson
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fe53552f41 |
Doxygen updates
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48277 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Olle Johansson
|
00bf07b12e |
Well, yes...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48259 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Olle Johansson
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b8fcae6d75 |
Reserving flags for coming code (currently in the "videocaps" branch)
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 |
19 years ago |
|
Olle Johansson
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c23bc46089 |
- Disable RTP timeouts during T.38 transmission
- 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. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48200 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
|
869101028b |
Merged revisions 48168 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48169 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Olle Johansson
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2bee4aba4d |
Change logging for p2p rtp bridge mode
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48111 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
|
d44b349211 |
Merged revisions 48107 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ 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 ........ 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) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48108 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Olle Johansson
|
7991366506 |
- Adding comment on suspicious memory allocation. Seems like it's never 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. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48003 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
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b50fc7a502 |
Merged revisions 47944 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47945 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
|
a69ac09748 |
Merged revisions 47897 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47898 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
|
03a7adf8ce |
Use RTP/RTCP fds on the RTP structure, don't bother storing them.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47854 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
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b2b966eda8 |
Merged revisions 47852 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47853 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
|
993c6823e6 |
Merged revisions 47645 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47646 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
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5861048fb6 |
Merged revisions 47639 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47640 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
Tilghman Lesher
|
79f75ec09a |
Merged revisions 47053 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47054 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Olle Johansson
|
2cb07fbaa4 |
In debug mode, recognize that someone is sending zrtp, even though we
can't do anything with it yet. Ideally a first step would be a passthrough mode. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46439 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Olle Johansson
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52a5d63a2d |
Bind RTCP to the same IP as RTP.
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. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46409 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Russell Bryant
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0ca6a42d7e |
fix various spelling mistakes in comments (issue #8237, jmls)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46339 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Kevin P. Fleming
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88efcea05e |
Merged revisions 46154 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46155 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
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bb8926d50c |
Merged revisions 45452 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@45453 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
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62e6417b21 |
Merged revisions 44628 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44630 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
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85625f3505 |
Merged revisions 44605 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44606 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
Matt O'Gorman
|
ae8cc3e18b |
bug #8076 check option_debug before printing to debug channel.
patch provided in bugnote, with minor changes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44253 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
Paul Cadach
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3cea4702a3 |
Merged revisions 44090 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44090 | pcadach | 2006-10-01 01:20:38 +0600 (Вск, 01 Окт 2006) | 1 line Allow one-way RTP streams (device->Asterisk) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44091 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Joshua Colp
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6df7c274d8 |
Merged revisions 43798 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43799 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Paul Cadach
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04cf782862 |
Small Cisco's RTP DTMF update
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43546 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Paul Cadach
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1d50a8e881 |
Correct behavior on Cisco's DTMF
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43539 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Tilghman Lesher
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2b55678e1f |
Remove deprecated CLI apps from the core
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43449 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Joshua Colp
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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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43437 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Tilghman Lesher
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70af28270d |
Constify the result of a config retrieval function, to avoid mutilation (issue 7983).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43364 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |