aboutsummaryrefslogtreecommitdiffstats
path: root/main/rtp.c
AgeCommit message (Collapse)AuthorFilesLines
2006-12-30Merged revisions 49066 via svnmerge from file1-0/+4
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: http://svn.digium.com/svn/asterisk/trunk@49067 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-27Merged revisions 49006 via svnmerge from kpfleming1-7/+7
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: http://svn.digium.com/svn/asterisk/trunk@49008 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-26Merged revisions 48964 via svnmerge from file1-6/+18
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: http://svn.digium.com/svn/asterisk/trunk@48965 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-15Merged revisions 48506 via svnmerge from file1-41/+55
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: http://svn.digium.com/svn/asterisk/trunk@48507 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-14Merged revisions 48472 via svnmerge from file1-11/+61
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: http://svn.digium.com/svn/asterisk/trunk@48473 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-14Merged revisions 48461 via svnmerge from file1-75/+1
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: http://svn.digium.com/svn/asterisk/trunk@48462 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-11Merged revisions 48381 via svnmerge from file1-36/+52
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: http://svn.digium.com/svn/asterisk/trunk@48383 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-06Staticize one, and Constify a bunch of usage strings for CLI commands.russell1-8/+8
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48303 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-05Doxygen updatesoej1-1/+28
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48277 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-05Well, yes... oej1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48259 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-05Reserving flags for coming code (currently in the "videocaps" branch) oej1-0/+2
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: http://svn.digium.com/svn/asterisk/trunk@48258 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-02- Disable RTP timeouts during T.38 transmissionoej1-0/+52
- 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: http://svn.digium.com/svn/asterisk/trunk@48200 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-30Merged revisions 48168 via svnmerge from file1-2/+8
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: http://svn.digium.com/svn/asterisk/trunk@48169 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-29Change logging for p2p rtp bridge modeoej1-10/+10
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48111 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-29Merged revisions 48107 via svnmerge from file1-0/+2
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: http://svn.digium.com/svn/asterisk/trunk@48108 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-25- Adding comment on suspicious memory allocation. Seems like it's never ↵oej1-6/+11
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: http://svn.digium.com/svn/asterisk/trunk@48003 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-22Merged revisions 47944 via svnmerge from file1-22/+2
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: http://svn.digium.com/svn/asterisk/trunk@47945 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-21Merged revisions 47897 via svnmerge from file1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@47898 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-20Use RTP/RTCP fds on the RTP structure, don't bother storing them.file1-20/+17
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@47854 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-20Merged revisions 47852 via svnmerge from file1-2/+3
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: http://svn.digium.com/svn/asterisk/trunk@47853 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-15Merged revisions 47645 via svnmerge from file1-2/+4
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: http://svn.digium.com/svn/asterisk/trunk@47646 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-15Merged revisions 47639 via svnmerge from file1-1/+2
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: http://svn.digium.com/svn/asterisk/trunk@47640 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-02Merged revisions 47053 via svnmerge from tilghman1-13/+13
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: http://svn.digium.com/svn/asterisk/trunk@47054 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-30In debug mode, recognize that someone is sending zrtp, even though weoej1-0/+8
can't do anything with it yet. Ideally a first step would be a passthrough mode. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@46439 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-29Bind RTCP to the same IP as RTP. oej1-1/+5
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: http://svn.digium.com/svn/asterisk/trunk@46409 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-26fix various spelling mistakes in comments (issue #8237, jmls)russell1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@46339 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-25Merged revisions 46154 via svnmerge from kpfleming1-0/+2
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: http://svn.digium.com/svn/asterisk/trunk@46155 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-18Merged revisions 45452 via svnmerge from file1-2/+2
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: http://svn.digium.com/svn/asterisk/trunk@45453 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-06Merged revisions 44628 via svnmerge from file1-8/+1
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: http://svn.digium.com/svn/asterisk/trunk@44630 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-06Merged revisions 44605 via svnmerge from file1-0/+4
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: http://svn.digium.com/svn/asterisk/trunk@44606 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-03bug #8076 check option_debug before printing to debug channel.mogorman1-12/+24
patch provided in bugnote, with minor changes. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@44253 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-30Merged revisions 44090 via svnmerge from pcadach1-3/+7
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: http://svn.digium.com/svn/asterisk/trunk@44091 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-27Merged revisions 43798 via svnmerge from file1-0/+6
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: http://svn.digium.com/svn/asterisk/trunk@43799 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-24Small Cisco's RTP DTMF updatepcadach1-12/+5
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43546 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-23Correct behavior on Cisco's DTMFpcadach1-6/+63
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43539 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-21Remove deprecated CLI apps from the coretilghman1-129/+7
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43449 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-21SS7 marked the start of an open season for trunk again but here's something ↵file1-35/+36
minor - abstract early bridging into the technology so that we don't always assume they use RTP and try it. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43437 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-20Constify the result of a config retrieval function, to avoid mutilation ↵tilghman1-1/+1
(issue 7983). git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43364 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-20Totally break a P2P bridge upon going on hold, and re-establish it upon ↵file1-2/+18
going off hold. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43343 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-20Expand codec check so that raw formats must be equal for a Packet2Packet ↵file1-7/+13
bridge to occur git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43340 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-18allow for packetization on rtp channel drivers, need to addmogorman1-89/+49
option for setting our own packetization as apposed to just doing what is asked. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43243 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-18merge qwell's CLI verbification workkpfleming1-56/+175
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43212 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-09Optimize a bitfile1-7/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@42583 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-09It's another round of RTP updates!file1-27/+38
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@42569 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-01Unbridge the RTP streams at the correct placefile1-8/+9
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@41735 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-01If we are doing video and we can't reinvite, then resort to generic bridging ↵file1-1/+7
instead of Packet2Packet since video isn't supported there yet. (reported by PCadach in #asterisk-bugs) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@41718 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-31Tweak the DTMF muting stuff a bit to take into account VLDTMF and compensation.file1-1/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@41632 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-31Only write a received packet out if we are actually bridged to somethingfile1-4/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@41574 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-31Merge in VLDTMF support with Zaptel/Core done by the ever great Darumkilla ↵file1-102/+167
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! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@41507 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-30Only feed a DTMF frame into the core if we need tofile1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@41338 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-09-22 11:04:17 +0000