94 Commits (7173f3378fb67babc4991c30aaa19ade43c8f9bc)

Author	SHA1	Message	Date
Joshua Colp	ed4726769a	Merged revisions 66437 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r66437 | file | 2007-05-29 12:44:34 -0400 (Tue, 29 May 2007) | 2 lines Handle cases where a frame may have no data. (issue #9519 reported by dmb) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66438 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	bcd2bd8294	Make this build on *my* machine again, and hopefully not break others. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65964 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	e4191c375f	Merged revisions 65863 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r65863 | file | 2007-05-24 11:08:17 -0400 (Thu, 24 May 2007) | 2 lines I like it when the RTP stack compiles myself... ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65869 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	89b0e6049a	Merged revisions 65842 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r65842 | russell | 2007-05-24 09:49:05 -0500 (Thu, 24 May 2007) | 5 lines Fix the calculation of the RTT for RTCP. The previous code would result in oscillating and incorrect data. Additionally, the RTT would sometimes report negative values due to incorrect calculations. (issue #9601, patch from davetroy) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65843 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	b419fc1134	Add support for setting the CoS for VLAN traffic (802.1p) in Linux. The ... file doc/qos.tex has been updated to document the new functionality. (issue #9540, patch submitted by IgorG) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62457 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	82d5673c81	Merged revisions 61707 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61707 | qwell | 2007-04-20 16:35:27 -0500 (Fri, 20 Apr 2007) | 8 lines Avoid invalid seqno cycling detection. Per comment from Dave Troy: This adds back in some simple typecasting I had in an earlier version which I realize now may be breaking things. Issue #9554. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61708 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	c21f118a65	Merged revisions 61697 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61697 | russell | 2007-04-20 15:42:02 -0500 (Fri, 20 Apr 2007) | 2 lines Remove a stray debug message introduced by a recent commit. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61698 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	16a080781d	Merged revisions 61676 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61676 | oej | 2007-04-18 22:46:23 +0200 (Wed, 18 Apr 2007) | 2 lines Clean upp formatting, add some doxygen stuff while we're in cleaning mode... Thanks Kevin! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61677 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	c4cd1b6761	Merged revisions 61674 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61674 | oej | 2007-04-18 22:28:53 +0200 (Wed, 18 Apr 2007) | 2 lines Issue #9554 - Improve RTCP (Dave Troy) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61675 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	c4f42601d6	Merged revisions 59358 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r59358 | russell | 2007-03-29 12:17:41 -0500 (Thu, 29 Mar 2007) | 13 lines Merged revisions 59357 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59357 | russell | 2007-03-29 12:14:33 -0500 (Thu, 29 Mar 2007) | 5 lines If an error occurs when reading from an RTP socket, and the error code does not indicate that we should try again, then return NULL instead of a "null frame". This will prevent Asterisk from trying over and over again, and eventually causing the system to crash. (issue #8285, john) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59359 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	08e3a9bdc8	Merged revisions 59207 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59207 | russell | 2007-03-26 12:45:55 -0500 (Mon, 26 Mar 2007) | 7 lines The AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in some because they get set in sip_hangup. So, there are common situations where the variables will not be available in the dialplan at all. So, this patch provides an alternate method for getting to this information by introducing AUDIORTPQOS and VIDEORTPQOS dialplan functions. (issue #9370, patch by Corydon76, with some testing by blitzrage) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59208 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	ddca41798b	Merged revisions 58783 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58783 | file | 2007-03-11 21:21:12 -0400 (Sun, 11 Mar 2007) | 2 lines Allow RFC2833 compensation to compensate for even stupider implementations by queueing up the end frame at the start, not the actual end. (issue #8963 reported by AndrewZ) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58784 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	2ab6ed30cd	Merged revisions 58436 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58436 | file | 2007-03-08 13:01:00 -0500 (Thu, 08 Mar 2007) | 2 lines Make early SDP seeding even smarter! We have to check codecs in the make_compatible function too. (issue #9221 reported by marcelbarbulescu) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58437 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	e7da006562	Merged revisions 58240 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58240 | file | 2007-03-07 12:52:58 -0500 (Wed, 07 Mar 2007) | 2 lines Ensure we have (or should have) at least one matching codec before attempting early bridge SDP seeding. (issue #9221 reported by marcelbarbulescu) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58241 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	aabe0abaee	Merged revisions 57768 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57768 | file | 2007-03-04 22:22:17 -0500 (Sun, 04 Mar 2007) | 2 lines Preserve marker bit when P2P bridging. (issue #9198 reported by edgreenberg) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57769 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	75d387acbc	Doxygen additions, corrections ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56665 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	ba32ee49d0	Adding Realtime Text support (T.140) to Asterisk ... T.140/RFC 2793 is a live communication channel, originally created for IP based text phones for hearing impaired. Feels very much like the old Unix talk application. This code is developed and disclaimed by John Martin of Aupix, UK. Tested for interoperability by myself and Omnitor in Sweden, the company that wrote most of the specifications. A big thank you to everyone involved in this. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54838 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	8f6d9918a7	Merged revisions 53434 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53434 | file | 2007-02-07 12:53:03 -0500 (Wed, 07 Feb 2007) | 2 lines We can not reliably do P2P bridging with DTMF passing back with compensation if we need to listen for DTMF frames. (issue #8962 reported by caio1982) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53435 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	dfb5ef7f55	Merged revisions 53429 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53429 | russell | 2007-02-07 11:39:31 -0600 (Wed, 07 Feb 2007) | 7 lines When parsing the NTP timestamp in a sender report message, you are supposed to take the low 16 bits of the integer part, and the high 16 bits of the fractional part. However, the code here was erroneously taking the low 16 bits of the fractional part. It then shifted the result 16 bits down, so the result was always zero. This fix makes it grab the appropriate high 16 bits, instead. (issue #8991, pointed out by andre_abrantes) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53431 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	2cc011e005	Merged revisions 53120 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53120 | file | 2007-02-02 11:15:22 -0600 (Fri, 02 Feb 2007) | 2 lines Correct a copy/pasted error message line for RTCP. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53122 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	493126cf0c	Merged revisions 53052 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53052 | file | 2007-01-31 18:24:20 -0600 (Wed, 31 Jan 2007) | 2 lines When going on hold have the side that was put on hold reinvite back to Asterisk. When going off hold have the side that was taken off hold reinvited back to the other party. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53053 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	fa66a0bf03	Merged revisions 53050 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53050 | file | 2007-01-31 18:19:48 -0600 (Wed, 31 Jan 2007) | 2 lines Add more frame types to forward in the RTP bridge loops. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53051 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	7ca426c5b4	Merged revisions 53040 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r53040 | russell | 2007-01-31 11:45:05 -0600 (Wed, 31 Jan 2007) | 11 lines Merged revisions 53039 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r53039 | russell | 2007-01-31 11:41:51 -0600 (Wed, 31 Jan 2007) | 3 lines Use the proper format string to print unsigned values in the rtp debug output. (issue #8954, wmis) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53041 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	2d0e8864aa	Merged revisions 52645 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52645 | russell | 2007-01-29 15:26:27 -0600 (Mon, 29 Jan 2007) | 6 lines Fix a problem with packet-to-packet bridging and DTMF mode translation. P2P bridging can only be used when the DTMF modes don't match if the core is monitoring DTMF in both directions. Then, the core will handle the translation. Otherwise, this bridging method can not be used. (issue #8936) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@52646 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	a1d764c00a	Only use locking for bridge information if intense P2P bridging is enabled. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@52635 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
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	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	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	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	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	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	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	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