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93 Commits

Author SHA1 Message Date
Joshua Colp
0f3e461074 Bring both DTMF begin and end frames up through to the core for DTMF feature handling. 
(closes issue #10826)
Reported by: dimas


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@85559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-10-15 16:22:02 +00:00
Joshua Colp
3cc997694d If Monitor or a spy was added to a P2P or native bridged channel bring the channel back to the generic bridging core so the monitor or spy operations work. 
(closes issue #10943)
Reported by: julianjm


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@85552 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-10-15 14:55:04 +00:00
Joshua Colp
ba0bb743e6 Only update codec information if the channel has a technology private structure. 
(issue #10915)
Reported by: ramonpeek


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@85057 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-10-08 20:06:33 +00:00
Joshua Colp
3dcf938f73 Update codec information as well as address when doing hold reinvites. 
(issue #10868)
Reported by: mavince


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@85023 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-10-08 15:37:46 +00:00
Joshua Colp
35ed1f7148 Update the remembered RTP peer information when putting an endpoint on hold or taking it off hold so that the RTP stack does not initiate a needless reinvite. 
(closes issue #10868)
Reported by: mavince


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@84818 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-10-05 18:55:36 +00:00
Tilghman Lesher
1e9edf1338 When an RFC 2833 event is sent that we don't recognize, ignore it, don't queue a NULL digit (closes issue #10877) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@84581 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-10-03 22:59:17 +00:00
Russell Bryant
d6b8fb4dc0 gcc 4.2 has a new set of warnings dealing with cosnt pointers. This set of 
changes gets all of Asterisk (minus chan_alsa for now) to compile with gcc 4.2.
(closes issue #10774, patch from qwell)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@83432 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-09-21 14:37:20 +00:00
Joshua Colp
968be2def2 (closes issue #10562) 
Reported by: idkpmiller
Correct jitter value output in the CLI to be as expected.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@80974 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-08-27 13:20:31 +00:00
Joshua Colp
d13b192018 (closes issue #10526) 
Reported by: sinistermidget
Revert commit from issue #10355 and return timestamp skew to 640. 


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@80255 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-08-22 16:14:38 +00:00
Joshua Colp
6dbbfcdc25 (closes issue #10440) 
Reported by: irroot
(closes issue #10454)
Reported by: flo_turc
Increase maximum timestamp skew to 120. 20 was apparently far too low.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@79553 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-08-15 14:40:23 +00:00
Joshua Colp
88b982d7ff (closes issue #10355) 
Reported by: wdecarne
Now that we pass through RTP timestamp information we need to make the allowed timestamp skew considerably less. There are situations where a source may change and due to the timestamp difference the receiver will experience an audio gap since we did not indicate by setting the marker bit that the source changed.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@78172 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-08-06 15:27:24 +00:00
Luigi Rizzo
19ec0f8fe1 set the sequence number in a frame for all frame types 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@77022 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-25 09:34:01 +00:00
Russell Bryant
c83a0b2a3c cast arguments to ast_log so that it builds without warnings for me 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@75447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-17 20:51:25 +00:00
Joshua Colp
396e723f17 Ensure that the pointer to STUN data does not go to unaccessible memory. (ASA-2007-017) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@75439 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-07-17 20:40:57 +00:00
Russell Bryant
d528f7791d Only output debug information related to RTCP timestamps when RTCP debug 
is turned on (issue #10066, patch by me)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@72112 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-27 16:34:24 +00:00
Jason Parker
8a7cb1ec48 Don't dereference a pointer that may be NULL here. 
Issue 10017.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@71915 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-26 20:36:09 +00:00
Joshua Colp
164b3bca49 Do not Packet2Packet bridge if packetization settings do not allow it. (issue #9117 reported by phsultan) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@70727 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-21 15:22:39 +00:00
Joshua Colp
00a80bcb7d Put the speex packetization values back in but disable it when setting up the smoother. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@70360 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-20 17:52:57 +00:00
Joshua Colp
8b71bbd6ed Merged revisions 69992 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r69992 | file | 2007-06-19 13:00:58 -0400 (Tue, 19 Jun 2007) | 2 lines

Handle the CC field in the RTP header. (issue #9384 reported by DoodleHu)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@70003 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-19 17:07:40 +00:00
Joshua Colp
8c01036a9f Merged revisions 68921 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r68921 | file | 2007-06-12 10:18:57 -0400 (Tue, 12 Jun 2007) | 2 lines

Bring RTP back to Asterisk at the end of a native bridge no matter what.

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@68922 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-12 14:23:11 +00:00
Joshua Colp
1f6872a45e Merged revisions 67649 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r67649 | file | 2007-06-06 09:28:34 -0400 (Wed, 06 Jun 2007) | 2 lines

Reinvite the RTP back to the Asterisk machine when the timeout happens. (issue #9888 reported by gasparz)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67650 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-06 13:30:25 +00:00
Russell Bryant
9074935852 Add a missing \n. (pointed out by jcmoore on IRC) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67071 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-06-04 21:47:36 +00:00
Joshua Colp
13afc81f91 Handle cases where a frame may have no data. (issue #9519 reported by dmb) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66437 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-29 16:44:34 +00:00
Russell Bryant
a2a73cde51 Make 1.4 build on my machine, too.. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66068 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-24 22:06:13 +00:00
Kevin P. Fleming
521b241738 don't use uninitialized variables 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65965 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-24 18:24:55 +00:00
Joshua Colp
9f88eccda6 I like it when the RTP stack compiles myself... 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65863 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-24 15:08:17 +00:00
Russell Bryant
f32662deb4 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/branches/1.4@65842 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-05-24 14:49:05 +00:00
Jason Parker
9378347264 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/branches/1.4@61707 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-04-20 21:35:27 +00:00
Russell Bryant
477ff09bdf Remove a stray debug message introduced by a recent commit. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@61697 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-04-20 20:42:02 +00:00
Olle Johansson
521e1698e1 Clean upp formatting, add some doxygen stuff while we're in cleaning mode... Thanks Kevin! 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@61676 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-04-18 20:46:23 +00:00
Olle Johansson
0ef4dad39c Issue #9554 - Improve RTCP (Dave Troy) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@61674 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-04-18 20:28:53 +00:00
Russell Bryant
6cdc236f09 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/branches/1.4@59358 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-03-29 17:17:41 +00:00
Russell Bryant
fa97f6c381 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/branches/1.4@59207 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-03-26 17:45:55 +00:00
Joshua Colp
dd1a0aa1dd 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/branches/1.4@58783 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-03-12 01:21:12 +00:00
Joshua Colp
2b3c0dc1e4 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/branches/1.4@58436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-03-08 18:01:00 +00:00
Joshua Colp
240355d78b 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/branches/1.4@58240 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-03-07 17:52:58 +00:00
Joshua Colp
44dd29ff61 Preserve marker bit when P2P bridging. (issue #9198 reported by edgreenberg) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@57768 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-03-05 03:22:17 +00:00
Joshua Colp
204b58a23c 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/branches/1.4@53434 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-07 17:53:03 +00:00
Russell Bryant
2b5990387c 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/branches/1.4@53429 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-07 17:39:31 +00:00
Joshua Colp
f0329105f3 Correct a copy/pasted error message line for RTCP. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-02 17:15:22 +00:00
Joshua Colp
28dacda047 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/branches/1.4@53052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-01 00:24:20 +00:00
Joshua Colp
7229c2ba48 Add more frame types to forward in the RTP bridge loops. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-01 00:19:48 +00:00
Russell Bryant
0bc7ef3b06 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/branches/1.4@53040 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-31 17:45:05 +00:00
Russell Bryant
06a3110ed8 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/branches/1.4@52645 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-29 21:26:27 +00:00
Russell Bryant
33235b40d6 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/branches/1.4@51311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-19 17:49:38 +00:00
Joshua Colp
9461aa0027 Pass data as well for hold/unhold/vidupdate frames. (issue #8840 reported by mdu113) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51211 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-18 00:18:44 +00:00
Joshua Colp
9f0ad6093e 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/branches/1.4@51182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-17 06:36:41 +00:00
Jason Parker
1825fc3ee2 Fix issue with dtmf continuation packets when the dtmf digit is 0... 
Issue 8831


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51170 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-17 00:20:56 +00:00
Joshua Colp
240ca25bea 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/branches/1.4@50466 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-11 05:19:39 +00:00
Joshua Colp
46700b62af Disable the more intense packet2packet bridging until the bugs can be worked out. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@50032 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-08 18:21:31 +00:00