Commit graph

207 commits

Author SHA1 Message Date
Russell Bryant
5590f67f58 Convert so more logging to ast_debug (issue #10045, dimas)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@71557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-25 13:42:51 +00:00
Russell Bryant
80166c6de8 Conversions to ast_debug()
(issue #9984, patches from eliel and dimas)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@71338 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-24 18:51:41 +00:00
Joshua Colp
6f98665672 Behold the magic of casting!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@71146 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-22 16:14:00 +00:00
Steve Murphy
6a2af3c983 Merged revisions 71063 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r71063 | murf | 2007-06-22 08:10:24 -0600 (Fri, 22 Jun 2007) | 1 line

My conditions for merging amaflags info was naive; DOCUMENTATION is the default, although null is possible; theft of user-settable fields is not good. Just copy them, leave them alone.
This is for bug 10016. (plus a small fix to rtp, to elim a compiler warning (dev mode))
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@71093 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-22 15:15:35 +00:00
Jason Parker
7a1c2d94bb Add manager events for RTCP statistics.
Also adds a new "reporting" permission for manager, since it can be incredibly spammy.
  This permission was discussed on the -dev mailing list some months back.

Issue 8613, patch by johann8384, with some minor changes by me.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@70961 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-21 23:07:20 +00:00
Joshua Colp
80cdeaef55 Merged revisions 70727 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r70727 | file | 2007-06-21 11:22:39 -0400 (Thu, 21 Jun 2007) | 2 lines

Do not Packet2Packet bridge if packetization settings do not allow it. (issue #9117 reported by phsultan)

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2007-06-21 15:25:13 +00:00
Joshua Colp
a2b3357a9d Merged revisions 70360 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r70360 | file | 2007-06-20 13:52:57 -0400 (Wed, 20 Jun 2007) | 2 lines

Put the speex packetization values back in but disable it when setting up the smoother.

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2007-06-20 17:55:09 +00:00
Joshua Colp
9bec4f4b58 Merged revisions 70003 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r70003 | file | 2007-06-19 13:07:40 -0400 (Tue, 19 Jun 2007) | 10 lines

Merged revisions 69992 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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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)

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2007-06-19 17:09:20 +00:00
Joshua Colp
d0eaf1e389 Merged revisions 68922 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r68922 | file | 2007-06-12 10:23:11 -0400 (Tue, 12 Jun 2007) | 10 lines

Merged revisions 68921 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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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.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@68923 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-12 14:26:12 +00:00
Russell Bryant
1d57ccb6f7 Fix a bunch of doxygen errors and document more things
(issue #9842, snuffy)


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2007-06-07 23:07:25 +00:00
Tilghman Lesher
9d05ff8ed5 Issue 9869 - replace malloc and memset with ast_calloc, and other coding guidelines changes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67864 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-06 21:20:11 +00:00
Joshua Colp
40df1fb464 Merged revisions 67650 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r67650 | file | 2007-06-06 09:30:25 -0400 (Wed, 06 Jun 2007) | 10 lines

Merged revisions 67649 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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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)

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2007-06-06 13:32:11 +00:00
Russell Bryant
b5089b4a58 Merged revisions 67071 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r67071 | russell | 2007-06-04 16:47:36 -0500 (Mon, 04 Jun 2007) | 2 lines

Add a missing \n.  (pointed out by jcmoore on IRC)

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2007-06-04 21:48:15 +00:00
Joshua Colp
ed4726769a Merged revisions 66437 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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)

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2007-05-29 16:46:49 +00:00
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
2007-05-24 18:07:56 +00:00
Joshua Colp
e4191c375f Merged revisions 65863 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r65863 | file | 2007-05-24 11:08:17 -0400 (Thu, 24 May 2007) | 2 lines

I like it when the RTP stack compiles myself...

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65869 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-24 15:10:13 +00:00
Russell Bryant
89b0e6049a Merged revisions 65842 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65843 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-24 14:50:25 +00:00
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
2007-04-30 16:16:26 +00:00
Jason Parker
82d5673c81 Merged revisions 61707 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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.

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2007-04-20 21:37:04 +00:00
Russell Bryant
c21f118a65 Merged revisions 61697 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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.

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2007-04-20 20:43:05 +00:00
Olle Johansson
16a080781d Merged revisions 61676 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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!

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2007-04-18 20:48:13 +00:00
Olle Johansson
c4cd1b6761 Merged revisions 61674 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r61674 | oej | 2007-04-18 22:28:53 +0200 (Wed, 18 Apr 2007) | 2 lines

Issue #9554 - Improve RTCP (Dave Troy)

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2007-04-18 20:39:31 +00:00
Russell Bryant
c4f42601d6 Merged revisions 59358 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

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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)

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2007-03-29 17:20:43 +00:00
Russell Bryant
08e3a9bdc8 Merged revisions 59207 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59208 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-26 17:51:27 +00:00
Joshua Colp
ddca41798b Merged revisions 58783 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58784 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-12 01:22:29 +00:00
Joshua Colp
2ab6ed30cd Merged revisions 58436 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58437 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-08 18:05:54 +00:00
Joshua Colp
e7da006562 Merged revisions 58240 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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)

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2007-03-07 17:55:11 +00:00
Joshua Colp
aabe0abaee Merged revisions 57768 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-05 03:24:18 +00:00
Olle Johansson
75d387acbc Doxygen additions, corrections
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56665 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-24 20:29:41 +00:00
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.


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2007-02-16 13:35:44 +00:00
Joshua Colp
8f6d9918a7 Merged revisions 53434 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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)

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2007-02-07 17:57:37 +00:00
Russell Bryant
dfb5ef7f55 Merged revisions 53429 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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)

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2007-02-07 17:46:42 +00:00
Joshua Colp
2cc011e005 Merged revisions 53120 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r53120 | file | 2007-02-02 11:15:22 -0600 (Fri, 02 Feb 2007) | 2 lines

Correct a copy/pasted error message line for RTCP.

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2007-02-02 17:16:05 +00:00
Joshua Colp
493126cf0c Merged revisions 53052 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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.

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2007-02-01 00:24:50 +00:00
Joshua Colp
fa66a0bf03 Merged revisions 53050 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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.

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2007-02-01 00:23:19 +00:00
Russell Bryant
7ca426c5b4 Merged revisions 53040 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

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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)

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2007-01-31 17:45:43 +00:00
Russell Bryant
2d0e8864aa Merged revisions 52645 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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)

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2007-01-29 21:27:34 +00:00
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
2007-01-29 21:03:07 +00:00
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
2007-01-22 23:53:16 +00:00
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
2007-01-22 23:51:42 +00:00
Russell Bryant
dcca8f345f Merged revisions 51311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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...)

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2007-01-19 18:06:03 +00:00
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.



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2007-01-19 17:48:48 +00:00
Joshua Colp
461d49d2bd Merged revisions 51211 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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)

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2007-01-18 00:20:50 +00:00
Joshua Colp
3e6d6e0e62 Merged revisions 51182 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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)

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2007-01-17 06:37:47 +00:00
Jason Parker
9ca780a271 Merged revisions 51170 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

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2007-01-17 00:22:20 +00:00
Joshua Colp
4942fd94d2 Merged revisions 50466 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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)

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2007-01-11 05:21:03 +00:00
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
2007-01-08 20:10:23 +00:00
Joshua Colp
91a7ca8df7 Merged revisions 50032 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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.

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2007-01-08 18:23:39 +00:00
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
2007-01-08 11:49:23 +00:00
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
2006-12-30 18:27:13 +00:00
Joshua Colp
c6c83cf01e Merged revisions 49066 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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
Kevin P. Fleming
adca0ff14b Merged revisions 49006 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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
Joshua Colp
7f61b822c1 Merged revisions 48964 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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
Joshua Colp
915647d267 Merged revisions 48506 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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
Joshua Colp
f6649ae0af Merged revisions 48472 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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
Joshua Colp
1c4c365377 Merged revisions 48461 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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
Joshua Colp
c3052f7a7e Merged revisions 48381 via svnmerge from
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
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
2006-12-06 07:28:56 +00:00
Olle Johansson
fe53552f41 Doxygen updates
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48277 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-05 20:39:13 +00:00
Olle Johansson
00bf07b12e Well, yes...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48259 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-05 11:09:23 +00:00
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
2006-12-05 10:52:53 +00:00
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.



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2006-12-02 12:05:40 +00:00
Joshua Colp
869101028b Merged revisions 48168 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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
Olle Johansson
2bee4aba4d Change logging for p2p rtp bridge mode
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48111 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-11-29 19:44:06 +00:00
Joshua Colp
d44b349211 Merged revisions 48107 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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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
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.



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2006-11-25 09:45:57 +00:00
Joshua Colp
b50fc7a502 Merged revisions 47944 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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
Joshua Colp
a69ac09748 Merged revisions 47897 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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
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
2006-11-20 16:06:10 +00:00
Joshua Colp
b2b966eda8 Merged revisions 47852 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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
Joshua Colp
993c6823e6 Merged revisions 47645 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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
Joshua Colp
5861048fb6 Merged revisions 47639 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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
Tilghman Lesher
79f75ec09a Merged revisions 47053 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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
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
2006-10-30 16:59:02 +00:00
Olle Johansson
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.



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2006-10-29 20:21:33 +00:00
Russell Bryant
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
2006-10-26 17:52:15 +00:00
Kevin P. Fleming
88efcea05e Merged revisions 46154 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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
Joshua Colp
bb8926d50c Merged revisions 45452 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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
Joshua Colp
62e6417b21 Merged revisions 44628 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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
Joshua Colp
85625f3505 Merged revisions 44605 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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
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
2006-10-03 15:53:07 +00:00
Paul Cadach
3cea4702a3 Merged revisions 44090 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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
Joshua Colp
6df7c274d8 Merged revisions 43798 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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
Paul Cadach
04cf782862 Small Cisco's RTP DTMF update
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43546 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-09-24 12:15:49 +00:00
Paul Cadach
1d50a8e881 Correct behavior on Cisco's DTMF
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2006-09-23 18:25:13 +00:00
Tilghman Lesher
2b55678e1f Remove deprecated CLI apps from the core
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43449 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-09-21 21:17:39 +00:00
Joshua Colp
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
2006-09-21 19:27:26 +00:00
Tilghman Lesher
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
2006-09-20 20:40:39 +00:00
Joshua Colp
3c6d5053ba Totally break a P2P bridge upon going on hold, and re-establish it upon going off hold.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43343 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-09-20 17:08:44 +00:00
Joshua Colp
659d467720 Expand codec check so that raw formats must be equal for a Packet2Packet bridge to occur
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43340 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-09-20 16:55:09 +00:00
Matt O'Gorman
465adf2bf1 allow for packetization on rtp channel drivers, need to add
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
Kevin P. Fleming
fcb999c01c merge qwell's CLI verbification work
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2006-09-18 19:54:18 +00:00
Joshua Colp
d2a359e57f Optimize a bit
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2006-09-09 19:12:52 +00:00
Joshua Colp
10e361763c It's another round of RTP updates!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@42569 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-09-09 18:07:59 +00:00
Joshua Colp
f912c9e69f Unbridge the RTP streams at the correct place
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2006-09-01 18:57:10 +00:00
Joshua Colp
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41718 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-09-01 17:54:22 +00:00
Joshua Colp
f2b836ff4f Tweak the DTMF muting stuff a bit to take into account VLDTMF and compensation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41632 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-31 21:00:16 +00:00
Joshua Colp
29ee02bfce Only write a received packet out if we are actually bridged to something
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41574 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-31 14:46:46 +00:00
Joshua Colp
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!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41507 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-31 01:59:02 +00:00
Joshua Colp
0855df6a5a Only feed a DTMF frame into the core if we need to
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41338 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-30 14:57:06 +00:00
Joshua Colp
245aa1a62d Only switch the second alert fd (which is RTCP) to callback mode if it is in use
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41317 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-30 03:18:04 +00:00
Joshua Colp
12b6ec4e11 Use an API call (ast_rtp_get_bridged) to return the RTP stream we are bridged to, and also use it in chan_sip so we know to ignore the no RTP activity checking
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-30 03:16:03 +00:00
Joshua Colp
da7d969ae1 If the RTP stack is already being operated in callback mode, then suspend it upon switching to P2P callback bridging. Once P2P callback bridging has ended, then restore callback mode.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41303 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-30 01:29:59 +00:00
Joshua Colp
ca33d2ecc6 This is the last round of RTP bridge optimizations. Basically it introduces a way that under a straight bridge (ie: no transcoding and no DTMF detection) the core is not touched at all and no frames pass through (not even null frames). This is accomplished by stealing the file descriptors from the channel and using the provided IO context with a custom callback. When a channel is placed on hold this bridge is broken so audio can flow from the core to the other side. When a channel is off hold this bridge is re-established.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41302 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-30 01:22:46 +00:00
Joshua Colp
a2e2a51d8d Move the direct bridge write to after the NAT handling code
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41285 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-29 23:41:16 +00:00
Joshua Colp
c70ed7614a Merge in RTP-level packet bridging. Packet comes in, packet goes out - that's what RTP-level packet bridging is all about!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41235 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-28 17:37:56 +00:00
Kevin P. Fleming
0a27d8bfe5 merge new_loader_completion branch, including (at least):
- restructured build tree and makefiles to eliminate recursion problems
  - support for embedded modules
  - support for static builds
  - simpler cross-compilation support
  - simpler module/loader interface (no exported symbols)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@40722 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-21 02:11:39 +00:00
Renamed from rtp.c (Browse further)