Commit Graph

12 Commits

Author SHA1 Message Date
Kinsey Moore c3b1d8f8c1 res_rtp_multicast: Ensure SSRC is set properly
This fixes a bug where the SSRC field on multicast RTP can be stuck at
0 which can cause problems for endpoints trying to make sense of
incoming streams.

(closes issue ASTERISK-22567)
Reported by: Simone Camporeale
Patches:
    22567_res_mulitcast_ssrc.patch uploaded by Simone Camporeale (License 6536)
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Merged revisions 400393 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 400394 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 400395 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400397 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-03 18:32:59 +00:00
Matthew Jordan 2dd40e9707 Calculate the timestamp for outbound RTP if we don't have timing information
This patch calculates the timestamp for outbound RTP when we don't have timing
information. This uses the same approach in res_rtp_asterisk. Thanks to both
Pietro and Tzafrir for providing patches.

(closes issue ASTERISK-19883)
Reported by: Giacomo Trovato
Tested by: Pietro Bertera, Tzafrir Cohen
patches:
  rtp-timestamp-1.8.patch uploaded by tzafrir (License 5035)
  rtp-timestamp.patch uploaded by pbertera (License 5943)
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Merged revisions 385636 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 385637 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385638 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-04-14 03:01:33 +00:00
Joshua Colp cdcbffeed0 Fix an issue where a caller to ast_write on a MulticastRTP channel would determine it failed when in reality it did not.
When sending RTP packets via multicast the amount of data sent is stored in a variable and returned
from the write function. This is incorrect as any non-zero value returned is considered a failure while
a return value of 0 is success. For callers (such as ast_streamfile) that checked the return value
they would have considered it a failure when in reality nothing went wrong and it was actually a success.

The write function for the multicast RTP engine now returns -1 on failure and 0 on success, as it should.

(closes issue ASTERISK-17254)
Reported by: wybecom
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Merged revisions 373550 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 373551 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 373552 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373553 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-25 12:12:20 +00:00
Jonathan Rose e5ac65bb43 Fix sequence number overflow over 16 bits causing codec change in RTP packets.
Sequence number was handled as an unsigned integer (usually 32 bits I think, more
depending on the architecture) and was put into the rtp packet which is basically
just a bunch of bits using an or operation. Sequence number only has 16 bits
allocated to it in an RTP packet anyway, so it would add to the next field which
just happened to be the codec. This makes sure the sequence number is set to be
a 16 bit integer regardless of architecture (hopefully) and also makes it so the
incrementing of the sequence number does bitwise or at the peak of a 16 bit number
so that the value will be set back to 0 when going beyond 65535 anyway.

(closes issue ASTERISK-18291)
Reported by: Will Schick
Review: https://reviewboard.asterisk.org/r/1542/
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Merged revisions 342602 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 342603 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342604 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-27 19:48:23 +00:00
Leif Madsen a525edea59 Merged revisions 328247 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.10

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  r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines
  
  Merged revisions 328209 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
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    r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines
    
    Introduce <support_level> tags in MODULEINFO.
    This change introduces MODULEINFO into many modules in Asterisk in order to show
    the community support level for those modules. This is used by changes committed
    to menuselect by Russell Bryant recently (r917 in menuselect). More information about
    the support level types and what they mean is available on the wiki at
    https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328259 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-14 20:28:54 +00:00
David Vossel c26c190711 Asterisk media architecture conversion - no more format bitfields
This patch is the foundation of an entire new way of looking at media in Asterisk.
The code present in this patch is everything required to complete phase1 of my
Media Architecture proposal.  For more information about this project visit the link below.
https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal

The primary function of this patch is to convert all the usages of format
bitfields in Asterisk to use the new format and format_cap APIs.  Functionally
no change in behavior should be present in this patch.  Thanks to twilson
and russell for all the time they spent reviewing these changes.

Review: https://reviewboard.asterisk.org/r/1083/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-03 16:22:10 +00:00
Russell Bryant cc0b7e7df5 Some scheduler API cleanup and improvements.
Previously, I had added the ast_sched_thread stuff that was a generic scheduler
thread implementation.  However, if you used it, it required using different
functions for modifying scheduler contents.  This patch reworks how this is
done and just allows you to optionally start a thread on the original scheduler
context structure that has always been there.  This makes it trivial to switch
to the generic scheduler thread implementation without having to touch any of
the other code that adds or removes scheduler entries.

In passing, I made some naming tweaks to add ast_ prefixes where they were not
there before.

Review: https://reviewboard.asterisk.org/r/1007/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299091 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-20 17:15:54 +00:00
Tilghman Lesher b4e18d5660 Add load priority order, such that preload becomes unnecessary in most cases
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278132 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-20 19:35:02 +00:00
Mark Michelson cd4ebd336f Add IPv6 to Asterisk.
This adds a generic API for accommodating IPv6 and IPv4 addresses
within Asterisk. While many files have been updated to make use of the
API, chan_sip and the RTP code are the files which actually support
IPv6 addresses at the time of this commit. The way has been paved for
easier upgrading for other files in the near future, though.

Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne
for their hard work on this.

(closes issue #17565)
Reported by: russell
Patches: 
      asteriskv6-test-report.pdf uploaded by russell (license 2)

Review: https://reviewboard.asterisk.org/r/743



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274783 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-08 22:08:07 +00:00
Tilghman Lesher d8e0c58437 Expand codec bitfield from 32 bits to 64 bits.
Reviewboard: https://reviewboard.asterisk.org/r/416/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-04 14:05:12 +00:00
Russell Bryant 844a01b27e Add an "Asterisk Architecture Overview" section to the doxygen documentation.
This is a side project I've been poking at this week.  The intent is to discuss
Asterisk architecture in a top down fashion to help new developers understand how
Asterisk is put together.  There is a ton of stuff to write about, so this will
just continue to evolve over time.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-10-30 04:08:39 +00:00
Joshua Colp ae87ba45b5 Add support for multicast RTP paging.
(closes issue #11797)
Reported by: macbrody

Review: https://reviewboard.asterisk.org/r/270/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-25 18:25:24 +00:00