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- 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: http://svn.digium.com/svn/asterisk/trunk@40722 f38db490-d61c-443f-a65b-d21fe96a405b
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Essentially make sure an RTCP structure exists before trying to delete it's scheduled item. (issue #7514 reported by jmls fixed by AuPix)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@38573 f38db490-d61c-443f-a65b-d21fe96a405b
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inet_ntoa, which uses thread specific data (aka thread local storage) instead
of stack allocatted buffers to store the result.
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been requested (Sipura/Grandstream ATAs and others will need this)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@37501 f38db490-d61c-443f-a65b-d21fe96a405b
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split support for G726-32 into RFC3551 and AAL2 packing orders, since both are in use
change "G726-32" to be RFC3551 packing order, in spite of devices that use AAL2 order with this MIME type
add ability to directly transcode between packing orders
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@37494 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r37307 | file | 2006-07-07 19:57:53 -0400 (Fri, 07 Jul 2006) | 2 lines
Change message regarding marker bit forcing when SSRC changes to be shown only during debug so it doesn't overload high capacity systems
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- Add rtcp report to SIP history
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@33371 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@33108 f38db490-d61c-443f-a65b-d21fe96a405b
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again :-)
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and some patches (all disclaimed).
- Don't change RTP properties if we reject a re-INVITE
- Don't add video to an outbound channel if there's no video on the inbound channel
- Don't include video in the "preferred codec" list for codec selection
- Clean up and document code that parses and adds SDP attachments
Since we do not transcode video, we can't handle video the same way as audio. This is a
bug fix patch. In future releases, we need to work on a solution for video negotiation,
not codecs but formats and framerates instead.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@32597 f38db490-d61c-443f-a65b-d21fe96a405b
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Thanks to everyone involved in working with this!
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@32230 f38db490-d61c-443f-a65b-d21fe96a405b
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- Change formatting of RTP debug to align better
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@32088 f38db490-d61c-443f-a65b-d21fe96a405b
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- Debug message change
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@32087 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r31161 | kpfleming | 2006-05-31 17:26:38 -0500 (Wed, 31 May 2006) | 2 lines
silence a warning message that is not a warning
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@31162 f38db490-d61c-443f-a65b-d21fe96a405b
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a new implementation of a fixed size jitterbuffer, as well as support for the
existing adaptive jitterbuffer implementation. (issue #3854, Slav Klenov)
Thank you very much to Slav Klenov of Securax and all of the people involved
in the testing of this feature for all of your hard work!
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@31052 f38db490-d61c-443f-a65b-d21fe96a405b
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allow native bridging of RTP sessions that are not carrying DTMF even when the bridge needs to listen to DTMF (when SIP INFO is used for DTMF, for example)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@27559 f38db490-d61c-443f-a65b-d21fe96a405b
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lengths, ever)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@27524 f38db490-d61c-443f-a65b-d21fe96a405b
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- Adding if(option_debug) in front of complicated debug messages...
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@27305 f38db490-d61c-443f-a65b-d21fe96a405b
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support in SIP.
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- add information about subscriptions and realtime dial plans in sip.conf.sample
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@24342 f38db490-d61c-443f-a65b-d21fe96a405b
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by Mithraen)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@22905 f38db490-d61c-443f-a65b-d21fe96a405b
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- localize a variable
- replace three replicas of the same code with a much simpler construct
removing replicated var = var->next;
- use '?' instead of 'if' for an assignment
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@20224 f38db490-d61c-443f-a65b-d21fe96a405b
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- fix indentation in a few places
- use a variable to store the 'other' channel, thus removing
the need for some duplicated code;
- use '=' instead of memcpy to copy struct sockaddr_in
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@20197 f38db490-d61c-443f-a65b-d21fe96a405b
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rand() to threadsafe ast_random()
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@17627 f38db490-d61c-443f-a65b-d21fe96a405b
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M rtp.c
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r13095 | tilghman | 2006-03-15 12:07:06 -0600 (Wed, 15 Mar 2006) | 2 lines
Reverting patch from bug 6667
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r12792 | tilghman | 2006-03-13 09:28:35 -0600 (Mon, 13 Mar 2006) | 2 lines
Bug 6667 - Fix for RFC2833 issues
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const-ify some more APIs
remove 'type' field from ast_channel, in favor of the one in the channel's tech structure
allow string field module users to specify the 'chunk size' for pool allocations
update chan_alsa to be compatible with recent const-ification patches
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@9060 f38db490-d61c-443f-a65b-d21fe96a405b
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have to allocate one on the stack
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checks have passed and we're actually going to try it
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minor cleanups and simpler usage of list macros
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@7739 f38db490-d61c-443f-a65b-d21fe96a405b
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doxygen docs
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old local rtp stuff...
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