asterisk

Commit Graph

Author	SHA1	Message	Date
Russell Bryant	0c0664027d	Merged revisions 233692 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r233692 \| russell \| 2009-12-08 12:00:16 -0600 (Tue, 08 Dec 2009) \| 16 lines Set a module load priority for format modules. A recent change to app_voicemail made it such that the module now assumes that all format modules are available while processing voicemail configuration. However, when autoloading modules, it was possible that app_voicemail was loaded before the format modules. Since format modules don't depend on anything, set a module load priority on them to ensure that they get loaded first when autoloading. This fix applies to trunk, 1.6.1, and 1.6.2. The fix for 1.4 and 1.6.0 will require a different approach since the module load priority functionality is not present in the module API. (issue #16412) Reported by: jiddings ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@233694 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	2a53f2ec98	Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14) This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported. Along the way, some related work was done: 1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way. 2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec. 3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip must decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result). Review: http://reviewboard.digium.com/r/158/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago

Author

SHA1

Message

Date

Russell Bryant

0c0664027d

Merged revisions 233692 via svnmerge from

https://origsvn.digium.com/svn/asterisk/trunk

........
  r233692 | russell | 2009-12-08 12:00:16 -0600 (Tue, 08 Dec 2009) | 16 lines
  
  Set a module load priority for format modules.
  
  A recent change to app_voicemail made it such that the module now assumes that
  all format modules are available while processing voicemail configuration.
  However, when autoloading modules, it was possible that app_voicemail was
  loaded before the format modules.  Since format modules don't depend on
  anything, set a module load priority on them to ensure that they get loaded
  first when autoloading.
  
  This fix applies to trunk, 1.6.1, and 1.6.2.  The fix for 1.4 and 1.6.0 will
  require a different approach since the module load priority functionality is
  not present in the module API.
  
  (issue #16412)
  Reported by: jiddings
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@233694 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Kevin P. Fleming

2a53f2ec98

Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14)

This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported.

Along the way, some related work was done:

1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way.

2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec.

3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result).

Review: http://reviewboard.digium.com/r/158/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2 Commits (cef5e9d4340f94de8801aaecfb004a080df8a68f)