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Use rtp properties instead of adding a callback

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Thanks, Josh.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221278 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Terry Wilson
2009-09-30 18:21:03 +00:00
parent 865daf4858
commit 10ce6cd757
4 changed files with 11 additions and 48 deletions
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19
include/asterisk/rtp_engine.h
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@@ -94,6 +94,8 @@ enum ast_rtp_property {
AST_RTP_PROPERTY_RTCP,
/*! Maximum number of RTP properties supported */
AST_RTP_PROPERTY_MAX,
/*! Don't force a new SSRC on new source */
AST_RTP_PROPERTY_CONSTANT_SSRC,
};
/*! Additional RTP options */
@@ -1184,23 +1186,6 @@ int ast_rtp_instance_dtmf_mode_set(struct ast_rtp_instance *instance, enum ast_r
*/
enum ast_rtp_dtmf_mode ast_rtp_instance_dtmf_mode_get(struct ast_rtp_instance *instance);
/*!
* \brief Mark an RTP instance not to update SSRC on a new source
*
* \param instance Instance to update
*
* Example usage:
*
* \code
* ast_rtp_instance_set_constantssrc(instance);
* \endcode
*
* This sets the indicated instance to not update the RTP SSRC when new_source
* is called.
*
* \since 1.6.3
*/
void ast_rtp_instance_set_constantssrc(struct ast_rtp_instance *instance);
/*!
* \brief Indicate a new source of audio has dropped in
*