diff options
author | Sean Bright <sean@malleable.com> | 2009-07-03 02:02:50 +0000 |
---|---|---|
committer | Sean Bright <sean@malleable.com> | 2009-07-03 02:02:50 +0000 |
commit | c381cf82e72779142b4863e3d49e37fb4612fe7c (patch) | |
tree | 3bcece3bf3ad36f7582b7419bf6e611b8f3e35f5 /include/asterisk | |
parent | a894c33cb32bcee4bc6152fca7067c1e13752b66 (diff) |
Wrap rtp_engine.h header comments to 80 characters.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'include/asterisk')
-rw-r--r-- | include/asterisk/rtp_engine.h | 62 |
1 files changed, 34 insertions, 28 deletions
diff --git a/include/asterisk/rtp_engine.h b/include/asterisk/rtp_engine.h index 2baae32cf..d87feb9f4 100644 --- a/include/asterisk/rtp_engine.h +++ b/include/asterisk/rtp_engine.h @@ -26,34 +26,40 @@ /*! * \page AstRTPEngine Asterisk RTP Engine API * - * The purpose of this API is to provide a way for multiple RTP stacks to be used inside - * of Asterisk without any module that uses RTP knowing any different. To the module each RTP - * stack behaves the same. - * - * An RTP session is called an instance and is made up of a combination of codec information, - * RTP engine, RTP properties, and address information. An engine name may be passed in to explicitly - * choose an RTP stack to be used but a default one will be used if none is provided. An address to use - * for RTP may also be provided but the underlying RTP engine may choose a different address depending on - * it's configuration. - * - * An RTP engine is the layer between the RTP engine core and the RTP stack itself. The RTP engine core provides - * a set of callbacks to do various things (such as write audio out) that the RTP engine has to have implemented. - * - * Glue is what binds an RTP instance to a channel. It is used to retrieve RTP instance information when - * performing remote or local bridging and is used to have the channel driver tell the remote side to change - * destination of the RTP stream. - * - * Statistics from an RTP instance can be retrieved using the ast_rtp_instance_get_stats API call. This essentially - * asks the RTP engine in use to fill in a structure with the requested values. It is not required for an RTP engine - * to support all statistic values. - * - * Properties allow behavior of the RTP engine and RTP engine core to be changed. For example, there is a property named - * AST_RTP_PROPERTY_NAT which is used to tell the RTP engine to enable symmetric RTP if it supports it. It is not required - * for an RTP engine to support all properties. - * - * Codec information is stored using a separate data structure which has it's own set of API calls to add/remove/retrieve - * information. They are used by the module after an RTP instance is created so that payload information is available for - * the RTP engine. + * The purpose of this API is to provide a way for multiple RTP stacks to be + * used inside of Asterisk without any module that uses RTP knowing any + * different. To the module each RTP stack behaves the same. + * + * An RTP session is called an instance and is made up of a combination of codec + * information, RTP engine, RTP properties, and address information. An engine + * name may be passed in to explicitly choose an RTP stack to be used but a + * default one will be used if none is provided. An address to use for RTP may + * also be provided but the underlying RTP engine may choose a different address + * depending on it's configuration. + * + * An RTP engine is the layer between the RTP engine core and the RTP stack + * itself. The RTP engine core provides a set of callbacks to do various things + * (such as write audio out) that the RTP engine has to have implemented. + * + * Glue is what binds an RTP instance to a channel. It is used to retrieve RTP + * instance information when performing remote or local bridging and is used to + * have the channel driver tell the remote side to change destination of the RTP + * stream. + * + * Statistics from an RTP instance can be retrieved using the + * ast_rtp_instance_get_stats API call. This essentially asks the RTP engine in + * use to fill in a structure with the requested values. It is not required for + * an RTP engine to support all statistic values. + * + * Properties allow behavior of the RTP engine and RTP engine core to be + * changed. For example, there is a property named AST_RTP_PROPERTY_NAT which is + * used to tell the RTP engine to enable symmetric RTP if it supports it. It is + * not required for an RTP engine to support all properties. + * + * Codec information is stored using a separate data structure which has it's + * own set of API calls to add/remove/retrieve information. They are used by the + * module after an RTP instance is created so that payload information is + * available for the RTP engine. */ #ifndef _ASTERISK_RTP_ENGINE_H |