ast_rtp_glue Struct Reference

#include <rtp_engine.h>

Collaboration diagram for ast_rtp_glue:

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Data Fields

int(* allow_rtp_remote )(struct ast_channel *chan1, struct ast_rtp_instance *instance)
 Used to prevent two channels from remotely bridging audio rtp if the channel tech has a reason for prohibiting it based on qualities that need to be compared from both channels.
int(* allow_vrtp_remote )(struct ast_channel *chan1, struct ast_rtp_instance *instance)
 Used to prevent two channels from remotely bridging video rtp if the channel tech has a reason for prohibiting it based on qualities that need to be compared from both channels.
struct {
   struct ast_rtp_glue *   next
entry
void(* get_codec )(struct ast_channel *chan, struct ast_format_cap *result_cap)
enum ast_rtp_glue_result(* get_rtp_info )(struct ast_channel *chan, struct ast_rtp_instance **instance)
 Callback for retrieving the RTP instance carrying audio.
enum ast_rtp_glue_result(* get_trtp_info )(struct ast_channel *chan, struct ast_rtp_instance **instance)
 Callback for retrieving the RTP instance carrying text.
enum ast_rtp_glue_result(* get_vrtp_info )(struct ast_channel *chan, struct ast_rtp_instance **instance)
 Callback for retrieving the RTP instance carrying video.
struct ast_modulemod
const char * type
int(* update_peer )(struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_rtp_instance *vinstance, struct ast_rtp_instance *tinstance, const struct ast_format_cap *cap, int nat_active)

Detailed Description

Structure that represents the glue that binds an RTP instance to a channel

Definition at line 593 of file rtp_engine.h.


Field Documentation

struct ast_rtp_glue* next [read]

Definition at line 631 of file rtp_engine.h.

const char* type

Name of the channel driver that this glue is responsible for

Definition at line 595 of file rtp_engine.h.

Referenced by ast_rtp_glue_register2(), ast_rtp_glue_unregister(), ast_rtp_instance_get_glue(), and ast_rtp_instance_make_compatible().


The documentation for this struct was generated from the following file:

Generated on Thu Apr 16 06:35:29 2015 for Asterisk - The Open Source Telephony Project by  doxygen 1.5.6