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Add support for using a jitterbuffer for RTP on bridged calls. This includes

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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: https://origsvn.digium.com/svn/asterisk/trunk@31052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Russell Bryant
2006-05-31 16:56:50 +00:00
parent 503486e99d
commit bb7dd96cfe
20 changed files with 983 additions and 20 deletions
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10
include/asterisk/channel.h
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@@ -86,6 +86,8 @@
#ifndef _ASTERISK_CHANNEL_H
#define _ASTERISK_CHANNEL_H
#include "asterisk/abstract_jb.h"
#include <unistd.h>
#ifdef POLLCOMPAT
#include "asterisk/poll-compat.h"
@@ -445,6 +447,9 @@ struct ast_channel {
/*! For easy linking */
AST_LIST_ENTRY(ast_channel) chan_list;
/*! The jitterbuffer state */
struct ast_jb jb;
};
/* \defgroup chanprop Channel tech properties:
@@ -452,6 +457,11 @@ struct ast_channel {
/* @{ */
#define AST_CHAN_TP_WANTSJITTER (1 << 0)
/* \defgroup chanprop Channel tech properties:
\brief Channels have this property if they can create jitter; i.e. most VoIP channels */
/* @{ */
#define AST_CHAN_TP_CREATESJITTER (1 << 1)
/* This flag has been deprecated by the transfercapbilty data member in struct ast_channel */
/* #define AST_FLAG_DIGITAL (1 << 0) */ /* if the call is a digital ISDN call */
#define AST_FLAG_DEFER_DTMF (1 << 1) /*!< if dtmf should be deferred */