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Add jitterbuffer support for chan_local. To enable it, you use the 'j' option 

in the Dial command.  The 'j' option _must_ be used in conjunction with the 'n'
option.

This feature will allow you to use the existing jitterbuffer implementation to
put a jitterbuffer on incoming SIP calls connecting to Asterisk applications by
putting a local channel in the middle.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85097 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Russell Bryant 2007-10-09 15:10:14 +00:00
parent d9909a0151
commit 46b9ca721b
2 changed files with 22 additions and 0 deletions
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20
channels/chan_local.c
View File

@ -66,6 +66,13 @@ static const char tdesc[] = "Local Proxy Channel Driver";
#define IS_OUTBOUND(a,b) (a == b->chan ? 1 : 0)
static struct ast_jb_conf g_jb_conf = {
.flags = 0,
.max_size = -1,
.resync_threshold = -1,
.impl = "",
};
static struct ast_channel *local_request(const char *type, int format, void *data, int *cause);
static int local_digit_begin(struct ast_channel *ast, char digit);
static int local_digit_end(struct ast_channel *ast, char digit, unsigned int duration);
@ -108,6 +115,7 @@ struct local_pvt {
char context[AST_MAX_CONTEXT]; /* Context to call */
char exten[AST_MAX_EXTENSION]; /* Extension to call */
int reqformat; /* Requested format */
struct ast_jb_conf jb_conf; /*!< jitterbuffer configuration for this local channel */
struct ast_channel *owner; /* Master Channel */
struct ast_channel *chan; /* Outbound channel */
struct ast_module_user *u_owner; /*! reference to keep the module loaded while in use */
@ -563,11 +571,21 @@ static struct local_pvt *local_alloc(const char *data, int format)
ast_mutex_init(&tmp->lock);
ast_copy_string(tmp->exten, data, sizeof(tmp->exten));
memcpy(&tmp->jb_conf, &g_jb_conf, sizeof(tmp->jb_conf));
/* Look for options */
if ((opts = strchr(tmp->exten, '/'))) {
*opts++ = '\0';
if (strchr(opts, 'n'))
ast_set_flag(tmp, LOCAL_NO_OPTIMIZATION);
if (strchr(opts, 'j')) {
if (ast_test_flag(tmp, LOCAL_NO_OPTIMIZATION))
ast_set_flag(&tmp->jb_conf, AST_JB_ENABLED);
else {
ast_log(LOG_ERROR, "You must use the 'n' option for chan_local "
"to use the 'j' option to enable the jitterbuffer\n");
}
}
}
/* Look for a context */
@ -652,6 +670,8 @@ static struct ast_channel *local_new(struct local_pvt *p, int state)
tmp->priority = 1;
tmp2->priority = 1;
ast_jb_configure(tmp, &p->jb_conf);
return tmp;
}

2
doc/tex/localchannel.tex
View File

@ -9,6 +9,8 @@ chan\_local is a pseudo-channel. Use of this channel simply loops calls back int
Adding "/n" at the end of the string will make the Local channel not do a native transfer (the "n" stands for "n"o release) upon the remote end answering the line. This is an esoteric, but important feature if you expect the Local channel to handle calls exactly like a normal channel. If you do not have the "no release" feature set, then as soon as the destination (inside of the Local channel) answers the line, the variables and dial plan will revert back to that of the original call, and the Local channel will become a zombie and be removed from the active channels list. This is desirable in some circumstances, but can result in unexpected dialplan behavior if you are doing fancy things with variables in your call handling.
There is another option that can be used with local channels, which is the "j" option. The "j" option must be used with the "n" option to make sure that the local channel does not get optimized out of the call. This option will enable a jitterbuffer on the local channel. The jitterbuffer will be used to de-jitter audio that it receives from the channel that called the local channel. This is especially in the case of putting chan\_local in between an incoming SIP call and Asterisk applications, so that the incoming audio will be de-jittered.
\subsection{Purpose}
The Local channel construct can be used to establish dialing into any part of the dialplan.