diff options
| -rw-r--r-- | apps/app_dial.c | 12 | ||||
| -rw-r--r-- | include/asterisk/rtp.h | 4 | ||||
| -rw-r--r-- | rtp.c | 2 |
3 files changed, 10 insertions, 8 deletions
diff --git a/apps/app_dial.c b/apps/app_dial.c index 56855337b..5361bdc6a 100644 --- a/apps/app_dial.c +++ b/apps/app_dial.c @@ -576,8 +576,8 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct dial_l OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR | OPT_CALLEE_PARK | OPT_CALLER_PARK | DIAL_NOFORWARDHTML); - /* Setup early media if appropriate */ - ast_rtp_early_media(in, peer); + /* Setup RTP early bridge if appropriate */ + ast_rtp_early_bridge(in, peer); } /* If call has been answered, then the eventual hangup is likely to be normal hangup */ in->hangupcause = AST_CAUSE_NORMAL_CLEARING; @@ -606,7 +606,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct dial_l ast_verbose(VERBOSE_PREFIX_3 "%s is ringing\n", c->name); /* Setup early media if appropriate */ if (single) - ast_rtp_early_media(in, c); + ast_rtp_early_bridge(in, c); if (!(*sentringing) && !ast_test_flag(outgoing, OPT_MUSICBACK)) { ast_indicate(in, AST_CONTROL_RINGING); (*sentringing)++; @@ -617,7 +617,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct dial_l ast_verbose (VERBOSE_PREFIX_3 "%s is making progress passing it to %s\n", c->name, in->name); /* Setup early media if appropriate */ if (single) - ast_rtp_early_media(in, c); + ast_rtp_early_bridge(in, c); if (!ast_test_flag(outgoing, OPT_RINGBACK)) ast_indicate(in, AST_CONTROL_PROGRESS); break; @@ -630,7 +630,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct dial_l if (option_verbose > 2) ast_verbose (VERBOSE_PREFIX_3 "%s is proceeding passing it to %s\n", c->name, in->name); if (single) - ast_rtp_early_media(in, c); + ast_rtp_early_bridge(in, c); if (!ast_test_flag(outgoing, OPT_RINGBACK)) ast_indicate(in, AST_CONTROL_PROCEEDING); break; @@ -1608,7 +1608,7 @@ out: sentringing = 0; ast_indicate(chan, -1); } - ast_rtp_early_media(chan, NULL); + ast_rtp_early_bridge(chan, NULL); hanguptree(outgoing, NULL); pbx_builtin_setvar_helper(chan, "DIALSTATUS", status); if (option_debug) diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h index b7be53a1c..3bd105169 100644 --- a/include/asterisk/rtp.h +++ b/include/asterisk/rtp.h @@ -182,7 +182,9 @@ void ast_rtp_proto_unregister(struct ast_rtp_protocol *proto); int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, int media); -int ast_rtp_early_media(struct ast_channel *dest, struct ast_channel *src); +/*! \brief If possible, create an early bridge directly between the devices without + having to send a re-invite later */ +int ast_rtp_early_bridge(struct ast_channel *dest, struct ast_channel *src); void ast_rtp_stop(struct ast_rtp *rtp); @@ -1274,7 +1274,7 @@ static struct ast_rtp_protocol *get_proto(struct ast_channel *chan) return cur; } -int ast_rtp_early_media(struct ast_channel *dest, struct ast_channel *src) +int ast_rtp_early_bridge(struct ast_channel *dest, struct ast_channel *src) { struct ast_rtp *destp, *srcp=NULL; /* Audio RTP Channels */ struct ast_rtp *vdestp, *vsrcp=NULL; /* Video RTP channels */ |