[asterisk-commits] file: trunk r43437 - in /trunk: ./ apps/
channels/ include/asterisk/ main/
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Thu Sep 21 12:27:27 MST 2006
Author: file
Date: Thu Sep 21 14:27:26 2006
New Revision: 43437
URL: http://svn.digium.com/view/asterisk?rev=43437&view=rev
Log:
SS7 marked the start of an open season for trunk again but here's something minor - abstract early bridging into the technology so that we don't always assume they use RTP and try it.
Modified:
trunk/ (props changed)
trunk/apps/app_dial.c
trunk/channels/chan_sip.c
trunk/include/asterisk/channel.h
trunk/include/asterisk/rtp.h
trunk/main/channel.c
trunk/main/rtp.c
Propchange: trunk/
------------------------------------------------------------------------------
automerge-email = jcolp at digium.com
Propchange: trunk/
------------------------------------------------------------------------------
svnmerge-integrated = /trunk:1-43433
Modified: trunk/apps/app_dial.c
URL: http://svn.digium.com/view/asterisk/trunk/apps/app_dial.c?rev=43437&r1=43436&r2=43437&view=diff
==============================================================================
--- trunk/apps/app_dial.c (original)
+++ trunk/apps/app_dial.c Thu Sep 21 14:27:26 2006
@@ -575,8 +575,8 @@
OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
OPT_CALLEE_PARK | OPT_CALLER_PARK |
DIAL_NOFORWARDHTML);
- /* Setup RTP early bridge if appropriate */
- ast_rtp_early_bridge(in, peer);
+ /* Setup early bridge if appropriate */
+ ast_channel_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;
@@ -605,7 +605,7 @@
ast_verbose(VERBOSE_PREFIX_3 "%s is ringing\n", c->name);
/* Setup early media if appropriate */
if (single)
- ast_rtp_early_bridge(in, c);
+ ast_channel_early_bridge(in, c);
if (!(*sentringing) && !ast_test_flag(outgoing, OPT_MUSICBACK)) {
ast_indicate(in, AST_CONTROL_RINGING);
(*sentringing)++;
@@ -616,7 +616,7 @@
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_bridge(in, c);
+ ast_channel_early_bridge(in, c);
if (!ast_test_flag(outgoing, OPT_RINGBACK))
ast_indicate(in, AST_CONTROL_PROGRESS);
break;
@@ -629,7 +629,7 @@
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_bridge(in, c);
+ ast_channel_early_bridge(in, c);
if (!ast_test_flag(outgoing, OPT_RINGBACK))
ast_indicate(in, AST_CONTROL_PROCEEDING);
break;
@@ -1624,7 +1624,7 @@
sentringing = 0;
ast_indicate(chan, -1);
}
- ast_rtp_early_bridge(chan, NULL);
+ ast_channel_early_bridge(chan, NULL);
hanguptree(outgoing, NULL);
pbx_builtin_setvar_helper(chan, "DIALSTATUS", status);
if (option_debug)
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?rev=43437&r1=43436&r2=43437&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Thu Sep 21 14:27:26 2006
@@ -1527,6 +1527,7 @@
.send_digit_begin = sip_senddigit_begin,
.send_digit_end = sip_senddigit_end,
.bridge = ast_rtp_bridge,
+ .early_bridge = ast_rtp_early_bridge,
.send_text = sip_sendtext,
};
Modified: trunk/include/asterisk/channel.h
URL: http://svn.digium.com/view/asterisk/trunk/include/asterisk/channel.h?rev=43437&r1=43436&r2=43437&view=diff
==============================================================================
--- trunk/include/asterisk/channel.h (original)
+++ trunk/include/asterisk/channel.h Thu Sep 21 14:27:26 2006
@@ -237,6 +237,9 @@
/*! \brief Bridge two channels of the same type together */
enum ast_bridge_result (* const bridge)(struct ast_channel *c0, struct ast_channel *c1, int flags,
struct ast_frame **fo, struct ast_channel **rc, int timeoutms);
+
+ /*! \brief Bridge two channels of the same type together (early) */
+ enum ast_bridge_result (* const early_bridge)(struct ast_channel *c0, struct ast_channel *c1);
/*! \brief Indicate a particular condition (e.g. AST_CONTROL_BUSY or AST_CONTROL_RINGING or AST_CONTROL_CONGESTION */
int (* const indicate)(struct ast_channel *c, int condition, const void *data, size_t datalen);
@@ -965,6 +968,13 @@
* \return Returns 0 on success and -1 if it could not be done */
int ast_channel_make_compatible(struct ast_channel *c0, struct ast_channel *c1);
+/*! Bridge two channels together (early)
+ * \param c0 first channel to bridge
+ * \param c1 second channel to bridge
+ * Bridge two channels (c0 and c1) together early. This implies either side may not be answered yet.
+ * \return Returns 0 on success and -1 if it could not be done */
+int ast_channel_early_bridge(struct ast_channel *c0, struct ast_channel *c1);
+
/*! Bridge two channels together
* \param c0 first channel to bridge
* \param c1 second channel to bridge
Modified: trunk/include/asterisk/rtp.h
URL: http://svn.digium.com/view/asterisk/trunk/include/asterisk/rtp.h?rev=43437&r1=43436&r2=43437&view=diff
==============================================================================
--- trunk/include/asterisk/rtp.h (original)
+++ trunk/include/asterisk/rtp.h Thu Sep 21 14:27:26 2006
@@ -196,7 +196,7 @@
/*! \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);
+int ast_rtp_early_bridge(struct ast_channel *c0, struct ast_channel *c1);
void ast_rtp_stop(struct ast_rtp *rtp);
Modified: trunk/main/channel.c
URL: http://svn.digium.com/view/asterisk/trunk/main/channel.c?rev=43437&r1=43436&r2=43437&view=diff
==============================================================================
--- trunk/main/channel.c (original)
+++ trunk/main/channel.c Thu Sep 21 14:27:26 2006
@@ -3675,6 +3675,16 @@
return res;
}
+/*! \brief Bridge two channels together (early) */
+int ast_channel_early_bridge(struct ast_channel *c0, struct ast_channel *c1)
+{
+ /* Make sure we can early bridge, if not error out */
+ if (!c0->tech->early_bridge || (c1 && (!c1->tech->early_bridge || c0->tech->early_bridge != c1->tech->early_bridge)))
+ return -1;
+
+ return c0->tech->early_bridge(c0, c1);
+}
+
/*! \brief Bridge two channels together */
enum ast_bridge_result ast_channel_bridge(struct ast_channel *c0, struct ast_channel *c1,
struct ast_bridge_config *config, struct ast_frame **fo, struct ast_channel **rc)
Modified: trunk/main/rtp.c
URL: http://svn.digium.com/view/asterisk/trunk/main/rtp.c?rev=43437&r1=43436&r2=43437&view=diff
==============================================================================
--- trunk/main/rtp.c (original)
+++ trunk/main/rtp.c Thu Sep 21 14:27:26 2006
@@ -1374,8 +1374,9 @@
return cur;
}
-int ast_rtp_early_bridge(struct ast_channel *dest, struct ast_channel *src)
-{
+int ast_rtp_early_bridge(struct ast_channel *c0, struct ast_channel *c1)
+{
+ // dest = c0, src = c1
struct ast_rtp *destp = NULL, *srcp = NULL; /* Audio RTP Channels */
struct ast_rtp *vdestp = NULL, *vsrcp = NULL; /* Video RTP channels */
struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL;
@@ -1384,68 +1385,68 @@
int srccodec;
/* Lock channels */
- ast_channel_lock(dest);
- if (src) {
- while(ast_channel_trylock(src)) {
- ast_channel_unlock(dest);
+ ast_channel_lock(c0);
+ if (c1) {
+ while(ast_channel_trylock(c1)) {
+ ast_channel_unlock(c0);
usleep(1);
- ast_channel_lock(dest);
+ ast_channel_lock(c0);
}
}
/* Find channel driver interfaces */
- destpr = get_proto(dest);
- if (src)
- srcpr = get_proto(src);
+ destpr = get_proto(c0);
+ if (c1)
+ srcpr = get_proto(c1);
if (!destpr) {
if (option_debug)
- ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name);
- ast_channel_unlock(dest);
- if (src)
- ast_channel_unlock(src);
- return 0;
+ ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", c0->name);
+ ast_channel_unlock(c0);
+ if (c1)
+ ast_channel_unlock(c1);
+ return -1;
}
if (!srcpr) {
if (option_debug)
- ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", src ? src->name : "<unspecified>");
- ast_channel_unlock(dest);
- if (src)
- ast_channel_unlock(src);
- return 0;
+ ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", c1 ? c1->name : "<unspecified>");
+ ast_channel_unlock(c0);
+ if (c1)
+ ast_channel_unlock(c1);
+ return -1;
}
/* Get audio and video interface (if native bridge is possible) */
- audio_dest_res = destpr->get_rtp_info(dest, &destp);
- video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(dest, &vdestp) : AST_RTP_GET_FAILED;
+ audio_dest_res = destpr->get_rtp_info(c0, &destp);
+ video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(c0, &vdestp) : AST_RTP_GET_FAILED;
if (srcpr) {
- audio_src_res = srcpr->get_rtp_info(src, &srcp);
- video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(src, &vsrcp) : AST_RTP_GET_FAILED;
+ audio_src_res = srcpr->get_rtp_info(c1, &srcp);
+ video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(c1, &vsrcp) : AST_RTP_GET_FAILED;
}
/* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
if (audio_dest_res != AST_RTP_TRY_NATIVE) {
/* Somebody doesn't want to play... */
- ast_channel_unlock(dest);
- if (src)
- ast_channel_unlock(src);
- return 0;
+ ast_channel_unlock(c0);
+ if (c1)
+ ast_channel_unlock(c1);
+ return -1;
}
if (audio_src_res == AST_RTP_TRY_NATIVE && srcpr->get_codec)
- srccodec = srcpr->get_codec(src);
+ srccodec = srcpr->get_codec(c1);
else
srccodec = 0;
/* Consider empty media as non-existant */
if (audio_src_res == AST_RTP_TRY_NATIVE && !srcp->them.sin_addr.s_addr)
srcp = NULL;
/* Bridge media early */
- if (destpr->set_rtp_peer(dest, srcp, vsrcp, srccodec, srcp ? ast_test_flag(srcp, FLAG_NAT_ACTIVE) : 0))
- ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", dest->name, src ? src->name : "<unspecified>");
- ast_channel_unlock(dest);
- if (src)
- ast_channel_unlock(src);
+ if (destpr->set_rtp_peer(c0, srcp, vsrcp, srccodec, srcp ? ast_test_flag(srcp, FLAG_NAT_ACTIVE) : 0))
+ ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge to '%s'\n", c0->name, c1 ? c1->name : "<unspecified>");
+ ast_channel_unlock(c0);
+ if (c1)
+ ast_channel_unlock(c1);
if (option_debug)
- ast_log(LOG_DEBUG, "Setting early bridge SDP of '%s' with that of '%s'\n", dest->name, src ? src->name : "<unspecified>");
- return 1;
+ ast_log(LOG_DEBUG, "Setting early bridge SDP of '%s' with that of '%s'\n", c0->name, c1 ? c1->name : "<unspecified>");
+ return 0;
}
int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, int media)
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