[svn-commits] trunk r26019 - in /trunk: ./ apps/ channels/
include/asterisk/
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Tue May 9 04:44:50 MST 2006
Author: markster
Date: Tue May 9 06:44:50 2006
New Revision: 26019
URL: http://svn.digium.com/view/asterisk?rev=26019&view=rev
Log:
Make SIP early media work more efficiently without so many reinvites
Modified:
trunk/apps/app_dial.c
trunk/channels/chan_sip.c
trunk/include/asterisk/rtp.h
trunk/rtp.c
Modified: trunk/apps/app_dial.c
URL: http://svn.digium.com/view/asterisk/trunk/apps/app_dial.c?rev=26019&r1=26018&r2=26019&view=diff
==============================================================================
--- trunk/apps/app_dial.c (original)
+++ trunk/apps/app_dial.c Tue May 9 06:44:50 2006
@@ -482,7 +482,7 @@
ast_clear_flag(o, DIAL_STILLGOING);
HANDLE_CAUSE(cause, in);
} else {
- ast_rtp_make_compatible(c, in);
+ ast_rtp_make_compatible(c, in, single);
if (c->cid.cid_num)
free(c->cid.cid_num);
c->cid.cid_num = NULL;
@@ -550,6 +550,8 @@
OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
DIAL_NOFORWARDHTML);
+ /* Setup early media if appropriate */
+ ast_rtp_early_media(in, peer);
}
/* If call has been answered, then the eventual hangup is likely to be normal hangup */
in->hangupcause = AST_CAUSE_NORMAL_CLEARING;
@@ -576,6 +578,9 @@
case AST_CONTROL_RINGING:
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "%s is ringing\n", c->name);
+ /* Setup early media if appropriate */
+ if (single)
+ ast_rtp_early_media(in, c);
if (!(*sentringing) && !ast_test_flag(outgoing, OPT_MUSICBACK)) {
ast_indicate(in, AST_CONTROL_RINGING);
(*sentringing)++;
@@ -584,6 +589,9 @@
case AST_CONTROL_PROGRESS:
if (option_verbose > 2)
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);
if (!ast_test_flag(outgoing, OPT_RINGBACK))
ast_indicate(in, AST_CONTROL_PROGRESS);
break;
@@ -595,6 +603,8 @@
case AST_CONTROL_PROCEEDING:
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);
if (!ast_test_flag(outgoing, OPT_RINGBACK))
ast_indicate(in, AST_CONTROL_PROCEEDING);
break;
@@ -1056,7 +1066,7 @@
}
/* Setup outgoing SDP to match incoming one */
- ast_rtp_make_compatible(tmp->chan, chan);
+ ast_rtp_make_compatible(tmp->chan, chan, !outgoing && !rest);
/* Inherit specially named variables from parent channel */
ast_channel_inherit_variables(chan, tmp->chan);
@@ -1550,6 +1560,7 @@
sentringing = 0;
ast_indicate(chan, -1);
}
+ ast_rtp_early_media(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=26019&r1=26018&r2=26019&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Tue May 9 06:44:50 2006
@@ -13588,6 +13588,7 @@
static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs, int nat_active)
{
struct sip_pvt *p;
+ int changed = 0;
p = chan->tech_pvt;
if (!p)
@@ -13598,17 +13599,23 @@
ast_mutex_unlock(&p->lock);
return 0;
}
- if (rtp)
- ast_rtp_get_peer(rtp, &p->redirip);
+ if (rtp)
+ changed |= ast_rtp_get_peer(rtp, &p->redirip);
else
memset(&p->redirip, 0, sizeof(p->redirip));
if (vrtp)
- ast_rtp_get_peer(vrtp, &p->vredirip);
+ changed |= ast_rtp_get_peer(vrtp, &p->vredirip);
else
memset(&p->vredirip, 0, sizeof(p->vredirip));
- p->redircodecs = codecs;
- if (!ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
- if (!p->pendinginvite) {
+ if (p->redircodecs != codecs) {
+ p->redircodecs = codecs;
+ changed = 1;
+ }
+ if (changed && !ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
+ if (chan->_state != AST_STATE_UP) {
+ char iabuf[INET_ADDRSTRLEN];
+ ast_log(LOG_DEBUG, "Early media setting SIP '%s' - Sending early media to %s\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), rtp ? p->redirip.sin_addr : p->ourip));
+ } else if (!p->pendinginvite) {
if (option_debug > 2) {
char iabuf[INET_ADDRSTRLEN];
ast_log(LOG_DEBUG, "Sending reinvite on SIP '%s' - It's audio soon redirected to IP %s\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), rtp ? p->redirip.sin_addr : p->ourip));
Modified: trunk/include/asterisk/rtp.h
URL: http://svn.digium.com/view/asterisk/trunk/include/asterisk/rtp.h?rev=26019&r1=26018&r2=26019&view=diff
==============================================================================
--- trunk/include/asterisk/rtp.h (original)
+++ trunk/include/asterisk/rtp.h Tue May 9 06:44:50 2006
@@ -97,7 +97,8 @@
void ast_rtp_set_peer(struct ast_rtp *rtp, struct sockaddr_in *them);
-void ast_rtp_get_peer(struct ast_rtp *rtp, struct sockaddr_in *them);
+/* Copies from rtp to them and returns 1 if there was a change or 0 if it was already the same */
+int ast_rtp_get_peer(struct ast_rtp *rtp, struct sockaddr_in *them);
void ast_rtp_get_us(struct ast_rtp *rtp, struct sockaddr_in *us);
@@ -154,7 +155,9 @@
void ast_rtp_proto_unregister(struct ast_rtp_protocol *proto);
-int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src);
+int 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);
void ast_rtp_stop(struct ast_rtp *rtp);
Modified: trunk/rtp.c
URL: http://svn.digium.com/view/asterisk/trunk/rtp.c?rev=26019&r1=26018&r2=26019&view=diff
==============================================================================
--- trunk/rtp.c (original)
+++ trunk/rtp.c Tue May 9 06:44:50 2006
@@ -733,11 +733,83 @@
return cur;
}
-int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src)
+int ast_rtp_early_media(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 */
+ struct ast_rtp_protocol *destpr, *srcpr=NULL;
+ int srccodec;
+ /* Lock channels */
+ ast_channel_lock(dest);
+ if (src) {
+ while(ast_channel_trylock(src)) {
+ ast_channel_unlock(dest);
+ usleep(1);
+ ast_channel_lock(dest);
+ }
+ }
+
+ /* Find channel driver interfaces */
+ destpr = get_proto(dest);
+ if (src)
+ srcpr = get_proto(src);
+ 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;
+ }
+ if (!srcpr) {
+ if (option_debug)
+ ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", src->name);
+ ast_channel_unlock(dest);
+ if (src)
+ ast_channel_unlock(src);
+ return 0;
+ }
+
+ /* Get audio and video interface (if native bridge is possible) */
+ destp = destpr->get_rtp_info(dest);
+ vdestp = (destpr->get_vrtp_info) ? destpr->get_vrtp_info(dest) : NULL;
+ if (srcpr) {
+ srcp = srcpr->get_rtp_info(src);
+ vsrcp = (srcpr->get_vrtp_info) ? srcpr->get_vrtp_info(src) : NULL;
+ }
+
+ /* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
+ if (!destp) {
+ /* Somebody doesn't want to play... */
+ ast_channel_unlock(dest);
+ if (src)
+ ast_channel_unlock(src);
+ return 0;
+ }
+ if (srcpr && srcpr->get_codec)
+ srccodec = srcpr->get_codec(src);
+ else
+ srccodec = 0;
+ /* Consider empty media as non-existant */
+ if (srcp && !srcp->them.sin_addr.s_addr)
+ srcp = NULL;
+ /* Bridge early media */
+ 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 send early media to '%s'\n", dest->name, src ? src->name : "<unspecified>");
+ ast_channel_unlock(dest);
+ if (src)
+ ast_channel_unlock(src);
+ if (option_debug)
+ ast_log(LOG_DEBUG, "Setting early media SDP of '%s' with that of '%s'\n", dest->name, src ? src->name : "<unspecified>");
+ return 1;
+}
+
+int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, int media)
{
struct ast_rtp *destp, *srcp; /* Audio RTP Channels */
struct ast_rtp *vdestp, *vsrcp; /* Video RTP channels */
struct ast_rtp_protocol *destpr, *srcpr;
+ int srccodec;
/* Lock channels */
ast_channel_lock(dest);
while(ast_channel_trylock(src)) {
@@ -780,6 +852,15 @@
ast_rtp_pt_copy(destp, srcp);
if (vdestp && vsrcp)
ast_rtp_pt_copy(vdestp, vsrcp);
+ if (srcpr->get_codec)
+ srccodec = srcpr->get_codec(src);
+ else
+ srccodec = 0;
+ if (media) {
+ /* Bridge early media */
+ if (destpr->set_rtp_peer(dest, srcp, vsrcp, srccodec, ast_test_flag(srcp, FLAG_NAT_ACTIVE)))
+ ast_log(LOG_WARNING, "Channel '%s' failed to send early media to '%s'\n", dest->name, src->name);
+ }
ast_channel_unlock(dest);
ast_channel_unlock(src);
if (option_debug)
@@ -1086,11 +1167,17 @@
rtp->rxseqno = 0;
}
-void ast_rtp_get_peer(struct ast_rtp *rtp, struct sockaddr_in *them)
-{
- them->sin_family = AF_INET;
- them->sin_port = rtp->them.sin_port;
- them->sin_addr = rtp->them.sin_addr;
+int ast_rtp_get_peer(struct ast_rtp *rtp, struct sockaddr_in *them)
+{
+ if ((them->sin_family != AF_INET) ||
+ (them->sin_port != rtp->them.sin_port) ||
+ (them->sin_addr.s_addr != rtp->them.sin_addr.s_addr)) {
+ them->sin_family = AF_INET;
+ them->sin_port = rtp->them.sin_port;
+ them->sin_addr = rtp->them.sin_addr;
+ return 1;
+ }
+ return 0;
}
void ast_rtp_get_us(struct ast_rtp *rtp, struct sockaddr_in *us)
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