diff --git a/apps/app_dial.c b/apps/app_dial.c
index 6dddb855b4bc1ea86b227adbe4bbb920ae349b1a..7d97187e3ab8c72c00fc7484db60b6bea869bf61 100644
--- a/apps/app_dial.c
+++ b/apps/app_dial.c
@@ -482,7 +482,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct dial_l
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 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct dial_l
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 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct dial_l
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 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct dial_l
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 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct dial_l
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 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
}
/* 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 @@ out:
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)
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 4d4a3e1efa85cc0b93df004dac931546be5152cd..5881d2d96390723e10c1d4b9b858efd84e89838c 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -13588,6 +13588,7 @@ static struct ast_rtp *sip_get_vrtp_peer(struct ast_channel *chan)
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 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
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));
diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h
index 3b19da0cc7709f4d73f64a4e9cef88f7d0509227..1dc133cd4567b0e9bd3e617327a65e7a74e28095 100644
--- a/include/asterisk/rtp.h
+++ b/include/asterisk/rtp.h
@@ -97,7 +97,8 @@ struct ast_rtp *ast_rtp_new_with_bindaddr(struct sched_context *sched, struct io
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 @@ int ast_rtp_proto_register(struct ast_rtp_protocol *proto);
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);
diff --git a/rtp.c b/rtp.c
index 88aa1eefd53e92ffeeccb32c7c0a5d1ae28a0bcb..b80fb306968a7e7a73e1115930eee33cfbdede02 100644
--- a/rtp.c
+++ b/rtp.c
@@ -733,11 +733,83 @@ static struct ast_rtp_protocol *get_proto(struct ast_channel *chan)
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 @@ int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src)
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 @@ void ast_rtp_set_peer(struct ast_rtp *rtp, struct sockaddr_in *them)
rtp->rxseqno = 0;
}
-void ast_rtp_get_peer(struct ast_rtp *rtp, struct sockaddr_in *them)
+int 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;
+ 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)