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Make SIP early media work more efficiently without so many reinvites

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer
2006-05-09 11:44:50 +00:00
parent 66dda468d1
commit 9953f4f40e
4 changed files with 123 additions and 15 deletions
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15
apps/app_dial.c
View File

@@ -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)

19
channels/chan_sip.c
View File

@@ -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));

7
include/asterisk/rtp.h
View File

@@ -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);

97
rtp.c
View File

@@ -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)