Update of /usr/cvsroot/asterisk
In directory mongoose.digium.com:/tmp/cvs-serv22143
Modified Files:
rtp.c
Log Message:
Improve RTP comments (bug #4792 with mods)
Index: rtp.c
===================================================================
RCS file: /usr/cvsroot/asterisk/rtp.c,v
retrieving revision 1.139
retrieving revision 1.140
diff -u -d -r1.139 -r1.140
--- rtp.c 15 Jul 2005 23:00:46 -0000 1.139
+++ rtp.c 3 Aug 2005 05:00:42 -0000 1.140
@@ -1370,6 +1370,7 @@
return 0;
}
+/*--- ast_rtp_proto_unregister: Unregister interface to channel driver */
void ast_rtp_proto_unregister(struct ast_rtp_protocol *proto)
{
struct ast_rtp_protocol *cur, *prev;
@@ -1389,6 +1390,7 @@
}
}
+/*--- ast_rtp_proto_register: Register interface to channel driver */
int ast_rtp_proto_register(struct ast_rtp_protocol *proto)
{
struct ast_rtp_protocol *cur;
@@ -1405,9 +1407,11 @@
return 0;
}
+/*--- get_proto: Get channel driver interface structure */
static struct ast_rtp_protocol *get_proto(struct ast_channel *chan)
{
struct ast_rtp_protocol *cur;
+
cur = protos;
while(cur) {
if (cur->type == chan->type) {
@@ -1425,8 +1429,8 @@
{
struct ast_frame *f;
struct ast_channel *who, *cs[3];
- struct ast_rtp *p0, *p1;
- struct ast_rtp *vp0, *vp1;
+ struct ast_rtp *p0, *p1; /* Audio RTP Channels */
+ struct ast_rtp *vp0, *vp1; /* Video RTP channels */
struct ast_rtp_protocol *pr0, *pr1;
struct sockaddr_in ac0, ac1;
struct sockaddr_in vac0, vac1;
@@ -1446,12 +1450,16 @@
/* if need DTMF, cant native bridge */
if (flags & (AST_BRIDGE_DTMF_CHANNEL_0 | AST_BRIDGE_DTMF_CHANNEL_1))
return -2;
+
+ /* Lock channels */
ast_mutex_lock(&c0->lock);
while(ast_mutex_trylock(&c1->lock)) {
ast_mutex_unlock(&c0->lock);
usleep(1);
ast_mutex_lock(&c0->lock);
}
+
+ /* Find channel driver interfaces */
pr0 = get_proto(c0);
pr1 = get_proto(c1);
if (!pr0) {
@@ -1466,8 +1474,12 @@
ast_mutex_unlock(&c1->lock);
return -1;
}
+
+ /* Get channel specific interface structures */
pvt0 = c0->tech_pvt;
pvt1 = c1->tech_pvt;
+
+ /* Get audio and video interface (if native bridge is possible) */
p0 = pr0->get_rtp_info(c0);
if (pr0->get_vrtp_info)
vp0 = pr0->get_vrtp_info(c0);
@@ -1478,12 +1490,15 @@
vp1 = pr1->get_vrtp_info(c1);
else
vp1 = NULL;
+
+ /* Check if bridge is still possible (In SIP canreinvite=no stops this,
like NAT) */
if (!p0 || !p1) {
/* Somebody doesn't want to play... */
ast_mutex_unlock(&c0->lock);
ast_mutex_unlock(&c1->lock);
return -2;
}
+ /* Get codecs from both sides */
if (pr0->get_codec)
codec0 = pr0->get_codec(c0);
else
@@ -1502,6 +1517,7 @@
return -2;
}
}
+
/* Ok, we should be able to redirect the media. Start with one channel
*/
if (pr0->set_rtp_peer(c0, p1, vp1, codec1))
ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n",
c0->name, c1->name);
@@ -1522,28 +1538,32 @@
}
ast_mutex_unlock(&c0->lock);
ast_mutex_unlock(&c1->lock);
+ /* External RTP Bridge up, now loop and see if something happes that
force us to take the
+ media back to Asterisk */
cs[0] = c0;
cs[1] = c1;
cs[2] = NULL;
oldcodec0 = codec0;
oldcodec1 = codec1;
for (;;) {
+ /* Check if something changed... */
if ((c0->tech_pvt != pvt0) ||
(c1->tech_pvt != pvt1) ||
(c0->masq || c0->masqr || c1->masq || c1->masqr)) {
ast_log(LOG_DEBUG, "Oooh, something is weird,
backing out\n");
if (c0->tech_pvt == pvt0) {
if (pr0->set_rtp_peer(c0, NULL, NULL,
0))
- ast_log(LOG_WARNING, "Channel
'%s' failed to revert\n", c0->name);
+ ast_log(LOG_WARNING, "Channel
'%s' failed to break RTP bridge\n", c0->name);
}
if (c1->tech_pvt == pvt1) {
if (pr1->set_rtp_peer(c1, NULL, NULL,
0))
- ast_log(LOG_WARNING, "Channel
'%s' failed to revert back\n", c1->name);
+ ast_log(LOG_WARNING, "Channel
'%s' failed to break RTP bridge\n", c1->name);
}
/* Tell it to try again later */
return -3;
}
to = -1;
+ /* Now check if they have changed address */
ast_rtp_get_peer(p1, &t1);
ast_rtp_get_peer(p0, &t0);
if (pr0->get_codec)
@@ -1555,7 +1575,7 @@
if (vp0)
ast_rtp_get_peer(vp0, &vt0);
if (inaddrcmp(&t1, &ac1) || (vp1 && inaddrcmp(&vt1, &vac1)) ||
(codec1 != oldcodec1)) {
- if (option_debug) {
+ if (option_debug > 1) {
ast_log(LOG_DEBUG, "Oooh, '%s' changed end
address to %s:%d (format %d)\n",
c1->name, ast_inet_ntoa(iabuf,
sizeof(iabuf), t1.sin_addr), ntohs(t1.sin_port), codec1);
ast_log(LOG_DEBUG, "Oooh, '%s' changed end
vaddress to %s:%d (format %d)\n",
@@ -1603,11 +1623,11 @@
ast_log(LOG_DEBUG, "Oooh, got a %s\n", f ?
"digit" : "hangup");
if ((c0->tech_pvt == pvt0) && (!c0->_softhangup)) {
if (pr0->set_rtp_peer(c0, NULL, NULL, 0))
- ast_log(LOG_WARNING, "Channel '%s'
failed to revert\n", c0->name);
+ ast_log(LOG_WARNING, "Channel '%s'
failed to break RTP bridge\n", c0->name);
}
if ((c1->tech_pvt == pvt1) && (!c1->_softhangup)) {
if (pr1->set_rtp_peer(c1, NULL, NULL, 0))
- ast_log(LOG_WARNING, "Channel '%s'
failed to revert back\n", c1->name);
+ ast_log(LOG_WARNING, "Channel '%s'
failed to break RTP bridge\n", c1->name);
}
/* That's all we needed */
return 0;
@@ -1666,7 +1686,7 @@
static int rtp_do_debug(int fd, int argc, char *argv[])
{
- if(argc != 2){
+ if(argc != 2) {
if(argc != 4)
return RESULT_SHOWUSAGE;
return rtp_do_debug_ip(fd, argc, argv);
@@ -1749,6 +1769,7 @@
}
+/*--- ast_rtp_init: Initialize the RTP system in Asterisk */
void ast_rtp_init(void)
{
ast_cli_register(&cli_debug);
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