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Improve RTP comments (bug #4792 with mods)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.2-netsec
Mark Spencer 20 years ago
parent 11258c02cb
commit a384e30271
1 changed files with 29 additions and 8 deletions
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35
rtp.c
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@ -1370,6 +1370,7 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
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 @@ void ast_rtp_proto_unregister(struct ast_rtp_protocol *proto)
}
}
/*--- 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 @@ int ast_rtp_proto_register(struct ast_rtp_protocol *proto)
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 @@ int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, st
{
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 @@ int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, st
/* 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 @@ int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, st
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 @@ int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, st
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 @@ int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, st
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 @@ int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, st
}
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 @@ int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, st
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 @@ int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, st
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;
@ -1749,6 +1769,7 @@ void ast_rtp_reload(void)
}
/*--- ast_rtp_init: Initialize the RTP system in Asterisk */
void ast_rtp_init(void)
{
ast_cli_register(&cli_debug);

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