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Major RTP fixes for using inbound SDP on outbound connection, get rid of

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old local rtp stuff...


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7551 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer
2005-12-20 17:52:31 +00:00
parent 93d1d0dcfb
commit 0d32a85be1
7 changed files with 151 additions and 47 deletions
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127
rtp.c
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@@ -124,7 +124,6 @@ struct ast_rtp {
int rtp_lookup_code_cache_isAstFormat;
int rtp_lookup_code_cache_code;
int rtp_lookup_code_cache_result;
int rtp_offered_from_local;
struct ast_rtcp *rtcp;
};
@@ -724,10 +723,98 @@ void ast_rtp_pt_default(struct ast_rtp* rtp)
rtp->rtp_lookup_code_cache_result = 0;
}
static void ast_rtp_pt_copy(struct ast_rtp *dest, struct ast_rtp *src)
{
int i;
/* Copy payload types from source to destination */
for (i=0; i < MAX_RTP_PT; ++i) {
dest->current_RTP_PT[i].isAstFormat =
src->current_RTP_PT[i].isAstFormat;
dest->current_RTP_PT[i].code =
src->current_RTP_PT[i].code;
}
dest->rtp_lookup_code_cache_isAstFormat = 0;
dest->rtp_lookup_code_cache_code = 0;
dest->rtp_lookup_code_cache_result = 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) {
return cur;
}
cur = cur->next;
}
return NULL;
}
int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src)
{
struct ast_rtp *destp, *srcp; /* Audio RTP Channels */
struct ast_rtp *vdestp, *vsrcp; /* Video RTP channels */
struct ast_rtp_protocol *destpr, *srcpr;
/* Lock channels */
ast_mutex_lock(&dest->lock);
while(ast_mutex_trylock(&src->lock)) {
ast_mutex_unlock(&dest->lock);
usleep(1);
ast_mutex_lock(&dest->lock);
}
/* Find channel driver interfaces */
destpr = get_proto(dest);
srcpr = get_proto(src);
if (!destpr) {
ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name);
ast_mutex_unlock(&dest->lock);
ast_mutex_unlock(&src->lock);
return 0;
}
if (!srcpr) {
ast_log(LOG_WARNING, "Channel '%s' has no RTP, not doing anything\n", src->name);
ast_mutex_unlock(&dest->lock);
ast_mutex_unlock(&src->lock);
return 0;
}
/* Get audio and video interface (if native bridge is possible) */
destp = destpr->get_rtp_info(dest);
if (destpr->get_vrtp_info)
vdestp = destpr->get_vrtp_info(dest);
else
vdestp = NULL;
srcp = srcpr->get_rtp_info(src);
if (srcpr->get_vrtp_info)
vsrcp = srcpr->get_vrtp_info(src);
else
vsrcp = NULL;
/* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
if (!destp || !srcp) {
/* Somebody doesn't want to play... */
ast_mutex_unlock(&dest->lock);
ast_mutex_unlock(&src->lock);
return 0;
}
ast_rtp_pt_copy(destp, srcp);
if (vdestp && vsrcp)
ast_rtp_pt_copy(vdestp, vsrcp);
ast_mutex_unlock(&dest->lock);
ast_mutex_unlock(&src->lock);
ast_log(LOG_DEBUG, "Seeded SDP of '%s' with that of '%s'\n", dest->name, src->name);
return 1;
}
/* Make a note of a RTP paymoad type that was seen in a SDP "m=" line. */
/* By default, use the well-known value for this type (although it may */
/* still be set to a different value by a subsequent "a=rtpmap:" line): */
void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt) {
void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt)
{
if (pt < 0 || pt > MAX_RTP_PT)
return; /* bogus payload type */
@@ -739,7 +826,9 @@ void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt) {
/* Make a note of a RTP payload type (with MIME type) that was seen in */
/* a SDP "a=rtpmap:" line. */
void ast_rtp_set_rtpmap_type(struct ast_rtp* rtp, int pt,
char* mimeType, char* mimeSubtype) {
char* mimeType, char* mimeSubtype)
{
int i;
if (pt < 0 || pt > MAX_RTP_PT)
@@ -770,13 +859,6 @@ void ast_rtp_get_current_formats(struct ast_rtp* rtp,
}
}
void ast_rtp_offered_from_local(struct ast_rtp* rtp, int local) {
if (rtp)
rtp->rtp_offered_from_local = local;
else
ast_log(LOG_WARNING, "rtp structure is null\n");
}
struct rtpPayloadType ast_rtp_lookup_pt(struct ast_rtp* rtp, int pt)
{
struct rtpPayloadType result;
@@ -786,8 +868,7 @@ struct rtpPayloadType ast_rtp_lookup_pt(struct ast_rtp* rtp, int pt)
return result; /* bogus payload type */
/* Start with the negotiated codecs */
if (!rtp->rtp_offered_from_local)
result = rtp->current_RTP_PT[pt];
result = rtp->current_RTP_PT[pt];
/* If it doesn't exist, check our static RTP type list, just in case */
if (!result.code)
@@ -829,7 +910,8 @@ int ast_rtp_lookup_code(struct ast_rtp* rtp, const int isAstFormat, const int co
return -1;
}
char* ast_rtp_lookup_mime_subtype(const int isAstFormat, const int code) {
char* ast_rtp_lookup_mime_subtype(const int isAstFormat, const int code)
{
int i;
@@ -1485,21 +1567,6 @@ 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) {
return cur;
}
cur = cur->next;
}
return NULL;
}
/* ast_rtp_bridge: Bridge calls. If possible and allowed, initiate
re-invite so the peers exchange media directly outside
of Asterisk. */
@@ -1698,11 +1765,11 @@ enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel
*rc = who;
if (option_debug)
ast_log(LOG_DEBUG, "Oooh, got a %s\n", f ? "digit" : "hangup");
if ((c0->tech_pvt == pvt0) && (!c0->_softhangup)) {
if ((c0->tech_pvt == pvt0)) {
if (pr0->set_rtp_peer(c0, NULL, NULL, 0, 0))
ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name);
}
if ((c1->tech_pvt == pvt1) && (!c1->_softhangup)) {
if ((c1->tech_pvt == pvt1)) {
if (pr1->set_rtp_peer(c1, NULL, NULL, 0, 0))
ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name);
}