[asterisk-commits] trunk - r7551 in /trunk: ./ apps/ channels/
include/asterisk/
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Tue Dec 20 11:52:36 CST 2005
Author: markster
Date: Tue Dec 20 11:52:31 2005
New Revision: 7551
URL: http://svn.digium.com/view/asterisk?rev=7551&view=rev
Log:
Major RTP fixes for using inbound SDP on outbound connection, get rid of
old local rtp stuff...
Modified:
trunk/apps/app_dial.c
trunk/channels/chan_mgcp.c
trunk/channels/chan_sip.c
trunk/frame.c
trunk/include/asterisk/frame.h
trunk/include/asterisk/rtp.h
trunk/rtp.c
Modified: trunk/apps/app_dial.c
URL: http://svn.digium.com/view/asterisk/trunk/apps/app_dial.c?rev=7551&r1=7550&r2=7551&view=diff
==============================================================================
--- trunk/apps/app_dial.c (original)
+++ trunk/apps/app_dial.c Tue Dec 20 11:52:31 2005
@@ -53,6 +53,7 @@
#include "asterisk/utils.h"
#include "asterisk/app.h"
#include "asterisk/causes.h"
+#include "asterisk/rtp.h"
#include "asterisk/manager.h"
#include "asterisk/privacy.h"
@@ -310,7 +311,7 @@
} while (0)
-static int onedigit_goto(struct ast_channel *chan, char *context, char exten, int pri)
+static int onedigit_goto(struct ast_channel *chan, const char *context, char exten, int pri)
{
char rexten[2] = { exten, '\0' };
@@ -380,7 +381,7 @@
int pos;
int single;
struct ast_channel *winner;
- char *context = NULL;
+ const char *context = NULL;
char cidname[AST_MAX_EXTENSION];
single = (outgoing && !outgoing->next && !ast_test_flag(outgoing, OPT_MUSICBACK | OPT_RINGBACK));
@@ -475,6 +476,7 @@
ast_clear_flag(o, DIAL_STILLGOING);
HANDLE_CAUSE(cause, in);
} else {
+ ast_rtp_make_compatible(o->chan, in);
if (o->chan->cid.cid_num)
free(o->chan->cid.cid_num);
o->chan->cid.cid_num = NULL;
@@ -744,16 +746,17 @@
long timelimit = 0;
long play_warning = 0;
long warning_freq=0;
- char *warning_sound=NULL;
- char *end_sound=NULL;
- char *start_sound=NULL;
+ const char *warning_sound=NULL;
+ const char *end_sound=NULL;
+ const char *start_sound=NULL;
char *dtmfcalled=NULL, *dtmfcalling=NULL;
- char *var;
+ const char *var;
char status[256];
int play_to_caller=0,play_to_callee=0;
int sentringing=0, moh=0;
- char *outbound_group = NULL;
- char *macro_result = NULL, *macro_transfer_dest = NULL;
+ const char *outbound_group = NULL;
+ const char *macro_result = NULL;
+ char *macro_transfer_dest = NULL;
int digit = 0, result = 0;
time_t start_time, answer_time, end_time;
struct ast_app *app = NULL;
@@ -1052,6 +1055,9 @@
}
}
+ /* Setup outgoing SDP to match incoming one */
+ ast_rtp_make_compatible(tmp->chan, chan);
+
/* Inherit specially named variables from parent channel */
ast_channel_inherit_variables(chan, tmp->chan);
@@ -1190,7 +1196,7 @@
if (peer->name)
pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", peer->name);
- number = pbx_builtin_getvar_helper(peer, "DIALEDPEERNUMBER");
+ number = (char *)pbx_builtin_getvar_helper(peer, "DIALEDPEERNUMBER");
if (!number)
number = numsubst;
pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", number);
@@ -1602,7 +1608,8 @@
static int retrydial_exec(struct ast_channel *chan, void *data)
{
- char *announce = NULL, *context = NULL, *dialdata = NULL;
+ char *announce = NULL, *dialdata = NULL;
+ const char *context = NULL;
int sleep = 0, loops = 0, res = 0;
struct localuser *u;
struct ast_flags peerflags;
Modified: trunk/channels/chan_mgcp.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_mgcp.c?rev=7551&r1=7550&r2=7551&view=diff
==============================================================================
--- trunk/channels/chan_mgcp.c (original)
+++ trunk/channels/chan_mgcp.c Tue Dec 20 11:52:31 2005
@@ -2095,7 +2095,6 @@
add_header(&resp, "X", sub->txident);
add_header(&resp, "I", sub->cxident);
/*add_header(&resp, "S", "");*/
- ast_rtp_offered_from_local(sub->rtp, 0);
add_sdp(&resp, sub, rtp);
/* SC: fill in new fields */
resp.cmd = MGCP_CMD_MDCX;
@@ -2129,7 +2128,6 @@
/* SC: X header should not be sent. kept for compatibility */
add_header(&resp, "X", sub->txident);
/*add_header(&resp, "S", "");*/
- ast_rtp_offered_from_local(sub->rtp, 1);
add_sdp(&resp, sub, rtp);
/* SC: fill in new fields */
resp.cmd = MGCP_CMD_CRCX;
@@ -3948,7 +3946,7 @@
/* XXX Is there such thing as video support with MGCP? XXX */
struct mgcp_subchannel *sub;
sub = chan->tech_pvt;
- if (sub) {
+ if (sub && !sub->alreadygone) {
transmit_modify_with_sdp(sub, rtp, codecs);
return 0;
}
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?rev=7551&r1=7550&r2=7551&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Tue Dec 20 11:52:31 2005
@@ -4509,7 +4509,6 @@
}
respprep(&resp, p, msg, req);
if (p->rtp) {
- ast_rtp_offered_from_local(p->rtp, 0);
add_sdp(&resp, p);
} else {
ast_log(LOG_ERROR, "Can't add SDP to response, since we have no RTP session allocated. Call-ID %s\n", p->callid);
@@ -4583,7 +4582,6 @@
add_header(&req, "Allow", ALLOWED_METHODS);
if (sipdebug)
add_header(&req, "X-asterisk-info", "SIP re-invite (RTP bridge)");
- ast_rtp_offered_from_local(p->rtp, 1);
add_sdp(&req, p);
/* Use this as the basis */
copy_request(&p->initreq, &req);
@@ -4924,7 +4922,6 @@
}
}
if (sdp && p->rtp) {
- ast_rtp_offered_from_local(p->rtp, 1);
add_sdp(&req, p);
} else {
add_header_contentLength(&req, 0);
@@ -12691,6 +12688,11 @@
if (!p)
return -1;
ast_mutex_lock(&p->lock);
+ if (ast_test_flag(p, SIP_ALREADYGONE)) {
+ /* If we're destroyed, don't bother */
+ ast_mutex_unlock(&p->lock);
+ return 0;
+ }
if (rtp)
ast_rtp_get_peer(rtp, &p->redirip);
else
Modified: trunk/frame.c
URL: http://svn.digium.com/view/asterisk/trunk/frame.c?rev=7551&r1=7550&r2=7551&view=diff
==============================================================================
--- trunk/frame.c (original)
+++ trunk/frame.c Tue Dec 20 11:52:31 2005
@@ -1299,3 +1299,28 @@
return 0;
}
+
+struct ast_frame *ast_frame_enqueue(struct ast_frame *head, struct ast_frame *f, int maxlen, int dupe)
+{
+ struct ast_frame *cur, *oldhead;
+ int len=0;
+ if (f && dupe)
+ f = ast_frdup(f);
+ if (!f)
+ return head;
+
+ f->next = NULL;
+ if (!head)
+ return f;
+ cur = head;
+ while(cur->next) {
+ cur = cur->next;
+ len++;
+ if (len >= maxlen) {
+ oldhead = head;
+ head = head->next;
+ ast_frfree(oldhead);
+ }
+ }
+ return head;
+}
Modified: trunk/include/asterisk/frame.h
URL: http://svn.digium.com/view/asterisk/trunk/include/asterisk/frame.h?rev=7551&r1=7550&r2=7551&view=diff
==============================================================================
--- trunk/include/asterisk/frame.h (original)
+++ trunk/include/asterisk/frame.h Tue Dec 20 11:52:31 2005
@@ -452,6 +452,10 @@
/*! \brief Returns the number of bytes for the number of samples of the given format */
extern int ast_codec_get_len(int format, int samples);
+/*! \brief Appends a frame to the end of a list of frames, truncating the maximum length of the list */
+extern struct ast_frame *ast_frame_enqueue(struct ast_frame *head, struct ast_frame *f, int maxlen, int dupe);
+
+
/*! \brief Gets duration in ms of interpolation frame for a format */
static inline int ast_codec_interp_len(int format)
{
Modified: trunk/include/asterisk/rtp.h
URL: http://svn.digium.com/view/asterisk/trunk/include/asterisk/rtp.h?rev=7551&r1=7550&r2=7551&view=diff
==============================================================================
--- trunk/include/asterisk/rtp.h (original)
+++ trunk/include/asterisk/rtp.h Tue Dec 20 11:52:31 2005
@@ -135,7 +135,6 @@
/* Mapping between RTP payload format codes and Asterisk codes: */
struct rtpPayloadType ast_rtp_lookup_pt(struct ast_rtp* rtp, int pt);
int ast_rtp_lookup_code(struct ast_rtp* rtp, int isAstFormat, int code);
-void ast_rtp_offered_from_local(struct ast_rtp* rtp, int local);
void ast_rtp_get_current_formats(struct ast_rtp* rtp,
int* astFormats, int* nonAstFormats);
@@ -154,6 +153,8 @@
void ast_rtp_proto_unregister(struct ast_rtp_protocol *proto);
+int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src);
+
void ast_rtp_stop(struct ast_rtp *rtp);
void ast_rtp_init(void);
Modified: trunk/rtp.c
URL: http://svn.digium.com/view/asterisk/trunk/rtp.c?rev=7551&r1=7550&r2=7551&view=diff
==============================================================================
--- trunk/rtp.c (original)
+++ trunk/rtp.c Tue Dec 20 11:52:31 2005
@@ -124,7 +124,6 @@
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 @@
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 @@
/* 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_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 @@
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 @@
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;
@@ -1483,21 +1565,6 @@
proto->next = protos;
protos = 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
@@ -1698,11 +1765,11 @@
*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);
}
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