[svn-commits] file: branch file/rtp_engine-mark2 r183691 - /team/file/rtp_engine-mark2/chan...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Mar 23 09:28:35 CDT 2009
Author: file
Date: Mon Mar 23 09:28:31 2009
New Revision: 183691
URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=183691
Log:
Move chan_unistim to the RTP engine architecture. A side effect of this is that this module can now participate in a local RTP bridge.
Modified:
team/file/rtp_engine-mark2/channels/chan_unistim.c
Modified: team/file/rtp_engine-mark2/channels/chan_unistim.c
URL: http://svn.digium.com/svn-view/asterisk/team/file/rtp_engine-mark2/channels/chan_unistim.c?view=diff&rev=183691&r1=183690&r2=183691
==============================================================================
--- team/file/rtp_engine-mark2/channels/chan_unistim.c (original)
+++ team/file/rtp_engine-mark2/channels/chan_unistim.c Mon Mar 23 09:28:31 2009
@@ -60,7 +60,7 @@
#include "asterisk/module.h"
#include "asterisk/pbx.h"
#include "asterisk/event.h"
-#include "asterisk/rtp.h"
+#include "asterisk/rtp_engine.h"
#include "asterisk/netsock.h"
#include "asterisk/acl.h"
#include "asterisk/callerid.h"
@@ -365,7 +365,7 @@
/*! Unistim line */
struct unistim_line *parent;
/*! RTP handle */
- struct ast_rtp *rtp;
+ struct ast_rtp_instance *rtp;
int alreadygone;
char ringvolume;
char ringstyle;
@@ -711,7 +711,7 @@
.send_digit_begin = unistim_senddigit_begin,
.send_digit_end = unistim_senddigit_end,
.send_text = unistim_sendtext,
-/* .bridge = ast_rtp_bridge, */
+ .bridge = ast_rtp_instance_bridge,
};
static void display_last_error(const char *sz_msg)
@@ -1854,7 +1854,7 @@
static void swap_subs(struct unistim_line *p, int a, int b)
{
/* struct ast_channel *towner; */
- struct ast_rtp *rtp;
+ struct ast_rtp_instance *rtp;
int fds;
if (unistimdebug)
@@ -2056,30 +2056,29 @@
/* Allocate the RTP */
if (unistimdebug)
ast_verb(0, "Starting RTP. Bind on %s\n", ast_inet_ntoa(sout.sin_addr));
- sub->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, sout.sin_addr);
+ sub->rtp = ast_rtp_instance_new(NULL, sched, &sout, NULL);
if (!sub->rtp) {
ast_log(LOG_WARNING, "Unable to create RTP session: %s binaddr=%s\n",
strerror(errno), ast_inet_ntoa(sout.sin_addr));
ast_mutex_unlock(&sub->lock);
return;
}
- if (sub->rtp && sub->owner) {
- sub->owner->fds[0] = ast_rtp_fd(sub->rtp);
- sub->owner->fds[1] = ast_rtcp_fd(sub->rtp);
- }
- if (sub->rtp) {
- ast_rtp_setqos(sub->rtp, qos.tos_audio, qos.cos_audio, "UNISTIM RTP");
- ast_rtp_setnat(sub->rtp, sub->parent->parent->nat);
- }
+ ast_rtp_instance_set_prop(sub->rtp, AST_RTP_PROPERTY_RTCP, 1);
+ if (sub->owner) {
+ sub->owner->fds[0] = ast_rtp_instance_fd(sub->rtp, 0);
+ sub->owner->fds[1] = ast_rtp_instance_fd(sub->rtp, 1);
+ }
+ ast_rtp_instance_set_qos(sub->rtp, qos.tos_audio, qos.cos_audio, "UNISTIM RTP");
+ ast_rtp_instance_set_prop(sub->rtp, AST_RTP_PROPERTY_NAT, sub->parent->parent->nat);
/* Create the RTP connection */
- ast_rtp_get_us(sub->rtp, &us);
+ memcpy(&us, &sub->rtp->local_address, sizeof(us));
sin.sin_family = AF_INET;
/* Setting up RTP for our side */
memcpy(&sin.sin_addr, &sub->parent->parent->session->sin.sin_addr,
sizeof(sin.sin_addr));
sin.sin_port = htons(sub->parent->parent->rtp_port);
- ast_rtp_set_peer(sub->rtp, &sin);
+ ast_rtp_instance_set_remote_address(sub->rtp, &sin);
if (!(sub->owner->nativeformats & sub->owner->readformat)) {
int fmt;
fmt = ast_best_codec(sub->owner->nativeformats);
@@ -2091,7 +2090,7 @@
sub->owner->readformat = fmt;
sub->owner->writeformat = fmt;
}
- codec = ast_rtp_lookup_code(sub->rtp, 1, sub->owner->readformat);
+ codec = ast_rtp_codecs_payload_code(&sub->rtp->codecs, 1, sub->owner->readformat);
/* Setting up RTP of the phone */
if (public_ip.sin_family == 0) /* NAT IP override ? */
memcpy(&public, &us, sizeof(public)); /* No defined, using IP from recvmsg */
@@ -3724,7 +3723,7 @@
if (sub->rtp) {
if (unistimdebug)
ast_verb(0, "Destroying RTP session\n");
- ast_rtp_destroy(sub->rtp);
+ ast_rtp_instance_destroy(sub->rtp);
sub->rtp = NULL;
}
return 0;
@@ -3769,7 +3768,7 @@
if (sub->rtp) {
if (unistimdebug)
ast_verb(0, "Destroying RTP session\n");
- ast_rtp_destroy(sub->rtp);
+ ast_rtp_instance_destroy(sub->rtp);
sub->rtp = NULL;
}
return 0;
@@ -3794,7 +3793,7 @@
if (sub->rtp) {
if (unistimdebug)
ast_verb(0, "Destroying RTP session\n");
- ast_rtp_destroy(sub->rtp);
+ ast_rtp_instance_destroy(sub->rtp);
sub->rtp = NULL;
} else if (unistimdebug)
ast_verb(0, "No RTP session to destroy\n");
@@ -3921,10 +3920,10 @@
switch (ast->fdno) {
case 0:
- f = ast_rtp_read(sub->rtp); /* RTP Audio */
+ f = ast_rtp_instance_read(sub->rtp, 0); /* RTP Audio */
break;
case 1:
- f = ast_rtcp_read(sub->rtp); /* RTCP Control Channel */
+ f = ast_rtp_instance_read(sub->rtp, 1); /* RTCP Control Channel */
break;
default:
f = &ast_null_frame;
@@ -3990,7 +3989,7 @@
if (sub) {
ast_mutex_lock(&sub->lock);
if (sub->rtp) {
- res = ast_rtp_write(sub->rtp, frame);
+ res = ast_rtp_instance_write(sub->rtp, frame);
}
ast_mutex_unlock(&sub->lock);
}
@@ -4456,8 +4455,8 @@
if ((sub->rtp) && (sub->subtype == 0)) {
if (unistimdebug)
ast_verb(0, "New unistim channel with a previous rtp handle ?\n");
- tmp->fds[0] = ast_rtp_fd(sub->rtp);
- tmp->fds[1] = ast_rtcp_fd(sub->rtp);
+ tmp->fds[0] = ast_rtp_instance_fd(sub->rtp, 0);
+ tmp->fds[1] = ast_rtp_instance_fd(sub->rtp, 1);
}
if (sub->rtp)
ast_jb_configure(tmp, &global_jbconf);
@@ -5527,51 +5526,18 @@
return 0;
}
-static enum ast_rtp_get_result unistim_get_vrtp_peer(struct ast_channel *chan,
- struct ast_rtp **rtp)
-{
- return AST_RTP_TRY_NATIVE;
-}
-
-static enum ast_rtp_get_result unistim_get_rtp_peer(struct ast_channel *chan,
- struct ast_rtp **rtp)
-{
- struct unistim_subchannel *sub;
- enum ast_rtp_get_result res = AST_RTP_GET_FAILED;
-
- if (unistimdebug)
- ast_verb(0, "unistim_get_rtp_peer called\n");
-
- sub = chan->tech_pvt;
- if (sub && sub->rtp) {
- *rtp = sub->rtp;
- res = AST_RTP_TRY_NATIVE;
- }
-
- return res;
-}
-
-static int unistim_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp,
- struct ast_rtp *vrtp, struct ast_rtp *trtp, int codecs, int nat_active)
-{
- struct unistim_subchannel *sub;
-
- if (unistimdebug)
- ast_verb(0, "unistim_set_rtp_peer called\n");
-
- sub = chan->tech_pvt;
-
- if (sub)
- return 0;
-
- return -1;
-}
-
-static struct ast_rtp_protocol unistim_rtp = {
+static enum ast_rtp_glue_result unistim_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
+{
+ struct unistim_subchannel *sub = chan->tech_pvt;
+
+ *instance = sub->rtp;
+
+ return AST_RTP_GLUE_RESULT_LOCAL;
+}
+
+static struct ast_rtp_glue unistim_rtp_glue = {
.type = channel_type,
.get_rtp_info = unistim_get_rtp_peer,
- .get_vrtp_info = unistim_get_vrtp_peer,
- .set_rtp_peer = unistim_set_rtp_peer,
};
/*--- load_module: PBX load module - initialization ---*/
@@ -5604,7 +5570,7 @@
goto chanreg_failed;
}
- ast_rtp_proto_register(&unistim_rtp);
+ ast_rtp_glue_register(&unistim_rtp_glue);
ast_cli_register_multiple(unistim_cli, ARRAY_LEN(unistim_cli));
@@ -5635,7 +5601,7 @@
ast_cli_unregister_multiple(unistim_cli, ARRAY_LEN(unistim_cli));
ast_channel_unregister(&unistim_tech);
- ast_rtp_proto_unregister(&unistim_rtp);
+ ast_rtp_glue_unregister(&unistim_rtp_glue);
ast_mutex_lock(&monlock);
if (monitor_thread && (monitor_thread != AST_PTHREADT_STOP) && (monitor_thread != AST_PTHREADT_NULL)) {
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