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Merge in the RTP engine API.

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This API provides a generic way for multiple RTP stacks to be
integrated into Asterisk. Right now there is only one present, res_rtp_asterisk,
which is the existing Asterisk RTP stack. Functionality wise this commit
performs the same as previously. API documentation can be viewed in the
rtp_engine.h header file.

Review: http://reviewboard.digium.com/r/209/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186078 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Joshua Colp
2009-04-02 17:20:52 +00:00
parent 08971ce205
commit 63de834395
24 changed files with 7119 additions and 6130 deletions
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97
channels/chan_unistim.c
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@@ -60,7 +60,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#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 @@ struct unistim_subchannel {
/*! 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 @@ static const struct ast_channel_tech unistim_tech = {
.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 cancel_dial(struct unistimsession *pte)
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 @@ static void start_rtp(struct unistim_subchannel *sub)
/* 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);
ast_rtp_instance_get_local_address(sub->rtp, &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 @@ static void start_rtp(struct unistim_subchannel *sub)
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(ast_rtp_instance_get_codecs(sub->rtp), 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 @@ static int unistim_hangup(struct ast_channel *ast)
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 @@ static int unistim_hangup(struct ast_channel *ast)
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 @@ static int unistim_hangup(struct ast_channel *ast)
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 @@ static struct ast_frame *unistim_rtp_read(const struct ast_channel *ast,
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 @@ static int unistim_write(struct ast_channel *ast, struct ast_frame *frame)
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);
}
@@ -4455,8 +4454,8 @@ static struct ast_channel *unistim_new(struct unistim_subchannel *sub, int state
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);
@@ -5526,51 +5525,19 @@ static int reload_config(void)
return 0;
}
static enum ast_rtp_get_result unistim_get_vrtp_peer(struct ast_channel *chan,
struct ast_rtp **rtp)
static enum ast_rtp_glue_result unistim_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
{
return AST_RTP_TRY_NATIVE;
struct unistim_subchannel *sub = chan->tech_pvt;
ao2_ref(sub->rtp, +1);
*instance = sub->rtp;
return AST_RTP_GLUE_RESULT_LOCAL;
}
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 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 ---*/
@@ -5603,7 +5570,7 @@ int load_module(void)
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));
@@ -5634,7 +5601,7 @@ static int unload_module(void)
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)) {