[svn-commits] file: branch file/rtp_engine-mark2 r183696 - /team/file/rtp_engine-mark2/chan...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Mar 23 11:11:55 CDT 2009
Author: file
Date: Mon Mar 23 11:11:52 2009
New Revision: 183696
URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=183696
Log:
Convert chan_mgcp to using the RTP engine architecture.
Modified:
team/file/rtp_engine-mark2/channels/chan_mgcp.c
Modified: team/file/rtp_engine-mark2/channels/chan_mgcp.c
URL: http://svn.digium.com/svn-view/asterisk/team/file/rtp_engine-mark2/channels/chan_mgcp.c?view=diff&rev=183696&r1=183695&r2=183696
==============================================================================
--- team/file/rtp_engine-mark2/channels/chan_mgcp.c (original)
+++ team/file/rtp_engine-mark2/channels/chan_mgcp.c Mon Mar 23 11:11:52 2009
@@ -52,7 +52,7 @@
#include "asterisk/pbx.h"
#include "asterisk/sched.h"
#include "asterisk/io.h"
-#include "asterisk/rtp.h"
+#include "asterisk/rtp_engine.h"
#include "asterisk/acl.h"
#include "asterisk/callerid.h"
#include "asterisk/cli.h"
@@ -282,7 +282,7 @@
int id;
struct ast_channel *owner;
struct mgcp_endpoint *parent;
- struct ast_rtp *rtp;
+ struct ast_rtp_instance *rtp;
struct sockaddr_in tmpdest;
char txident[80]; /*! \todo FIXME txident is replaced by rqnt_ident in endpoint.
This should be obsoleted */
@@ -408,7 +408,7 @@
static int transmit_notify_request(struct mgcp_subchannel *sub, char *tone);
static int transmit_modify_request(struct mgcp_subchannel *sub);
static int transmit_notify_request_with_callerid(struct mgcp_subchannel *sub, char *tone, char *callernum, char *callername);
-static int transmit_modify_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp *rtp, int codecs);
+static int transmit_modify_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp_instance *rtp, int codecs);
static int transmit_connection_del(struct mgcp_subchannel *sub);
static int transmit_audit_endpoint(struct mgcp_endpoint *p);
static void start_rtp(struct mgcp_subchannel *sub);
@@ -447,7 +447,7 @@
.fixup = mgcp_fixup,
.send_digit_begin = mgcp_senddigit_begin,
.send_digit_end = mgcp_senddigit_end,
- .bridge = ast_rtp_bridge,
+ .bridge = ast_rtp_instance_bridge,
};
static void mwi_event_cb(const struct ast_event *event, void *userdata)
@@ -504,7 +504,7 @@
sub->alreadygone = 0;
memset(&sub->tmpdest, 0, sizeof(sub->tmpdest));
if (sub->rtp) {
- ast_rtp_destroy(sub->rtp);
+ ast_rtp_instance_destroy(sub->rtp);
sub->rtp = NULL;
}
dump_cmd_queues(NULL, sub); /* SC */
@@ -1004,7 +1004,7 @@
/* Reset temporary destination */
memset(&sub->tmpdest, 0, sizeof(sub->tmpdest));
if (sub->rtp) {
- ast_rtp_destroy(sub->rtp);
+ ast_rtp_instance_destroy(sub->rtp);
sub->rtp = NULL;
}
@@ -1204,7 +1204,7 @@
/* Retrieve audio/etc from channel. Assumes sub->lock is already held. */
struct ast_frame *f;
- f = ast_rtp_read(sub->rtp);
+ f = ast_rtp_instance_read(sub->rtp, 0);
/* Don't send RFC2833 if we're not supposed to */
if (f && (f->frametype == AST_FRAME_DTMF) && !(sub->parent->dtmfmode & MGCP_DTMF_RFC2833))
return &ast_null_frame;
@@ -1262,7 +1262,7 @@
ast_mutex_lock(&sub->lock);
if ((sub->parent->sub == sub) || !sub->parent->singlepath) {
if (sub->rtp) {
- res = ast_rtp_write(sub->rtp, frame);
+ res = ast_rtp_instance_write(sub->rtp, frame);
}
}
ast_mutex_unlock(&sub->lock);
@@ -1298,7 +1298,7 @@
res = -1; /* Let asterisk play inband indications */
} else if (p->dtmfmode & MGCP_DTMF_RFC2833) {
ast_log(LOG_DEBUG, "Sending DTMF using RFC2833");
- ast_rtp_senddigit_begin(sub->rtp, digit);
+ ast_rtp_instance_dtmf_begin(sub->rtp, digit);
} else {
ast_log(LOG_ERROR, "Don't know about DTMF_MODE %d\n", p->dtmfmode);
}
@@ -1325,7 +1325,7 @@
tmp[2] = digit;
tmp[3] = '\0';
transmit_notify_request(sub, tmp);
- ast_rtp_senddigit_end(sub->rtp, digit);
+ ast_rtp_instance_dtmf_end(sub->rtp, digit);
} else {
ast_log(LOG_ERROR, "Don't know about DTMF_MODE %d\n", p->dtmfmode);
}
@@ -1454,7 +1454,7 @@
ast_moh_stop(ast);
break;
case AST_CONTROL_SRCUPDATE:
- ast_rtp_new_source(sub->rtp);
+ ast_rtp_instance_new_source(sub->rtp);
break;
case -1:
transmit_notify_request(sub, "");
@@ -1482,7 +1482,7 @@
fmt = ast_best_codec(tmp->nativeformats);
ast_string_field_build(tmp, name, "MGCP/%s@%s-%d", i->name, i->parent->name, sub->id);
if (sub->rtp)
- ast_channel_set_fd(tmp, 0, ast_rtp_fd(sub->rtp));
+ ast_channel_set_fd(tmp, 0, ast_rtp_instance_fd(sub->rtp, 0));
if (i->dtmfmode & (MGCP_DTMF_INBAND | MGCP_DTMF_HYBRID)) {
i->dsp = ast_dsp_new();
ast_dsp_set_features(i->dsp, DSP_FEATURE_DIGIT_DETECT);
@@ -1875,12 +1875,12 @@
sin.sin_family = AF_INET;
memcpy(&sin.sin_addr, hp->h_addr, sizeof(sin.sin_addr));
sin.sin_port = htons(portno);
- ast_rtp_set_peer(sub->rtp, &sin);
+ ast_rtp_instance_set_remote_address(sub->rtp, &sin);
#if 0
printf("Peer RTP is at port %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
#endif
/* Scan through the RTP payload types specified in a "m=" line: */
- ast_rtp_pt_clear(sub->rtp);
+ ast_rtp_codecs_payloads_clear(&sub->rtp->codecs, sub->rtp);
codecs = ast_strdupa(m + len);
while (!ast_strlen_zero(codecs)) {
if (sscanf(codecs, "%d%n", &codec, &len) != 1) {
@@ -1889,7 +1889,7 @@
ast_log(LOG_WARNING, "Error in codec string '%s' at '%s'\n", m, codecs);
return -1;
}
- ast_rtp_set_m_type(sub->rtp, codec);
+ ast_rtp_codecs_payloads_set_m_type(&sub->rtp->codecs, sub->rtp, codec);
codec_count++;
codecs += len;
}
@@ -1902,11 +1902,11 @@
if (sscanf(a, "rtpmap: %u %[^/]/", &codec, mimeSubtype) != 2)
continue;
/* Note: should really look at the 'freq' and '#chans' params too */
- ast_rtp_set_rtpmap_type(sub->rtp, codec, "audio", mimeSubtype, 0);
+ ast_rtp_codecs_payloads_set_rtpmap_type(&sub->rtp->codecs, sub->rtp, codec, "audio", mimeSubtype, 0);
}
/* Now gather all of the codecs that were asked for: */
- ast_rtp_get_current_formats(sub->rtp, &peercapability, &peerNonCodecCapability);
+ ast_rtp_codecs_payload_formats(&sub->rtp->codecs, &peercapability, &peerNonCodecCapability);
p->capability = capability & peercapability;
if (mgcpdebug) {
ast_verbose("Capabilities: us - %d, them - %d, combined - %d\n",
@@ -2044,7 +2044,7 @@
}
-static int add_sdp(struct mgcp_request *resp, struct mgcp_subchannel *sub, struct ast_rtp *rtp)
+static int add_sdp(struct mgcp_request *resp, struct mgcp_subchannel *sub, struct ast_rtp_instance *rtp)
{
int len;
int codec;
@@ -2067,9 +2067,9 @@
ast_log(LOG_WARNING, "No way to add SDP without an RTP structure\n");
return -1;
}
- ast_rtp_get_us(sub->rtp, &sin);
+ memcpy(&sin, &sub->rtp->local_address, sizeof(sin));
if (rtp) {
- ast_rtp_get_peer(rtp, &dest);
+ memcpy(&dest, &sub->rtp->remote_address, sizeof(dest));
} else {
if (sub->tmpdest.sin_addr.s_addr) {
dest.sin_addr = sub->tmpdest.sin_addr;
@@ -2095,11 +2095,11 @@
if (mgcpdebug) {
ast_verbose("Answering with capability %d\n", x);
}
- codec = ast_rtp_lookup_code(sub->rtp, 1, x);
+ codec = ast_rtp_codecs_payload_code(&sub->rtp->codecs, 1, x);
if (codec > -1) {
snprintf(costr, sizeof(costr), " %d", codec);
strncat(m, costr, sizeof(m) - strlen(m) - 1);
- snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype(1, x, 0));
+ snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype2(1, x, 0));
strncat(a, costr, sizeof(a) - strlen(a) - 1);
}
}
@@ -2109,11 +2109,11 @@
if (mgcpdebug) {
ast_verbose("Answering with non-codec capability %d\n", x);
}
- codec = ast_rtp_lookup_code(sub->rtp, 0, x);
+ codec = ast_rtp_codecs_payload_code(&sub->rtp->codecs, 0, x);
if (codec > -1) {
snprintf(costr, sizeof(costr), " %d", codec);
strncat(m, costr, sizeof(m) - strlen(m) - 1);
- snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype(0, x, 0));
+ snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype2(0, x, 0));
strncat(a, costr, sizeof(a) - strlen(a) - 1);
if (x == AST_RTP_DTMF) {
/* Indicate we support DTMF... Not sure about 16,
@@ -2137,7 +2137,7 @@
return 0;
}
-static int transmit_modify_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp *rtp, int codecs)
+static int transmit_modify_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp_instance *rtp, int codecs)
{
struct mgcp_request resp;
char local[256];
@@ -2148,13 +2148,13 @@
if (ast_strlen_zero(sub->cxident) && rtp) {
/* We don't have a CXident yet, store the destination and
wait a bit */
- ast_rtp_get_peer(rtp, &sub->tmpdest);
+ memcpy(&sub->tmpdest, &rtp->remote_address, sizeof(sub->tmpdest));
return 0;
}
ast_copy_string(local, "p:20", sizeof(local));
for (x = 1; x <= AST_FORMAT_AUDIO_MASK; x <<= 1) {
if (p->capability & x) {
- snprintf(tmp, sizeof(tmp), ", a:%s", ast_rtp_lookup_mime_subtype(1, x, 0));
+ snprintf(tmp, sizeof(tmp), ", a:%s", ast_rtp_lookup_mime_subtype2(1, x, 0));
strncat(local, tmp, sizeof(local) - strlen(local) - 1);
}
}
@@ -2173,7 +2173,7 @@
return send_request(p, sub, &resp, oseq); /* SC */
}
-static int transmit_connect_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp *rtp)
+static int transmit_connect_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp_instance *rtp)
{
struct mgcp_request resp;
char local[256];
@@ -2184,7 +2184,7 @@
ast_copy_string(local, "p:20", sizeof(local));
for (x = 1; x <= AST_FORMAT_AUDIO_MASK; x <<= 1) {
if (p->capability & x) {
- snprintf(tmp, sizeof(tmp), ", a:%s", ast_rtp_lookup_mime_subtype(1, x, 0));
+ snprintf(tmp, sizeof(tmp), ", a:%s", ast_rtp_lookup_mime_subtype2(1, x, 0));
strncat(local, tmp, sizeof(local) - strlen(local) - 1);
}
}
@@ -2612,21 +2612,17 @@
ast_mutex_lock(&sub->lock);
/* check again to be on the safe side */
if (sub->rtp) {
- ast_rtp_destroy(sub->rtp);
+ ast_rtp_instance_destroy(sub->rtp);
sub->rtp = NULL;
}
/* Allocate the RTP now */
- sub->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
+ sub->rtp = ast_rtp_instance_new(NULL, sched, &bindaddr, NULL);
if (sub->rtp && sub->owner)
- ast_channel_set_fd(sub->owner, 0, ast_rtp_fd(sub->rtp));
+ ast_channel_set_fd(sub->owner, 0, ast_rtp_instance_fd(sub->rtp, 0));
if (sub->rtp) {
- ast_rtp_setqos(sub->rtp, qos.tos_audio, qos.cos_audio, "MGCP RTP");
- ast_rtp_setnat(sub->rtp, sub->nat);
- }
-#if 0
- ast_rtp_set_callback(p->rtp, rtpready);
- ast_rtp_set_data(p->rtp, p);
-#endif
+ ast_rtp_instance_set_qos(sub->rtp, qos.tos_audio, qos.cos_audio, "MGCP RTP");
+ ast_rtp_instance_set_prop(sub->rtp, AST_RTP_PROPERTY_NAT, sub->nat);
+ }
/* Make a call*ID */
snprintf(sub->callid, sizeof(sub->callid), "%08lx%s", ast_random(), sub->txident);
/* Transmit the connection create */
@@ -3941,22 +3937,22 @@
return (gw_reload ? NULL : gw);
}
-static enum ast_rtp_get_result mgcp_get_rtp_peer(struct ast_channel *chan, struct ast_rtp **rtp)
+static enum ast_rtp_glue_result mgcp_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
{
struct mgcp_subchannel *sub = NULL;
if (!(sub = chan->tech_pvt) || !(sub->rtp))
- return AST_RTP_GET_FAILED;
-
- *rtp = sub->rtp;
+ return AST_RTP_GLUE_RESULT_FORBID;
+
+ *instance = sub->rtp;
if (sub->parent->canreinvite)
- return AST_RTP_TRY_NATIVE;
+ return AST_RTP_GLUE_RESULT_REMOTE;
else
- return AST_RTP_TRY_PARTIAL;
-}
-
-static int mgcp_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, struct ast_rtp *trtp, int codecs, int nat_active)
+ return AST_RTP_GLUE_RESULT_LOCAL;
+}
+
+static int mgcp_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *rtp, struct ast_rtp_instance *vrtp, struct ast_rtp_instance *trtp, int codecs, int nat_active)
{
/* XXX Is there such thing as video support with MGCP? XXX */
struct mgcp_subchannel *sub;
@@ -3968,10 +3964,10 @@
return -1;
}
-static struct ast_rtp_protocol mgcp_rtp = {
+static struct ast_rtp_glue mgcp_rtp_glue = {
.type = "MGCP",
.get_rtp_info = mgcp_get_rtp_peer,
- .set_rtp_peer = mgcp_set_rtp_peer,
+ .update_peer = mgcp_set_rtp_peer,
};
static void destroy_endpoint(struct mgcp_endpoint *e)
@@ -3985,7 +3981,7 @@
transmit_connection_del(sub);
}
if (sub->rtp) {
- ast_rtp_destroy(sub->rtp);
+ ast_rtp_instance_destroy(sub->rtp);
sub->rtp = NULL;
}
memset(sub->magic, 0, sizeof(sub->magic));
@@ -4277,7 +4273,7 @@
return AST_MODULE_LOAD_FAILURE;
}
- ast_rtp_proto_register(&mgcp_rtp);
+ ast_rtp_glue_register(&mgcp_rtp_glue);
ast_cli_register_multiple(cli_mgcp, sizeof(cli_mgcp) / sizeof(struct ast_cli_entry));
/* And start the monitor for the first time */
@@ -4380,7 +4376,7 @@
}
close(mgcpsock);
- ast_rtp_proto_unregister(&mgcp_rtp);
+ ast_rtp_glue_unregister(&mgcp_rtp_glue);
ast_cli_unregister_multiple(cli_mgcp, sizeof(cli_mgcp) / sizeof(struct ast_cli_entry));
sched_context_destroy(sched);
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