[svn-commits] file: branch file/rtp_engine-mark2 r185703 - in /team/file/rtp_engine-mark2: ...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Mar 31 19:31:45 CDT 2009
Author: file
Date: Tue Mar 31 19:31:38 2009
New Revision: 185703
URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=185703
Log:
Make the ast_rtp_instance structure opaque.
Modified:
team/file/rtp_engine-mark2/UPGRADE.txt
team/file/rtp_engine-mark2/channels/chan_gtalk.c
team/file/rtp_engine-mark2/channels/chan_h323.c
team/file/rtp_engine-mark2/channels/chan_jingle.c
team/file/rtp_engine-mark2/channels/chan_mgcp.c
team/file/rtp_engine-mark2/channels/chan_sip.c
team/file/rtp_engine-mark2/channels/chan_skinny.c
team/file/rtp_engine-mark2/channels/chan_unistim.c
team/file/rtp_engine-mark2/include/asterisk/rtp_engine.h
team/file/rtp_engine-mark2/main/rtp_engine.c
team/file/rtp_engine-mark2/res/res_rtp_asterisk.c
Modified: team/file/rtp_engine-mark2/UPGRADE.txt
URL: http://svn.digium.com/svn-view/asterisk/team/file/rtp_engine-mark2/UPGRADE.txt?view=diff&rev=185703&r1=185702&r2=185703
==============================================================================
--- team/file/rtp_engine-mark2/UPGRADE.txt (original)
+++ team/file/rtp_engine-mark2/UPGRADE.txt Tue Mar 31 19:31:38 2009
@@ -23,7 +23,7 @@
* The usage of RTP inside of Asterisk has now become modularized. This means
the Asterisk RTP stack now exists as a loadable module, res_rtp_asterisk.
If you are not using autoload=yes in modules.conf you will need to ensure
- it is set to load. If not then any module which uses RTP (such as chan_sip)
+ it is set to load. If not, then any module which uses RTP (such as chan_sip)
will not be able to send or receive calls.
From 1.6.1 to 1.6.2:
Modified: team/file/rtp_engine-mark2/channels/chan_gtalk.c
URL: http://svn.digium.com/svn-view/asterisk/team/file/rtp_engine-mark2/channels/chan_gtalk.c?view=diff&rev=185703&r1=185702&r2=185703
==============================================================================
--- team/file/rtp_engine-mark2/channels/chan_gtalk.c (original)
+++ team/file/rtp_engine-mark2/channels/chan_gtalk.c Tue Mar 31 19:31:38 2009
@@ -613,13 +613,13 @@
/* codec points to the first <payload-type/> tag */
codec = iks_first_tag(iks_first_tag(iks_first_tag(pak->x)));
while (codec) {
- ast_rtp_codecs_payloads_set_m_type(&tmp->rtp->codecs, tmp->rtp, atoi(iks_find_attrib(codec, "id")));
- ast_rtp_codecs_payloads_set_rtpmap_type(&tmp->rtp->codecs, tmp->rtp, atoi(iks_find_attrib(codec, "id")), "audio", iks_find_attrib(codec, "name"), 0);
+ ast_rtp_codecs_payloads_set_m_type(ast_rtp_instance_get_codecs(tmp->rtp), tmp->rtp, atoi(iks_find_attrib(codec, "id")));
+ ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(tmp->rtp), tmp->rtp, atoi(iks_find_attrib(codec, "id")), "audio", iks_find_attrib(codec, "name"), 0);
codec = iks_next_tag(codec);
}
/* Now gather all of the codecs that we are asked for */
- ast_rtp_codecs_payload_formats(&tmp->rtp->codecs, &tmp->peercapability, &peernoncodeccapability);
+ ast_rtp_codecs_payload_formats(ast_rtp_instance_get_codecs(tmp->rtp), &tmp->peercapability, &peernoncodeccapability);
/* at this point, we received an awser from the remote Gtalk client,
which allows us to compare capabilities */
@@ -806,7 +806,7 @@
goto safeout;
}
- memcpy(&sin, &p->rtp->local_address, sizeof(sin));
+ ast_rtp_instance_get_local_address(p->rtp, &sin);
ast_find_ourip(&us, bindaddr);
if (!strcmp(ast_inet_ntoa(us), "127.0.0.1")) {
ast_log(LOG_WARNING, "Found a loopback IP on the system, check your network configuration or set the bindaddr attribute.");
@@ -949,7 +949,7 @@
/* clear codecs */
tmp->rtp = ast_rtp_instance_new(NULL, sched, &bindaddr, NULL);
ast_rtp_instance_set_prop(tmp->rtp, AST_RTP_PROPERTY_RTCP, 1);
- ast_rtp_codecs_payloads_clear(&tmp->rtp->codecs, tmp->rtp);
+ ast_rtp_codecs_payloads_clear(ast_rtp_instance_get_codecs(tmp->rtp), tmp->rtp);
/* add user configured codec capabilites */
if (client->capability)
@@ -1011,7 +1011,7 @@
/* Set Frame packetization */
if (i->rtp)
- ast_rtp_codecs_packetization_set(&i->rtp->codecs, i->rtp, &i->prefs);
+ ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(i->rtp), i->rtp, &i->prefs);
tmp->nativeformats = ast_codec_choose(&i->prefs, what, 1) | (i->jointcapability & AST_FORMAT_VIDEO_MASK);
fmt = ast_best_codec(tmp->nativeformats);
@@ -1204,13 +1204,13 @@
codec = iks_first_tag(iks_first_tag(iks_first_tag(pak->x)));
while (codec) {
- ast_rtp_codecs_payloads_set_m_type(&p->rtp->codecs, p->rtp, atoi(iks_find_attrib(codec, "id")));
- ast_rtp_codecs_payloads_set_rtpmap_type(&p->rtp->codecs, p->rtp, atoi(iks_find_attrib(codec, "id")), "audio", iks_find_attrib(codec, "name"), 0);
+ ast_rtp_codecs_payloads_set_m_type(ast_rtp_instance_get_codecs(p->rtp), p->rtp, atoi(iks_find_attrib(codec, "id")));
+ ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(p->rtp), p->rtp, atoi(iks_find_attrib(codec, "id")), "audio", iks_find_attrib(codec, "name"), 0);
codec = iks_next_tag(codec);
}
/* Now gather all of the codecs that we are asked for */
- ast_rtp_codecs_payload_formats(&p->rtp->codecs, &p->peercapability, &peernoncodeccapability);
+ ast_rtp_codecs_payload_formats(ast_rtp_instance_get_codecs(p->rtp), &p->peercapability, &peernoncodeccapability);
p->jointcapability = p->capability & p->peercapability;
ast_mutex_unlock(&p->lock);
@@ -1274,7 +1274,7 @@
p->ourcandidates->username);
/* Find out the result of the STUN */
- memcpy(&aux, &p->rtp->remote_address, sizeof(aux));
+ ast_rtp_instance_get_remote_address(p->rtp, &aux);
/* If the STUN result is different from the IP of the hostname,
lock on the stun IP of the hostname advertised by the
Modified: team/file/rtp_engine-mark2/channels/chan_h323.c
URL: http://svn.digium.com/svn-view/asterisk/team/file/rtp_engine-mark2/channels/chan_h323.c?view=diff&rev=185703&r1=185702&r2=185703
==============================================================================
--- team/file/rtp_engine-mark2/channels/chan_h323.c (original)
+++ team/file/rtp_engine-mark2/channels/chan_h323.c Tue Mar 31 19:31:38 2009
@@ -972,12 +972,12 @@
ast_rtp_instance_set_prop(pvt->rtp, AST_RTP_PROPERTY_NAT, pvt->options.nat);
if (pvt->dtmf_pt[0] > 0)
- ast_rtp_codecs_payloads_set_rtpmap_type(&pvt->rtp->codecs, pvt->rtp, pvt->dtmf_pt[0], "audio", "telephone-event", 0);
+ ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(pvt->rtp), pvt->rtp, pvt->dtmf_pt[0], "audio", "telephone-event", 0);
if (pvt->dtmf_pt[1] > 0)
- ast_rtp_codecs_payloads_set_rtpmap_type(&pvt->rtp->codecs, pvt->rtp, pvt->dtmf_pt[1], "audio", "cisco-telephone-event", 0);
+ ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(pvt->rtp), pvt->rtp, pvt->dtmf_pt[1], "audio", "cisco-telephone-event", 0);
if (pvt->peercapability)
- ast_rtp_codecs_packetization_set(&pvt->rtp->codecs, pvt->rtp, &pvt->peer_prefs);
+ ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(pvt->rtp), pvt->rtp, &pvt->peer_prefs);
if (pvt->owner && !ast_channel_trylock(pvt->owner)) {
ast_jb_configure(pvt->owner, &global_jbconf);
@@ -1912,7 +1912,7 @@
return NULL;
}
/* figure out our local RTP port and tell the H.323 stack about it */
- memcpy(&us, &pvt->rtp->local_address, sizeof(us));
+ ast_rtp_instance_get_local_address(pvt->rtp, &us);
ast_mutex_unlock(&pvt->lock);
ast_copy_string(info->addr, ast_inet_ntoa(us.sin_addr), sizeof(info->addr));
@@ -1952,7 +1952,7 @@
__oh323_rtp_create(pvt);
if ((pt == 2) && (pvt->jointcapability & AST_FORMAT_G726_AAL2)) {
- ast_rtp_codecs_payloads_set_rtpmap_type(&pvt->rtp->codecs, pvt->rtp, pt, "audio", "G726-32", AST_RTP_OPT_G726_NONSTANDARD);
+ ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(pvt->rtp), pvt->rtp, pt, "audio", "G726-32", AST_RTP_OPT_G726_NONSTANDARD);
}
them.sin_family = AF_INET;
@@ -1977,7 +1977,7 @@
/* Change native format to reflect information taken from OLC/OLCAck */
nativeformats_changed = 0;
if (pt != 128 && pvt->rtp) { /* Payload type is invalid, so try to use previously decided */
- struct ast_rtp_payload_type rtptype = ast_rtp_codecs_payload_lookup(&pvt->rtp->codecs, pt);
+ struct ast_rtp_payload_type rtptype = ast_rtp_codecs_payload_lookup(ast_rtp_instance_get_codecs(pvt->rtp), pt);
if (h323debug)
ast_debug(1, "Native format is set to %d from %d by RTP payload type %d\n", rtptype.code, pvt->nativeformats, pt);
if (pvt->nativeformats != rtptype.code) {
@@ -2420,7 +2420,7 @@
return;
}
if (pvt->rtp) {
- ast_rtp_codecs_payloads_set_rtpmap_type(&pvt->rtp->codecs, pvt->rtp, payload, "audio", (is_cisco ? "cisco-telephone-event" : "telephone-event"), 0);
+ ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(pvt->rtp), pvt->rtp, payload, "audio", (is_cisco ? "cisco-telephone-event" : "telephone-event"), 0);
}
pvt->dtmf_pt[is_cisco ? 1 : 0] = payload;
ast_mutex_unlock(&pvt->lock);
@@ -2451,7 +2451,7 @@
}
}
if (pvt->rtp)
- ast_rtp_codecs_packetization_set(&pvt->rtp->codecs, pvt->rtp, &pvt->peer_prefs);
+ ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(pvt->rtp), pvt->rtp, &pvt->peer_prefs);
}
ast_mutex_unlock(&pvt->lock);
}
@@ -3177,8 +3177,8 @@
ast_log(LOG_ERROR, "No Private Structure, this is bad\n");
return -1;
}
- memcpy(&them, &rtp->remote_address, sizeof(them));
- memcpy(&us, &rtp->local_address, sizeof(us));
+ ast_rtp_instance_get_remote_address(rtp, &them);
+ ast_rtp_instance_get_local_address(rtp, &us);
#if 0 /* Native bridge still isn't ready */
h323_native_bridge(pvt->cd.call_token, ast_inet_ntoa(them.sin_addr), mode);
#endif
Modified: team/file/rtp_engine-mark2/channels/chan_jingle.c
URL: http://svn.digium.com/svn-view/asterisk/team/file/rtp_engine-mark2/channels/chan_jingle.c?view=diff&rev=185703&r1=185702&r2=185703
==============================================================================
--- team/file/rtp_engine-mark2/channels/chan_jingle.c (original)
+++ team/file/rtp_engine-mark2/channels/chan_jingle.c Tue Mar 31 19:31:38 2009
@@ -614,7 +614,7 @@
goto safeout;
}
- memcpy(&sin, &p->rtp->local_address, sizeof(sin));
+ ast_rtp_instance_get_local_address(p->rtp, &sin);
ast_find_ourip(&us, bindaddr);
/* Setup our first jingle candidate */
@@ -818,7 +818,7 @@
/* Set Frame packetization */
if (i->rtp)
- ast_rtp_codecs_packetization_set(&i->rtp->codecs, i->rtp, &i->prefs);
+ ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(i->rtp), i->rtp, &i->prefs);
tmp->nativeformats = ast_codec_choose(&i->prefs, what, 1) | (i->jointcapability & AST_FORMAT_VIDEO_MASK);
fmt = ast_best_codec(tmp->nativeformats);
@@ -1002,8 +1002,8 @@
ast_copy_string(p->audio_content_name, iks_find_attrib(content, "name"), sizeof(p->audio_content_name));
while (codec) {
- ast_rtp_codecs_payloads_set_m_type(&p->rtp->codecs, p->rtp, atoi(iks_find_attrib(codec, "id")));
- ast_rtp_codecs_payloads_set_rtpmap_type(&p->rtp->codecs, p->rtp, atoi(iks_find_attrib(codec, "id")), "audio", iks_find_attrib(codec, "name"), 0);
+ ast_rtp_codecs_payloads_set_m_type(ast_rtp_instance_get_codecs(p->rtp), p->rtp, atoi(iks_find_attrib(codec, "id")));
+ ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(p->rtp), p->rtp, atoi(iks_find_attrib(codec, "id")), "audio", iks_find_attrib(codec, "name"), 0);
codec = iks_next(codec);
}
}
@@ -1018,8 +1018,8 @@
ast_copy_string(p->video_content_name, iks_find_attrib(content, "name"), sizeof(p->video_content_name));
while (codec) {
- ast_rtp_codecs_payloads_set_m_type(&p->rtp->codecs, p->rtp, atoi(iks_find_attrib(codec, "id")));
- ast_rtp_codecs_payloads_set_rtpmap_type(&p->rtp->codecs, p->rtp, atoi(iks_find_attrib(codec, "id")), "audio", iks_find_attrib(codec, "name"), 0);
+ ast_rtp_codecs_payloads_set_m_type(ast_rtp_instance_get_codecs(p->rtp), p->rtp, atoi(iks_find_attrib(codec, "id")));
+ ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(p->rtp), p->rtp, atoi(iks_find_attrib(codec, "id")), "audio", iks_find_attrib(codec, "name"), 0);
codec = iks_next(codec);
}
}
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=185703&r1=185702&r2=185703
==============================================================================
--- team/file/rtp_engine-mark2/channels/chan_mgcp.c (original)
+++ team/file/rtp_engine-mark2/channels/chan_mgcp.c Tue Mar 31 19:31:38 2009
@@ -1879,7 +1879,7 @@
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_codecs_payloads_clear(&sub->rtp->codecs, sub->rtp);
+ ast_rtp_codecs_payloads_clear(ast_rtp_instance_get_codecs(sub->rtp), sub->rtp);
codecs = ast_strdupa(m + len);
while (!ast_strlen_zero(codecs)) {
if (sscanf(codecs, "%d%n", &codec, &len) != 1) {
@@ -1888,7 +1888,7 @@
ast_log(LOG_WARNING, "Error in codec string '%s' at '%s'\n", m, codecs);
return -1;
}
- ast_rtp_codecs_payloads_set_m_type(&sub->rtp->codecs, sub->rtp, codec);
+ ast_rtp_codecs_payloads_set_m_type(ast_rtp_instance_get_codecs(sub->rtp), sub->rtp, codec);
codec_count++;
codecs += len;
}
@@ -1901,11 +1901,11 @@
if (sscanf(a, "rtpmap: %u %[^/]/", &codec, mimeSubtype) != 2)
continue;
/* Note: should really look at the 'freq' and '#chans' params too */
- ast_rtp_codecs_payloads_set_rtpmap_type(&sub->rtp->codecs, sub->rtp, codec, "audio", mimeSubtype, 0);
+ ast_rtp_codecs_payloads_set_rtpmap_type(ast_rtp_instance_get_codecs(sub->rtp), sub->rtp, codec, "audio", mimeSubtype, 0);
}
/* Now gather all of the codecs that were asked for: */
- ast_rtp_codecs_payload_formats(&sub->rtp->codecs, &peercapability, &peerNonCodecCapability);
+ ast_rtp_codecs_payload_formats(ast_rtp_instance_get_codecs(sub->rtp), &peercapability, &peerNonCodecCapability);
p->capability = capability & peercapability;
if (mgcpdebug) {
ast_verbose("Capabilities: us - %d, them - %d, combined - %d\n",
@@ -2066,9 +2066,9 @@
ast_log(LOG_WARNING, "No way to add SDP without an RTP structure\n");
return -1;
}
- memcpy(&sin, &sub->rtp->local_address, sizeof(sin));
+ ast_rtp_instance_get_local_address(sub->rtp, &sin);
if (rtp) {
- memcpy(&dest, &sub->rtp->remote_address, sizeof(dest));
+ ast_rtp_instance_get_remote_address(sub->rtp, &dest);
} else {
if (sub->tmpdest.sin_addr.s_addr) {
dest.sin_addr = sub->tmpdest.sin_addr;
@@ -2094,7 +2094,7 @@
if (mgcpdebug) {
ast_verbose("Answering with capability %d\n", x);
}
- codec = ast_rtp_codecs_payload_code(&sub->rtp->codecs, 1, x);
+ codec = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(sub->rtp), 1, x);
if (codec > -1) {
snprintf(costr, sizeof(costr), " %d", codec);
strncat(m, costr, sizeof(m) - strlen(m) - 1);
@@ -2108,7 +2108,7 @@
if (mgcpdebug) {
ast_verbose("Answering with non-codec capability %d\n", x);
}
- codec = ast_rtp_codecs_payload_code(&sub->rtp->codecs, 0, x);
+ codec = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(sub->rtp), 0, x);
if (codec > -1) {
snprintf(costr, sizeof(costr), " %d", codec);
strncat(m, costr, sizeof(m) - strlen(m) - 1);
@@ -2147,7 +2147,7 @@
if (ast_strlen_zero(sub->cxident) && rtp) {
/* We don't have a CXident yet, store the destination and
wait a bit */
- memcpy(&sub->tmpdest, &rtp->remote_address, sizeof(sub->tmpdest));
+ ast_rtp_instance_get_remote_address(rtp, &sub->tmpdest);
return 0;
}
ast_copy_string(local, "p:20", sizeof(local));
Modified: team/file/rtp_engine-mark2/channels/chan_sip.c
URL: http://svn.digium.com/svn-view/asterisk/team/file/rtp_engine-mark2/channels/chan_sip.c?view=diff&rev=185703&r1=185702&r2=185703
==============================================================================
--- team/file/rtp_engine-mark2/channels/chan_sip.c (original)
+++ team/file/rtp_engine-mark2/channels/chan_sip.c Tue Mar 31 19:31:38 2009
@@ -4702,9 +4702,8 @@
if (!(dialog->vrtp = ast_rtp_instance_new(dialog->engine, sched, &bindaddr, NULL))) {
return -1;
}
- dialog->vrtp->timeout = global_rtptimeout;
- dialog->vrtp->holdtimeout = global_rtpholdtimeout;
- dialog->vrtp->keepalive = global_rtpkeepalive;
+ ast_rtp_instance_set_timeout(dialog->vrtp, global_rtptimeout);
+ ast_rtp_instance_set_hold_timeout(dialog->vrtp, global_rtpholdtimeout);
ast_rtp_instance_set_prop(dialog->vrtp, AST_RTP_PROPERTY_RTCP, 1);
}
@@ -4713,16 +4712,14 @@
if (!(dialog->trtp = ast_rtp_instance_new(dialog->engine, sched, &bindaddr, NULL))) {
return -1;
}
- dialog->trtp->timeout = global_rtptimeout;
- dialog->trtp->holdtimeout = global_rtpholdtimeout;
- dialog->trtp->keepalive = global_rtpkeepalive;
+ ast_rtp_instance_set_timeout(dialog->trtp, global_rtptimeout);
+ ast_rtp_instance_set_hold_timeout(dialog->trtp, global_rtpholdtimeout);
ast_rtp_instance_set_prop(dialog->trtp, AST_RTP_PROPERTY_RTCP, 1);
}
- dialog->rtp->timeout = global_rtptimeout;
- dialog->rtp->holdtimeout = global_rtpholdtimeout;
- dialog->rtp->keepalive = global_rtpkeepalive;
+ ast_rtp_instance_set_timeout(dialog->rtp, global_rtptimeout);
+ ast_rtp_instance_set_hold_timeout(dialog->rtp, global_rtpholdtimeout);
ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_RTCP, 1);
ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF, ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
@@ -4784,22 +4781,19 @@
if (dialog->rtp) { /* Audio */
ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF, ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
- dialog->rtp->timeout = peer->rtptimeout;
- dialog->rtp->holdtimeout = peer->rtpholdtimeout;
- dialog->rtp->keepalive = peer->rtpkeepalive;
+ ast_rtp_instance_set_timeout(dialog->rtp, peer->rtptimeout);
+ ast_rtp_instance_set_hold_timeout(dialog->rtp, peer->rtpholdtimeout);
/* Set Frame packetization */
- ast_rtp_codecs_packetization_set(&dialog->rtp->codecs, dialog->rtp, &dialog->prefs);
+ ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(dialog->rtp), dialog->rtp, &dialog->prefs);
dialog->autoframing = peer->autoframing;
}
if (dialog->vrtp) { /* Video */
- dialog->vrtp->timeout = peer->rtptimeout;
- dialog->vrtp->holdtimeout = peer->rtpholdtimeout;
- dialog->vrtp->keepalive = peer->rtpkeepalive;
+ ast_rtp_instance_set_timeout(dialog->vrtp, peer->rtptimeout);
+ ast_rtp_instance_set_hold_timeout(dialog->vrtp, peer->rtpholdtimeout);
}
if (dialog->trtp) { /* Realtime text */
- dialog->trtp->timeout = peer->rtptimeout;
- dialog->trtp->holdtimeout = peer->rtpholdtimeout;
- dialog->trtp->keepalive = peer->rtpkeepalive;
+ ast_rtp_instance_set_timeout(dialog->trtp, peer->rtptimeout);
+ ast_rtp_instance_set_hold_timeout(dialog->trtp, peer->rtpholdtimeout);
}
ast_string_field_set(dialog, peername, peer->name);
@@ -5985,7 +5979,7 @@
sip_pvt_lock(p);
switch (ast_test_flag(&p->flags[0], SIP_DTMF)) {
case SIP_DTMF_INBAND:
- if (p->rtp && p->rtp->dtmf_mode == AST_RTP_DTMF_MODE_INBAND) {
+ if (p->rtp && ast_rtp_instance_dtmf_mode_get(p->rtp) == AST_RTP_DTMF_MODE_INBAND) {
ast_rtp_instance_dtmf_begin(p->rtp, digit);
} else {
res = -1; /* Tell Asterisk to generate inband indications */
@@ -6021,7 +6015,7 @@
ast_rtp_instance_dtmf_end(p->rtp, digit);
break;
case SIP_DTMF_INBAND:
- if (p->rtp && p->rtp->dtmf_mode == AST_RTP_DTMF_MODE_INBAND) {
+ if (p->rtp && ast_rtp_instance_dtmf_mode_get(p->rtp) == AST_RTP_DTMF_MODE_INBAND) {
ast_rtp_instance_dtmf_end(p->rtp, digit);
} else {
res = -1; /* Tell Asterisk to stop inband indications */
@@ -7659,8 +7653,10 @@
if (udptlportno > 0) {
sin.sin_port = htons(udptlportno);
if (ast_test_flag(&p->flags[0], SIP_NAT) && ast_test_flag(&p->flags[1], SIP_PAGE2_UDPTL_DESTINATION)) {
- if (p->rtp->remote_address.sin_addr.s_addr) {
- memcpy(&sin.sin_addr, &p->rtp->remote_address.sin_addr, sizeof(&sin.sin_addr));
+ struct sockaddr_in remote_address;
+ ast_rtp_instance_get_remote_address(p->rtp, &remote_address);
+ if (remote_address.sin_addr.s_addr) {
+ memcpy(&sin, &remote_address, sizeof(sin));
if (debug) {
ast_log(LOG_DEBUG, "Peer T.38 UDPTL is set behind NAT and with destination, destination address now %s\n", ast_inet_ntoa(sin.sin_addr));
}
@@ -7771,17 +7767,17 @@
}
}
if (framing && p->autoframing) {
- struct ast_codec_pref *pref = &p->rtp->codecs.pref;
+ struct ast_codec_pref *pref = &ast_rtp_instance_get_codecs(p->rtp)->pref;
int codec_n;
for (codec_n = 0; codec_n < AST_RTP_MAX_PT; codec_n++) {
- struct ast_rtp_payload_type format = ast_rtp_codecs_payload_lookup(&p->rtp->codecs, codec_n);
+ struct ast_rtp_payload_type format = ast_rtp_codecs_payload_lookup(ast_rtp_instance_get_codecs(p->rtp), codec_n);
if (!format.asterisk_format || !format.code) /* non-codec or not found */
continue;
if (option_debug)
ast_log(LOG_DEBUG, "Setting framing for %d to %ld\n", format.code, framing);
ast_codec_pref_setsize(pref, format.code, framing);
}
- ast_rtp_codecs_packetization_set(&p->rtp->codecs, p->rtp, pref);
+ ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, pref);
}
continue;
}
@@ -8075,12 +8071,12 @@
p->red = 0;
}
- ast_rtp_codecs_payloads_copy(&newaudiortp, &p->rtp->codecs, p->rtp);
+ ast_rtp_codecs_payloads_copy(&newaudiortp, ast_rtp_instance_get_codecs(p->rtp), p->rtp);
if (p->vrtp) {
- ast_rtp_codecs_payloads_copy(&newvideortp, &p->vrtp->codecs, p->vrtp);
+ ast_rtp_codecs_payloads_copy(&newvideortp, ast_rtp_instance_get_codecs(p->vrtp), p->vrtp);
}
if (p->trtp) {
- ast_rtp_codecs_payloads_copy(&newtextrtp, &p->trtp->codecs, p->trtp);
+ ast_rtp_codecs_payloads_copy(&newtextrtp, ast_rtp_instance_get_codecs(p->trtp), p->trtp);
}
if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO) {
@@ -8996,11 +8992,11 @@
if (debug)
ast_verbose("Adding codec 0x%x (%s) to SDP\n", codec, ast_getformatname(codec));
- if ((rtp_code = ast_rtp_codecs_payload_code(&p->rtp->codecs, 1, codec)) == -1)
+ if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->rtp), 1, codec)) == -1)
return;
if (p->rtp) {
- struct ast_codec_pref *pref = &p->rtp->codecs.pref;
+ struct ast_codec_pref *pref = &ast_rtp_instance_get_codecs(p->rtp)->pref;
fmt = ast_codec_pref_getsize(pref, codec);
} else /* I dont see how you couldn't have p->rtp, but good to check for and error out if not there like earlier code */
return;
@@ -9055,7 +9051,7 @@
if (debug)
ast_verbose("Adding video codec 0x%x (%s) to SDP\n", codec, ast_getformatname(codec));
- if ((rtp_code = ast_rtp_codecs_payload_code(&p->vrtp->codecs, 1, codec)) == -1)
+ if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->vrtp), 1, codec)) == -1)
return;
ast_str_append(m_buf, 0, " %d", rtp_code);
@@ -9078,7 +9074,7 @@
if (debug)
ast_verbose("Adding text codec 0x%x (%s) to SDP\n", codec, ast_getformatname(codec));
- if ((rtp_code = ast_rtp_codecs_payload_code(&p->trtp->codecs, 1, codec)) == -1)
+ if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->trtp), 1, codec)) == -1)
return;
ast_str_append(m_buf, 0, " %d", rtp_code);
@@ -9088,7 +9084,7 @@
/* Add fmtp code here */
if (codec == AST_FORMAT_T140RED) {
- int t140code = ast_rtp_codecs_payload_code(&p->trtp->codecs, 1, AST_FORMAT_T140);
+ int t140code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->trtp), 1, AST_FORMAT_T140);
ast_str_append(a_buf, 0, "a=fmtp:%d %d/%d/%d\r\n", rtp_code,
t140code,
t140code,
@@ -9136,7 +9132,7 @@
if (debug)
ast_verbose("Adding non-codec 0x%x (%s) to SDP\n", format, ast_rtp_lookup_mime_subtype2(0, format, 0));
- if ((rtp_code = ast_rtp_codecs_payload_code(&p->rtp->codecs, 0, format)) == -1)
+ if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->rtp), 0, format)) == -1)
return;
ast_str_append(m_buf, 0, " %d", rtp_code);
@@ -9155,11 +9151,11 @@
struct sockaddr_in *dest, struct sockaddr_in *vdest)
{
/* First, get our address */
- memcpy(sin, &p->rtp->local_address, sizeof(*sin));
+ ast_rtp_instance_get_local_address(p->rtp, sin);
if (p->vrtp)
- memcpy(vsin, &p->vrtp->local_address, sizeof(*vsin));
+ ast_rtp_instance_get_local_address(p->vrtp, vsin);
if (p->trtp)
- memcpy(tsin, &p->trtp->local_address, sizeof(*tsin));
+ ast_rtp_instance_get_local_address(p->trtp, tsin);
/* Now, try to figure out where we want them to send data */
/* Is this a re-invite to move the media out, then use the original offer from caller */
@@ -9590,7 +9586,7 @@
if (p->rtp) {
if (!p->autoframing && !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
ast_debug(1, "Setting framing from config on incoming call\n");
- ast_rtp_codecs_packetization_set(&p->rtp->codecs, p->rtp, &p->prefs);
+ ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, &p->prefs);
}
try_suggested_sip_codec(p);
if (p->t38.state == T38_PEER_DIRECT || p->t38.state == T38_ENABLED) {
@@ -12915,7 +12911,7 @@
/* XXX what about p->prefs = peer->prefs; ? */
/* Set Frame packetization */
if (p->rtp) {
- ast_rtp_codecs_packetization_set(&p->rtp->codecs, p->rtp, &peer->prefs);
+ ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, &peer->prefs);
p->autoframing = peer->autoframing;
}
@@ -13014,7 +13010,7 @@
if (p->t38.peercapability)
p->t38.jointcapability &= p->t38.peercapability;
if (!dialog_initialize_rtp(p)) {
- ast_rtp_codecs_packetization_set(&p->rtp->codecs, p->rtp, &peer->prefs);
+ ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, &peer->prefs);
p->autoframing = peer->autoframing;
} else {
res = AUTH_RTP_FAILED;
@@ -13943,13 +13939,13 @@
}
/* We absolutely cannot destroy the rtp struct while a bridge is active or we WILL crash */
- if (dialog->rtp && dialog->rtp->bridged) {
+ if (dialog->rtp && ast_rtp_instance_get_bridged(dialog->rtp)) {
ast_debug(2, "Bridge still active. Delaying destroy of SIP dialog '%s' Method: %s\n", dialog->callid, sip_methods[dialog->method].text);
sip_pvt_unlock(dialog);
return 0;
}
- if (dialog->vrtp && dialog->vrtp->bridged) {
+ if (dialog->vrtp && ast_rtp_instance_get_bridged(dialog->vrtp)) {
ast_debug(2, "Bridge still active. Delaying destroy of SIP dialog '%s' Method: %s\n", dialog->callid, sip_methods[dialog->method].text);
sip_pvt_unlock(dialog);
return 0;
@@ -13963,13 +13959,13 @@
- if that's the case, wait with destruction */
if (dialog->needdestroy && !dialog->packets && !dialog->owner) {
/* We absolutely cannot destroy the rtp struct while a bridge is active or we WILL crash */
- if (dialog->rtp && dialog->rtp->bridged) {
+ if (dialog->rtp && ast_rtp_instance_get_bridged(dialog->rtp)) {
ast_debug(2, "Bridge still active. Delaying destruction of SIP dialog '%s' Method: %s\n", dialog->callid, sip_methods[dialog->method].text);
sip_pvt_unlock(dialog);
return 0;
}
- if (dialog->vrtp && dialog->vrtp->bridged) {
+ if (dialog->vrtp && ast_rtp_instance_get_bridged(dialog->vrtp)) {
ast_debug(2, "Bridge still active. Delaying destroy of SIP dialog '%s' Method: %s\n", dialog->callid, sip_methods[dialog->method].text);
sip_pvt_unlock(dialog);
return 0;
@@ -19784,11 +19780,11 @@
args.type = "audio";
if (!strcasecmp(args.type, "audio"))
- memcpy(&sin, &p->rtp->remote_address, sizeof(sin));
+ ast_rtp_instance_get_remote_address(p->rtp, &sin);
else if (!strcasecmp(args.type, "video"))
- memcpy(&sin, &p->vrtp->remote_address, sizeof(sin));
+ ast_rtp_instance_get_remote_address(p->vrtp, &sin);
else if (!strcasecmp(args.type, "text"))
- memcpy(&sin, &p->trtp->remote_address, sizeof(sin));
+ ast_rtp_instance_get_remote_address(p->trtp, &sin);
else
return -1;
@@ -21110,7 +21106,7 @@
return;
/* If we have no timers set, return now */
- if (!dialog->rtp->keepalive && !dialog->rtp->timeout && !dialog->rtp->holdtimeout) {
+ if (!ast_rtp_instance_get_timeout(dialog->rtp) && !ast_rtp_instance_get_hold_timeout(dialog->rtp)) {
return;
}
@@ -21121,32 +21117,32 @@
*/
/* Check AUDIO RTP timers */
- if (dialog->lastrtprx && (dialog->rtp->timeout || dialog->rtp->holdtimeout) && (t > dialog->lastrtprx + dialog->rtp->timeout)) {
- if (!ast_test_flag(&dialog->flags[1], SIP_PAGE2_CALL_ONHOLD) || (dialog->rtp->holdtimeout && (t > dialog->lastrtprx + dialog->rtp->holdtimeout))) {
- /* Needs a hangup */
- if (dialog->rtp->timeout) {
- while (dialog->owner && ast_channel_trylock(dialog->owner)) {
- sip_pvt_unlock(dialog);
- usleep(1);
- sip_pvt_lock(dialog);
- }
- ast_log(LOG_NOTICE, "Disconnecting call '%s' for lack of RTP activity in %ld seconds\n",
- dialog->owner->name, (long) (t - dialog->lastrtprx));
- /* Issue a softhangup */
- ast_softhangup_nolock(dialog->owner, AST_SOFTHANGUP_DEV);
- ast_channel_unlock(dialog->owner);
- /* forget the timeouts for this call, since a hangup
- has already been requested and we don't want to
- repeatedly request hangups
- */
- dialog->rtp->timeout = 0;
- dialog->rtp->holdtimeout = 0;
- if (dialog->vrtp) {
- dialog->vrtp->timeout = 0;
- dialog->vrtp->holdtimeout = 0;
- }
- }
- }
+ if (dialog->lastrtprx && (ast_rtp_instance_get_timeout(dialog->rtp) || ast_rtp_instance_get_hold_timeout(dialog->rtp)) && (t > dialog->lastrtprx + ast_rtp_instance_get_timeout(dialog->rtp))) {
+ if (!ast_test_flag(&dialog->flags[1], SIP_PAGE2_CALL_ONHOLD) || (ast_rtp_instance_get_hold_timeout(dialog->rtp) && (t > dialog->lastrtprx + ast_rtp_instance_get_hold_timeout(dialog->rtp)))) {
+ /* Needs a hangup */
+ if (ast_rtp_instance_get_timeout(dialog->rtp)) {
+ while (dialog->owner && ast_channel_trylock(dialog->owner)) {
+ sip_pvt_unlock(dialog);
+ usleep(1);
+ sip_pvt_lock(dialog);
+ }
+ ast_log(LOG_NOTICE, "Disconnecting call '%s' for lack of RTP activity in %ld seconds\n",
+ dialog->owner->name, (long) (t - dialog->lastrtprx));
+ /* Issue a softhangup */
+ ast_softhangup_nolock(dialog->owner, AST_SOFTHANGUP_DEV);
+ ast_channel_unlock(dialog->owner);
+ /* forget the timeouts for this call, since a hangup
+ has already been requested and we don't want to
+ repeatedly request hangups
+ */
+ ast_rtp_instance_set_timeout(dialog->rtp, 0);
+ ast_rtp_instance_set_hold_timeout(dialog->rtp, 0);
+ if (dialog->vrtp) {
+ ast_rtp_instance_set_timeout(dialog->vrtp, 0);
+ ast_rtp_instance_set_hold_timeout(dialog->vrtp, 0);
+ }
+ }
+ }
}
}
Modified: team/file/rtp_engine-mark2/channels/chan_skinny.c
URL: http://svn.digium.com/svn-view/asterisk/team/file/rtp_engine-mark2/channels/chan_skinny.c?view=diff&rev=185703&r1=185702&r2=185703
==============================================================================
--- team/file/rtp_engine-mark2/channels/chan_skinny.c (original)
+++ team/file/rtp_engine-mark2/channels/chan_skinny.c Tue Mar 31 19:31:38 2009
@@ -2632,7 +2632,7 @@
s = d->session;
if (rtp){
- memcpy(&them, &rtp->remote_address, sizeof(them));
+ ast_rtp_instance_get_remote_address(rtp, &them);
/* Shutdown any early-media or previous media on re-invite */
if (!(req = req_alloc(sizeof(struct stop_media_transmission_message), STOP_MEDIA_TRANSMISSION_MESSAGE)))
@@ -2656,7 +2656,7 @@
req->data.startmedia.conferenceId = htolel(sub->callid);
req->data.startmedia.passThruPartyId = htolel(sub->callid);
if (!(l->canreinvite) || (l->nat)){
- memcpy(&us, &rtp->local_address, sizeof(us));
+ ast_rtp_instance_get_local_address(rtp, &us);
req->data.startmedia.remoteIp = htolel(d->ourip.s_addr);
req->data.startmedia.remotePort = htolel(ntohs(us.sin_port));
} else {
@@ -3590,7 +3590,7 @@
}
/* Set Frame packetization */
if (sub->rtp)
- ast_rtp_codecs_packetization_set(&sub->rtp->codecs, sub->rtp, &l->prefs);
+ ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(sub->rtp), sub->rtp, &l->prefs);
/* Create the RTP connection */
transmit_connect(d, sub);
@@ -5547,7 +5547,7 @@
if (sub->rtp) {
ast_rtp_instance_set_remote_address(sub->rtp, &sin);
- memcpy(&us, &sub->rtp->local_address, sizeof(us));
+ ast_rtp_instance_get_local_address(sub->rtp, &us);
} else {
ast_log(LOG_ERROR, "No RTP structure, this is very bad\n");
return 0;
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=185703&r1=185702&r2=185703
==============================================================================
--- team/file/rtp_engine-mark2/channels/chan_unistim.c (original)
+++ team/file/rtp_engine-mark2/channels/chan_unistim.c Tue Mar 31 19:31:38 2009
@@ -2072,7 +2072,7 @@
ast_rtp_instance_set_prop(sub->rtp, AST_RTP_PROPERTY_NAT, sub->parent->parent->nat);
/* Create the RTP connection */
- memcpy(&us, &sub->rtp->local_address, sizeof(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,
@@ -2090,7 +2090,7 @@
sub->owner->readformat = fmt;
sub->owner->writeformat = fmt;
}
- codec = ast_rtp_codecs_payload_code(&sub->rtp->codecs, 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 */
Modified: team/file/rtp_engine-mark2/include/asterisk/rtp_engine.h
URL: http://svn.digium.com/svn-view/asterisk/team/file/rtp_engine-mark2/include/asterisk/rtp_engine.h?view=diff&rev=185703&r1=185702&r2=185703
==============================================================================
--- team/file/rtp_engine-mark2/include/asterisk/rtp_engine.h (original)
+++ team/file/rtp_engine-mark2/include/asterisk/rtp_engine.h Tue Mar 31 19:31:38 2009
@@ -364,30 +364,7 @@
};
/*! Structure that represents an RTP session (instance) */
-struct ast_rtp_instance {
- /*! Engine that is handling this RTP instance */
- struct ast_rtp_engine *engine;
- /*! Data unique to the RTP engine */
- void *data;
- /*! RTP properties that have been set and their value */
- int properties[AST_RTP_PROPERTY_MAX];
- /*! Address that we are expecting RTP to come in to */
- struct sockaddr_in local_address;
- /*! Address that we are sending RTP to */
- struct sockaddr_in remote_address;
- /*! Instance that we are bridged to if doing remote or local bridging */
- struct ast_rtp_instance *bridged;
- /*! Payload and packetization information */
- struct ast_rtp_codecs codecs;
- /*! RTP timeout time (negative or zero means disabled, negative value means temporarily disabled) */
- int timeout;
- /*! RTP timeout when on hold (negative or zero means disabled, negative value means temporarily disabled). */
- int holdtimeout;
- /*! Send RTP comfort noice packets for keepalive */
- int keepalive;
- /*! DTMF mode in use */
- enum ast_rtp_dtmf_mode dtmf_mode;
-};
+struct ast_rtp_instance;
/*! Structure that represents the glue that binds an RTP instance to a channel */
struct ast_rtp_glue {
@@ -552,6 +529,38 @@
int ast_rtp_instance_destroy(struct ast_rtp_instance *instance);
/*!
+ * \brief Set the data portion of an RTP instance
+ *
+ * \param instance The RTP instance to manipulate
+ * \param data Pointer to data
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_instance_set_data(instance, blob);
+ * \endcode
+ *
+ * This sets the data pointer on the RTP instance pointed to by 'instance' to
+ * blob.
+ */
+void ast_rtp_instance_set_data(struct ast_rtp_instance *instance, void *data);
+
+/*!
+ * \brief Get the data portion of an RTP instance
+ *
+ * \param instance The RTP instance we want the data portion from
+ *
+ * Example usage:
+ *
+ * \code
+ * struct *blob = ast_rtp_instance_get_data(instance);
+ ( \endcode
+ *
+ * This gets the data pointer on the RTP instance pointed to by 'instance'.
+ */
+void *ast_rtp_instance_get_data(struct ast_rtp_instance *instance);
+
+/*!
* \brief Send a frame out over RTP
*
* \param instance The RTP instance to send frame out on
@@ -609,6 +618,46 @@
* structure.
*/
int ast_rtp_instance_set_remote_address(struct ast_rtp_instance *instance, struct sockaddr_in *address);
+
+/*!
+ * \brief Set the address that we are expecting to receive RTP on
+ *
+ * \param instance The RTP instance to change the address on
+ * \param address Address to set it to
+ *
+ * \retval 0 success
+ * \retval -1 failure
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_instance_set_local_address(instance, &sin);
+ * \endcode
+ *
+ * This changes the local address that RTP is expected on to the address given in the sin
+ * structure.
+ */
+int ast_rtp_instance_set_local_address(struct ast_rtp_instance *instance, struct sockaddr_in *address);
+
+/*!
+ * \brief Get the local address that we are expecting RTP on
+ *
+ * \param instance The RTP instance to get the address from
+ * \param address The variable to store the address in
+ *
+ * \retval 0 success
+ * \retval -1 failure
+ *
+ * Example usage:
+ *
+ * \code
+ * struct sockaddr_in sin;
+ * ast_rtp_instance_get_local_address(instance, &sin);
+ * \endcode
+ *
+ * This gets the local address that we are expecting RTP on and stores it in the 'sin' structure.
+ */
+int ast_rtp_instance_get_local_address(struct ast_rtp_instance *instance, struct sockaddr_in *address);
/*!
* \brief Get the address of the remote endpoint that we are sending RTP to
@@ -666,6 +715,39 @@
* This enables the AST_RTP_PROPERTY_NAT property on the instance pointed to by instance.
*/
void ast_rtp_instance_set_prop(struct ast_rtp_instance *instance, enum ast_rtp_property property, int value);
+
+/*!
+ * \brief Get the value of an RTP instance property
+ *
+ * \param instance The RTP instance to get the property from
+ * \param property The property to get
+ *
+ * \retval Current value of the property
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_NAT);
+ * \endcode
+ *
+ * This returns the current value of the NAT property on the instance pointed to by instance.
+ */
+int ast_rtp_instance_get_prop(struct ast_rtp_instance *instance, enum ast_rtp_property property);
+
+/*!
+ * \brief Get the codecs structure of an RTP instance
+ *
+ * \param instance The RTP instance to get the codecs structure from
+ *
+ * Example usage:
+ *
+ * \code
+ * struct ast_rtp_codecs *codecs = ast_rtp_instance_get_codecs(instance);
+ * \endcode
+ *
+ * This gets the codecs structure on the RTP instance pointed to by 'instance'.
+ */
+struct ast_rtp_codecs *ast_rtp_instance_get_codecs(struct ast_rtp_instance *instance);
/*!
* \brief Clear payload information from an RTP instance
@@ -984,6 +1066,23 @@
int ast_rtp_instance_dtmf_mode_set(struct ast_rtp_instance *instance, enum ast_rtp_dtmf_mode dtmf_mode);
/*!
+ * \brief Get the DTMF mode of an RTP instance
+ *
+ * \param instance The RTP instance to get the DTMF mode of
+ *
+ * \retval DTMF mode
+ *
+ * Example usage:
+ *
+ * \code
[... 916 lines stripped ...]
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