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Use rtp properties instead of adding a callback 

Thanks, Josh.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@221278 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
twilson 2009-09-30 18:21:03 +00:00
parent bc354c76f4
commit b8e1d3fe36
4 changed files with 11 additions and 48 deletions
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20
channels/chan_sip.c
View File

@ -5191,11 +5191,9 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
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));
ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_CONSTANT_SSRC, ast_test_flag(&dialog->flags[1], SIP_PAGE2_CONSTANT_SSRC));
ast_rtp_instance_set_timeout(dialog->rtp, peer->rtptimeout);
ast_rtp_instance_set_hold_timeout(dialog->rtp, peer->rtpholdtimeout);
if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_CONSTANT_SSRC)) {
ast_rtp_instance_set_constantssrc(dialog->rtp);
}
/* Set Frame packetization */
ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(dialog->rtp), dialog->rtp, &dialog->prefs);
dialog->autoframing = peer->autoframing;
@ -5203,9 +5201,7 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
if (dialog->vrtp) { /* Video */
ast_rtp_instance_set_timeout(dialog->vrtp, peer->rtptimeout);
ast_rtp_instance_set_hold_timeout(dialog->vrtp, peer->rtpholdtimeout);
if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_CONSTANT_SSRC)) {
ast_rtp_instance_set_constantssrc(dialog->vrtp);
}
ast_rtp_instance_set_prop(dialog->vrtp, AST_RTP_PROPERTY_CONSTANT_SSRC, ast_test_flag(&dialog->flags[1], SIP_PAGE2_CONSTANT_SSRC));
}
if (dialog->trtp) { /* Realtime text */
ast_rtp_instance_set_timeout(dialog->trtp, peer->rtptimeout);
@ -20495,13 +20491,11 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
ast_debug(1, "No compatible codecs for this SIP call.\n");
return -1;
}
if (ast_test_flag(&p->flags[1], SIP_PAGE2_CONSTANT_SSRC)) {
if (p->rtp) {
ast_rtp_instance_set_constantssrc(p->rtp);
}
if (p->vrtp) {
ast_rtp_instance_set_constantssrc(p->vrtp);
}
if (p->rtp) {
ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_CONSTANT_SSRC, ast_test_flag(&p->flags[1], SIP_PAGE2_CONSTANT_SSRC));
}
if (p->vrtp) {
ast_rtp_instance_set_prop(p->vrtp, AST_RTP_PROPERTY_CONSTANT_SSRC, ast_test_flag(&p->flags[1], SIP_PAGE2_CONSTANT_SSRC));
}
} else { /* No SDP in invite, call control session */
p->jointcapability = p->capability;

19
include/asterisk/rtp_engine.h
View File

@ -94,6 +94,8 @@ enum ast_rtp_property {
AST_RTP_PROPERTY_RTCP,
/*! Maximum number of RTP properties supported */
AST_RTP_PROPERTY_MAX,
/*! Don't force a new SSRC on new source */
AST_RTP_PROPERTY_CONSTANT_SSRC,
};
/*! Additional RTP options */
@ -1184,23 +1186,6 @@ int ast_rtp_instance_dtmf_mode_set(struct ast_rtp_instance *instance, enum ast_r
*/
enum ast_rtp_dtmf_mode ast_rtp_instance_dtmf_mode_get(struct ast_rtp_instance *instance);
/*!
* \brief Mark an RTP instance not to update SSRC on a new source
*
* \param instance Instance to update
*
* Example usage:
*
* \code
* ast_rtp_instance_set_constantssrc(instance);
* \endcode
*
* This sets the indicated instance to not update the RTP SSRC when new_source
* is called.
*
* \since 1.6.3
*/
void ast_rtp_instance_set_constantssrc(struct ast_rtp_instance *instance);
/*!
* \brief Indicate a new source of audio has dropped in
*

7
main/rtp_engine.c
View File

@ -726,13 +726,6 @@ enum ast_rtp_dtmf_mode ast_rtp_instance_dtmf_mode_get(struct ast_rtp_instance *i
return instance->dtmf_mode;
}
void ast_rtp_instance_set_constantssrc(struct ast_rtp_instance *instance)
{
if (instance->engine->constant_ssrc_set) {
instance->engine->constant_ssrc_set(instance);
}
}
void ast_rtp_instance_new_source(struct ast_rtp_instance *instance)
{
if (instance->engine->new_source) {

13
res/res_rtp_asterisk.c
View File

@ -103,7 +103,6 @@ enum strict_rtp_state {
#define FLAG_NAT_INACTIVE_NOWARN (1 << 1)
#define FLAG_NEED_MARKER_BIT (1 << 3)
#define FLAG_DTMF_COMPENSATE (1 << 4)
#define FLAG_CONSTANT_SSRC (1 << 5)
/*! \brief RTP session description */
struct ast_rtp {
@ -254,7 +253,6 @@ static int ast_rtp_destroy(struct ast_rtp_instance *instance);
static int ast_rtp_dtmf_begin(struct ast_rtp_instance *instance, char digit);
static int ast_rtp_dtmf_end(struct ast_rtp_instance *instance, char digit);
static void ast_rtp_new_source(struct ast_rtp_instance *instance);
static void ast_rtp_set_constantssrc(struct ast_rtp_instance *instance);
static int ast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *frame);
static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtcp);
static void ast_rtp_prop_set(struct ast_rtp_instance *instance, enum ast_rtp_property property, int value);
@ -277,7 +275,6 @@ static struct ast_rtp_engine asterisk_rtp_engine = {
.dtmf_begin = ast_rtp_dtmf_begin,
.dtmf_end = ast_rtp_dtmf_end,
.new_source = ast_rtp_new_source,
.constant_ssrc_set = ast_rtp_set_constantssrc,
.write = ast_rtp_write,
.read = ast_rtp_read,
.prop_set = ast_rtp_prop_set,
@ -656,13 +653,6 @@ static int ast_rtp_dtmf_end(struct ast_rtp_instance *instance, char digit)
return 0;
}
void ast_rtp_set_constantssrc(struct ast_rtp_instance *instance)
{
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
ast_set_flag(rtp, FLAG_CONSTANT_SSRC);
}
static void ast_rtp_new_source(struct ast_rtp_instance *instance)
{
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
@ -670,7 +660,8 @@ static void ast_rtp_new_source(struct ast_rtp_instance *instance)
/* We simply set this bit so that the next packet sent will have the marker bit turned on */
ast_set_flag(rtp, FLAG_NEED_MARKER_BIT);
if (!ast_test_flag(rtp, FLAG_CONSTANT_SSRC)) {
if (!ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_CONSTANT_SSRC)) {
ast_log(LOG_ERROR, "Changing ssrc\n");
rtp->ssrc = ast_random();
}