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_qos(dialog->rtp, global_tos_audio, 0, "SIP RTP");
+ ast_rtp_instance_set_qos(dialog->rtp, global_tos_audio, global_cos_audio, "SIP RTP");
do_setnat(dialog);
static int ast_rtp_dtmf_compatible(struct ast_channel *chan0, struct ast_rtp_instance *instance0, struct ast_channel *chan1, struct ast_rtp_instance *instance1);
static void ast_rtp_stun_request(struct ast_rtp_instance *instance, struct ast_sockaddr *suggestion, const char *username);
static void ast_rtp_stop(struct ast_rtp_instance *instance);
+static int ast_rtp_qos_set(struct ast_rtp_instance *instance, int tos, int cos, const char* desc);
/* RTP Engine Declaration */
static struct ast_rtp_engine asterisk_rtp_engine = {
.dtmf_compatible = ast_rtp_dtmf_compatible,
.stun_request = ast_rtp_stun_request,
.stop = ast_rtp_stop,
+ .qos = ast_rtp_qos_set,
};
static inline int rtp_debug_test_addr(struct ast_sockaddr *addr)
ast_set_flag(rtp, FLAG_NEED_MARKER_BIT);
}
+static int ast_rtp_qos_set(struct ast_rtp_instance *instance, int tos, int cos, const char *desc)
+{
+ struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+
+ return ast_set_qos(rtp->s, tos, cos, desc);
+}
+
static char *rtp_do_debug_ip(struct ast_cli_args *a)
{
char *arg = ast_strdupa(a->argv[4]);