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Diffstat (limited to 'channels/chan_sip.c')
-rw-r--r--channels/chan_sip.c20
1 files changed, 7 insertions, 13 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 9c0076554..908bc80cc 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -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;