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authorzuul <zuul@gerrit.asterisk.org>2016-02-29 18:30:37 -0600
committerGerrit Code Review <gerrit2@gerrit.digium.api>2016-02-29 18:30:38 -0600
commitd1495bc0ebed86d322488bed647b43aba0ae9b1f (patch)
treea0073405ec7d8e62a32f4c64bd8c1ba888e4c86c
parent84d10362056611012968957183aa0220d79395d0 (diff)
parent18a323e54298fee27d91ef0fe634895960a732ed (diff)
Merge "chan_sip.c: Fix T.38 issues caused by leaving a bridge." into 13
-rw-r--r--channels/chan_sip.c39
1 files changed, 30 insertions, 9 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 3b6c00513..4e0d81159 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -7517,8 +7517,10 @@ static int interpret_t38_parameters(struct sip_pvt *p, const struct ast_control_
AST_SCHED_DEL_UNREF(sched, p->t38id, dialog_unref(p, "when you delete the t38id sched, you should dec the refcount for the stored dialog ptr"));
change_t38_state(p, T38_REJECTED);
transmit_response_reliable(p, "488 Not acceptable here", &p->initreq);
- } else if (p->t38.state == T38_ENABLED)
+ } else if (p->t38.state == T38_ENABLED) {
+ change_t38_state(p, T38_DISABLED);
transmit_reinvite_with_sdp(p, FALSE, FALSE);
+ }
break;
case AST_T38_REQUEST_PARMS: { /* Application wants remote's parameters re-sent */
struct ast_control_t38_parameters parameters = p->t38.their_parms;
@@ -10676,6 +10678,10 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
} else if (udptlportno > 0) {
if (debug)
ast_verbose("Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.\n");
+
+ /* Force media to go through us for T.38. */
+ memset(&p->redirip, 0, sizeof(p->redirip));
+
/* Prevent audio RTCP reads */
if (p->owner) {
ast_channel_set_fd(p->owner, 1, -1);
@@ -13787,6 +13793,29 @@ static int transmit_reinvite_with_sdp(struct sip_pvt *p, int t38version, int old
{
struct sip_request req;
+ if (t38version) {
+ /* Force media to go through us for T.38. */
+ memset(&p->redirip, 0, sizeof(p->redirip));
+ }
+ if (p->rtp) {
+ if (t38version) {
+ /* Silence RTCP while audio RTP is inactive */
+ ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 0);
+ if (p->owner) {
+ /* Prevent audio RTCP reads */
+ ast_channel_set_fd(p->owner, 1, -1);
+ }
+ } else if (ast_sockaddr_isnull(&p->redirip)) {
+ /* Enable RTCP since it will be inactive if we're coming back
+ * with this reinvite */
+ ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 1);
+ if (p->owner) {
+ /* Enable audio RTCP reads */
+ ast_channel_set_fd(p->owner, 1, ast_rtp_instance_fd(p->rtp, 1));
+ }
+ }
+ }
+
reqprep(&req, p, ast_test_flag(&p->flags[0], SIP_REINVITE_UPDATE) ? SIP_UPDATE : SIP_INVITE, 0, 1);
add_header(&req, "Allow", ALLOWED_METHODS);
@@ -32683,14 +32712,6 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *i
} else if (!ast_sockaddr_isnull(&p->redirip)) {
memset(&p->redirip, 0, sizeof(p->redirip));
changed = 1;
-
- if (p->rtp) {
- /* Enable RTCP since it will be inactive if we're coming back
- * from a reinvite */
- ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 1);
- /* Enable audio RTCP reads */
- ast_channel_set_fd(chan, 1, ast_rtp_instance_fd(p->rtp, 1));
- }
}
if (vinstance) {